Initial opus codec support

Synchronised with opus repo on github (https://github.com/freqmod/rockbox-opus)

Status:
* Seeking ported from speex, but fails on some cases (e.g. seek to granule 0)
* ReplayGain parsing needs to be reworked, we do vorbis-style replaygain now.
  http://wiki.xiph.org/OggOpus#Comment_Header explicitly forbids these in
  favour of R128_TRACK_GAIN tag.
* No optimisation yet, source files still nearly identical to opus upstream
* Multi-stream opus files may not be parsed correctly

Change-Id: Ia66f1027dc1d288083e3c57b2816700078376f9a
Reviewed-on: http://gerrit.rockbox.org/300
Reviewed-by: Bertrik Sikken <bertrik@sikken.nl>
Tested-by: Bertrik Sikken <bertrik@sikken.nl>

127 files changed, 28373 insertions, 2 deletions

diff --git a/apps/filetypes.c b/apps/filetypes.c
index b765e7a3b0..8a4cedc0ea 100644
--- a/apps/filetypes.c
+++ b/apps/filetypes.c
@@ -80,6 +80,7 @@ static const struct filetype inbuilt_filetypes[] = {
     { "aif", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
     { "aiff",FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
     { "spx" ,FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
+    { "opus",FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
     { "sid", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
     { "adx", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
     { "nsf", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
diff --git a/lib/rbcodec/codecs/SOURCES b/lib/rbcodec/codecs/SOURCES
index db6e82c75f..039772cf9a 100644
--- a/lib/rbcodec/codecs/SOURCES
+++ b/lib/rbcodec/codecs/SOURCES
@@ -1,5 +1,6 @@
 /* decoders */
 
+opus.c
 vorbis.c
 mpa.c
 flac.c
diff --git a/lib/rbcodec/codecs/codecs.make b/lib/rbcodec/codecs/codecs.make
index 919aef2024..f56c032c60 100644
--- a/lib/rbcodec/codecs/codecs.make
+++ b/lib/rbcodec/codecs/codecs.make
@@ -51,6 +51,7 @@ include $(RBCODECLIB_DIR)/codecs/libgme/libsgc.make
 include $(RBCODECLIB_DIR)/codecs/libgme/libvgm.make
 include $(RBCODECLIB_DIR)/codecs/libgme/libkss.make
 include $(RBCODECLIB_DIR)/codecs/libgme/libemu2413.make
+include $(RBCODECLIB_DIR)/codecs/libopus/libopus.make
 
 # compile flags for codecs
 CODECFLAGS = $(CFLAGS) $(RBCODEC_CFLAGS) -fstrict-aliasing \
@@ -74,6 +75,7 @@ $(KSSLIB) : CODECFLAGS +=  -O2
 $(M4ALIB) : CODECFLAGS += -O3
 $(MUSEPACKLIB) : CODECFLAGS += -O1
 $(NSFLIB) : CODECFLAGS +=  -O2
+$(OPUSLIB) : CODECFLAGS +=  -O2
 $(PCMSLIB) : CODECFLAGS += -O1
 $(RMLIB) : CODECFLAGS += -O3
 $(SGCLIB) : CODECFLAGS +=  -O2
@@ -168,6 +170,7 @@ $(CODECDIR)/nsf.codec : $(CODECDIR)/libnsf.a $(CODECDIR)/libemu2413.a
 $(CODECDIR)/sgc.codec : $(CODECDIR)/libsgc.a $(CODECDIR)/libemu2413.a
 $(CODECDIR)/vgm.codec : $(CODECDIR)/libvgm.a $(CODECDIR)/libemu2413.a
 $(CODECDIR)/kss.codec : $(CODECDIR)/libkss.a $(CODECDIR)/libemu2413.a
+$(CODECDIR)/opus.codec : $(CODECDIR)/libopus.a
 
 $(CODECS): $(CODEC_LIBS) # this must be last in codec dependency list
 
diff --git a/lib/rbcodec/codecs/lib/codeclib.c b/lib/rbcodec/codecs/lib/codeclib.c
index 09c96f921c..a12038eeb8 100644
--- a/lib/rbcodec/codecs/lib/codeclib.c
+++ b/lib/rbcodec/codecs/lib/codeclib.c
@@ -91,7 +91,7 @@ void* codec_realloc(void* ptr, size_t size)
 {
     void* x;
     x = codec_malloc(size);
-    memcpy(x, ptr, size);
+    ci->memcpy(x, ptr, size);
     codec_free(ptr);
     return(x);
 }
diff --git a/lib/rbcodec/codecs/libopus/README.rockbox b/lib/rbcodec/codecs/libopus/README.rockbox
new file mode 100644
index 0000000000..8a5ef41fba
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/README.rockbox
@@ -0,0 +1,26 @@
+Libraries:  Opus (snapshot) / Opus-tools (snapshot) / libogg 1.3
+Imported:   September 15th, 2012
+
+Steps taken to adapt original opus/opus-tool/ogg source files to rockbox
+(useful when for example syncing a new snapshot)
+
+Opus:
+* copied .c/.h files from opus/src lib/rbcodec/codecs/libopus
+* copied .h files from opus/include to lib/rbcodec/codecs/libopus
+* copied .c/.h files from opus/celt to lib/rbcodec/codecs/libopus/celt
+* copied .c/.h files from opus/silk to lib/rbcodec/codecs/libopus/silk
+* renamed opus config.h file to opus_config.h and replaced #include "config.h",
+  for example
+find . -name "*.h" -print | xargs sed -i 's/include "config.h"/include "opus_config.h"/g'
+find . -name "*.c" -print | xargs sed -i 's/include "config.h"/include "opus_config.h"/g'
+
+Opus-tools:
+* copied src/opus_header.h and src/opus_header.c to lib/rbcodec/codecs/libopus
+* changed #include <ogg/ogg.h> to #include "ogg/ogg.h" in opus_header.c
+
+Ogg:
+* copied libogg/src/framing.c to lib/rbcodec/codecs/libopus/ogg
+* copied libogg/include/ogg.h to lib/rbcodec/codecs/libopus/ogg
+* changed #include "ogg/ogg.h" to #include "ogg.h" in framing.c
+* added os_config.h to lib/rbcodec/codecs/libopus/ogg
+
diff --git a/lib/rbcodec/codecs/libopus/SOURCES b/lib/rbcodec/codecs/libopus/SOURCES
new file mode 100644
index 0000000000..df3f014f38
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/SOURCES
@@ -0,0 +1,70 @@
+/* CELT sources */
+celt/bands.c
+celt/celt.c
+celt/celt_lpc.c
+celt/cwrs.c
+celt/entcode.c
+celt/entdec.c
+celt/entenc.c
+celt/kiss_fft.c
+celt/laplace.c
+celt/mathops.c
+celt/mdct.c
+celt/modes.c
+celt/pitch.c
+celt/quant_bands.c
+celt/rate.c
+celt/vq.c
+
+/* SILK sources */
+silk/bwexpander_32.c
+silk/bwexpander.c
+silk/CNG.c
+silk/code_signs.c
+silk/dec_API.c
+silk/decode_core.c
+silk/decode_frame.c
+silk/decode_indices.c
+silk/decode_parameters.c
+silk/decode_pitch.c
+silk/decode_pulses.c
+silk/decoder_set_fs.c
+silk/gain_quant.c
+silk/init_decoder.c
+silk/lin2log.c
+silk/log2lin.c
+silk/LPC_analysis_filter.c
+silk/LPC_inv_pred_gain.c
+silk/NLSF2A.c
+silk/NLSF_decode.c
+silk/NLSF_stabilize.c
+silk/NLSF_unpack.c
+silk/NLSF_VQ_weights_laroia.c
+silk/pitch_est_tables.c
+silk/PLC.c
+silk/resampler.c
+silk/resampler_private_AR2.c
+silk/resampler_private_down_FIR.c
+silk/resampler_private_IIR_FIR.c
+silk/resampler_private_up2_HQ.c
+silk/resampler_rom.c
+silk/shell_coder.c
+silk/sort.c
+silk/stereo_decode_pred.c
+silk/stereo_MS_to_LR.c
+silk/sum_sqr_shift.c
+silk/table_LSF_cos.c
+silk/tables_gain.c
+silk/tables_LTP.c
+silk/tables_NLSF_CB_NB_MB.c
+silk/tables_NLSF_CB_WB.c
+silk/tables_other.c
+silk/tables_pitch_lag.c
+silk/tables_pulses_per_block.c
+
+/* OPUS sources */
+opus_decoder.c
+opus_header.c
+
+/* OGG sources */
+ogg/framing.c
diff --git a/lib/rbcodec/codecs/libopus/celt/_kiss_fft_guts.h b/lib/rbcodec/codecs/libopus/celt/_kiss_fft_guts.h
new file mode 100644
index 0000000000..33e62c6b3d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/_kiss_fft_guts.h
@@ -0,0 +1,175 @@
+/*Copyright (c) 2003-2004, Mark Borgerding
+
+  All rights reserved.
+
+  Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are met:
+
+    * Redistributions of source code must retain the above copyright notice,
+       this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright notice,
+       this list of conditions and the following disclaimer in the
+       documentation and/or other materials provided with the distribution.
+
+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+  AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+  IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+  ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+  LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+  CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+  SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+  INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+  CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+  ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+  POSSIBILITY OF SUCH DAMAGE.*/
+
+#ifndef KISS_FFT_GUTS_H
+#define KISS_FFT_GUTS_H
+
+#define MIN(a,b) ((a)<(b) ? (a):(b))
+#define MAX(a,b) ((a)>(b) ? (a):(b))
+
+/* kiss_fft.h
+   defines kiss_fft_scalar as either short or a float type
+   and defines
+   typedef struct { kiss_fft_scalar r; kiss_fft_scalar i; }kiss_fft_cpx; */
+#include "kiss_fft.h"
+
+/*
+  Explanation of macros dealing with complex math:
+
+   C_MUL(m,a,b)         : m = a*b
+   C_FIXDIV( c , div )  : if a fixed point impl., c /= div. noop otherwise
+   C_SUB( res, a,b)     : res = a - b
+   C_SUBFROM( res , a)  : res -= a
+   C_ADDTO( res , a)    : res += a
+ * */
+#ifdef FIXED_POINT
+#include "arch.h"
+
+
+#define SAMP_MAX 2147483647
+#define TWID_MAX 32767
+#define TRIG_UPSCALE 1
+
+#define SAMP_MIN -SAMP_MAX
+
+
+#   define S_MUL(a,b) MULT16_32_Q15(b, a)
+
+#   define C_MUL(m,a,b) \
+      do{ (m).r = SUB32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \
+          (m).i = ADD32(S_MUL((a).r,(b).i) , S_MUL((a).i,(b).r)); }while(0)
+
+#   define C_MULC(m,a,b) \
+      do{ (m).r = ADD32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \
+          (m).i = SUB32(S_MUL((a).i,(b).r) , S_MUL((a).r,(b).i)); }while(0)
+
+#   define C_MUL4(m,a,b) \
+      do{ (m).r = SHR32(SUB32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)),2); \
+          (m).i = SHR32(ADD32(S_MUL((a).r,(b).i) , S_MUL((a).i,(b).r)),2); }while(0)
+
+#   define C_MULBYSCALAR( c, s ) \
+      do{ (c).r =  S_MUL( (c).r , s ) ;\
+          (c).i =  S_MUL( (c).i , s ) ; }while(0)
+
+#   define DIVSCALAR(x,k) \
+        (x) = S_MUL(  x, (TWID_MAX-((k)>>1))/(k)+1 )
+
+#   define C_FIXDIV(c,div) \
+        do {    DIVSCALAR( (c).r , div);  \
+                DIVSCALAR( (c).i  , div); }while (0)
+
+#define  C_ADD( res, a,b)\
+    do {(res).r=ADD32((a).r,(b).r);  (res).i=ADD32((a).i,(b).i); \
+    }while(0)
+#define  C_SUB( res, a,b)\
+    do {(res).r=SUB32((a).r,(b).r);  (res).i=SUB32((a).i,(b).i); \
+    }while(0)
+#define C_ADDTO( res , a)\
+    do {(res).r = ADD32((res).r, (a).r);  (res).i = ADD32((res).i,(a).i);\
+    }while(0)
+
+#define C_SUBFROM( res , a)\
+    do {(res).r = ADD32((res).r,(a).r);  (res).i = SUB32((res).i,(a).i); \
+    }while(0)
+
+#else  /* not FIXED_POINT*/
+
+#   define S_MUL(a,b) ( (a)*(b) )
+#define C_MUL(m,a,b) \
+    do{ (m).r = (a).r*(b).r - (a).i*(b).i;\
+        (m).i = (a).r*(b).i + (a).i*(b).r; }while(0)
+#define C_MULC(m,a,b) \
+    do{ (m).r = (a).r*(b).r + (a).i*(b).i;\
+        (m).i = (a).i*(b).r - (a).r*(b).i; }while(0)
+
+#define C_MUL4(m,a,b) C_MUL(m,a,b)
+
+#   define C_FIXDIV(c,div) /* NOOP */
+#   define C_MULBYSCALAR( c, s ) \
+    do{ (c).r *= (s);\
+        (c).i *= (s); }while(0)
+#endif
+
+#ifndef CHECK_OVERFLOW_OP
+#  define CHECK_OVERFLOW_OP(a,op,b) /* noop */
+#endif
+
+#ifndef C_ADD
+#define  C_ADD( res, a,b)\
+    do { \
+            CHECK_OVERFLOW_OP((a).r,+,(b).r)\
+            CHECK_OVERFLOW_OP((a).i,+,(b).i)\
+            (res).r=(a).r+(b).r;  (res).i=(a).i+(b).i; \
+    }while(0)
+#define  C_SUB( res, a,b)\
+    do { \
+            CHECK_OVERFLOW_OP((a).r,-,(b).r)\
+            CHECK_OVERFLOW_OP((a).i,-,(b).i)\
+            (res).r=(a).r-(b).r;  (res).i=(a).i-(b).i; \
+    }while(0)
+#define C_ADDTO( res , a)\
+    do { \
+            CHECK_OVERFLOW_OP((res).r,+,(a).r)\
+            CHECK_OVERFLOW_OP((res).i,+,(a).i)\
+            (res).r += (a).r;  (res).i += (a).i;\
+    }while(0)
+
+#define C_SUBFROM( res , a)\
+    do {\
+            CHECK_OVERFLOW_OP((res).r,-,(a).r)\
+            CHECK_OVERFLOW_OP((res).i,-,(a).i)\
+            (res).r -= (a).r;  (res).i -= (a).i; \
+    }while(0)
+#endif /* C_ADD defined */
+
+#ifdef FIXED_POINT
+/*#  define KISS_FFT_COS(phase)  TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * cos (phase))))
+#  define KISS_FFT_SIN(phase)  TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * sin (phase))))*/
+#  define KISS_FFT_COS(phase)  floor(.5+TWID_MAX*cos (phase))
+#  define KISS_FFT_SIN(phase)  floor(.5+TWID_MAX*sin (phase))
+#  define HALF_OF(x) ((x)>>1)
+#elif defined(USE_SIMD)
+#  define KISS_FFT_COS(phase) _mm_set1_ps( cos(phase) )
+#  define KISS_FFT_SIN(phase) _mm_set1_ps( sin(phase) )
+#  define HALF_OF(x) ((x)*_mm_set1_ps(.5f))
+#else
+#  define KISS_FFT_COS(phase) (kiss_fft_scalar) cos(phase)
+#  define KISS_FFT_SIN(phase) (kiss_fft_scalar) sin(phase)
+#  define HALF_OF(x) ((x)*.5f)
+#endif
+
+#define  kf_cexp(x,phase) \
+        do{ \
+                (x)->r = KISS_FFT_COS(phase);\
+                (x)->i = KISS_FFT_SIN(phase);\
+        }while(0)
+
+#define  kf_cexp2(x,phase) \
+   do{ \
+      (x)->r = TRIG_UPSCALE*celt_cos_norm((phase));\
+      (x)->i = TRIG_UPSCALE*celt_cos_norm((phase)-32768);\
+}while(0)
+
+#endif /* KISS_FFT_GUTS_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/arch.h b/lib/rbcodec/codecs/libopus/celt/arch.h
new file mode 100644
index 0000000000..03cda40f69
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/arch.h
@@ -0,0 +1,209 @@
+/* Copyright (c) 2003-2008 Jean-Marc Valin
+   Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file arch.h
+   @brief Various architecture definitions for CELT
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef ARCH_H
+#define ARCH_H
+
+#include "opus_types.h"
+
+# if !defined(__GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define __GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define __GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#define CELT_SIG_SCALE 32768.f
+
+#define celt_fatal(str) _celt_fatal(str, __FILE__, __LINE__);
+#ifdef ENABLE_ASSERTIONS
+#include <stdio.h>
+#include <stdlib.h>
+#ifdef __GNUC__
+__attribute__((noreturn))
+#endif
+static inline void _celt_fatal(const char *str, const char *file, int line)
+{
+   fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str);
+   abort();
+}
+#define celt_assert(cond) {if (!(cond)) {celt_fatal("assertion failed: " #cond);}}
+#define celt_assert2(cond, message) {if (!(cond)) {celt_fatal("assertion failed: " #cond "\n" message);}}
+#else
+#define celt_assert(cond)
+#define celt_assert2(cond, message)
+#endif
+
+#define IMUL32(a,b) ((a)*(b))
+
+#define ABS(x) ((x) < 0 ? (-(x)) : (x))      /**< Absolute integer value. */
+#define ABS16(x) ((x) < 0 ? (-(x)) : (x))    /**< Absolute 16-bit value.  */
+#define MIN16(a,b) ((a) < (b) ? (a) : (b))   /**< Minimum 16-bit value.   */
+#define MAX16(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum 16-bit value.   */
+#define ABS32(x) ((x) < 0 ? (-(x)) : (x))    /**< Absolute 32-bit value.  */
+#define MIN32(a,b) ((a) < (b) ? (a) : (b))   /**< Minimum 32-bit value.   */
+#define MAX32(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum 32-bit value.   */
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))   /**< Minimum int value.   */
+#define IMAX(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum int value.   */
+#define UADD32(a,b) ((a)+(b))
+#define USUB32(a,b) ((a)-(b))
+
+#define PRINT_MIPS(file)
+
+#ifdef FIXED_POINT
+
+typedef opus_int16 opus_val16;
+typedef opus_int32 opus_val32;
+
+typedef opus_val32 celt_sig;
+typedef opus_val16 celt_norm;
+typedef opus_val32 celt_ener;
+
+#define Q15ONE 32767
+
+#define SIG_SHIFT 12
+
+#define NORM_SCALING 16384
+
+#define DB_SHIFT 10
+
+#define EPSILON 1
+#define VERY_LARGE16 ((opus_val16)32767)
+#define Q15_ONE ((opus_val16)32767)
+
+#define SCALEIN(a)      (a)
+#define SCALEOUT(a)     (a)
+
+#ifdef FIXED_DEBUG
+#include "fixed_debug.h"
+#else
+
+#include "fixed_generic.h"
+
+#ifdef ARM5E_ASM
+#include "fixed_arm5e.h"
+#elif defined (ARM4_ASM)
+#include "fixed_arm4.h"
+#elif defined (BFIN_ASM)
+#include "fixed_bfin.h"
+#elif defined (TI_C5X_ASM)
+#include "fixed_c5x.h"
+#elif defined (TI_C6X_ASM)
+#include "fixed_c6x.h"
+#endif
+
+#endif
+
+#else /* FIXED_POINT */
+
+typedef float opus_val16;
+typedef float opus_val32;
+
+typedef float celt_sig;
+typedef float celt_norm;
+typedef float celt_ener;
+
+#define Q15ONE 1.0f
+
+#define NORM_SCALING 1.f
+
+#define EPSILON 1e-15f
+#define VERY_LARGE16 1e15f
+#define Q15_ONE ((opus_val16)1.f)
+
+#define QCONST16(x,bits) (x)
+#define QCONST32(x,bits) (x)
+
+#define NEG16(x) (-(x))
+#define NEG32(x) (-(x))
+#define EXTRACT16(x) (x)
+#define EXTEND32(x) (x)
+#define SHR16(a,shift) (a)
+#define SHL16(a,shift) (a)
+#define SHR32(a,shift) (a)
+#define SHL32(a,shift) (a)
+#define PSHR32(a,shift) (a)
+#define VSHR32(a,shift) (a)
+
+#define PSHR(a,shift)   (a)
+#define SHR(a,shift)    (a)
+#define SHL(a,shift)    (a)
+#define SATURATE(x,a)   (x)
+
+#define ROUND16(a,shift)  (a)
+#define HALF16(x)       (.5f*(x))
+#define HALF32(x)       (.5f*(x))
+
+#define ADD16(a,b) ((a)+(b))
+#define SUB16(a,b) ((a)-(b))
+#define ADD32(a,b) ((a)+(b))
+#define SUB32(a,b) ((a)-(b))
+#define MULT16_16_16(a,b)     ((a)*(b))
+#define MULT16_16(a,b)     ((opus_val32)(a)*(opus_val32)(b))
+#define MAC16_16(c,a,b)     ((c)+(opus_val32)(a)*(opus_val32)(b))
+
+#define MULT16_32_Q15(a,b)     ((a)*(b))
+#define MULT16_32_Q16(a,b)     ((a)*(b))
+
+#define MULT32_32_Q31(a,b)     ((a)*(b))
+
+#define MAC16_32_Q15(c,a,b)     ((c)+(a)*(b))
+
+#define MULT16_16_Q11_32(a,b)     ((a)*(b))
+#define MULT16_16_Q13(a,b)     ((a)*(b))
+#define MULT16_16_Q14(a,b)     ((a)*(b))
+#define MULT16_16_Q15(a,b)     ((a)*(b))
+#define MULT16_16_P15(a,b)     ((a)*(b))
+#define MULT16_16_P13(a,b)     ((a)*(b))
+#define MULT16_16_P14(a,b)     ((a)*(b))
+#define MULT16_32_P16(a,b)     ((a)*(b))
+
+#define DIV32_16(a,b)     (((opus_val32)(a))/(opus_val16)(b))
+#define DIV32(a,b)     (((opus_val32)(a))/(opus_val32)(b))
+
+#define SCALEIN(a)      ((a)*CELT_SIG_SCALE)
+#define SCALEOUT(a)     ((a)*(1/CELT_SIG_SCALE))
+
+#endif /* !FIXED_POINT */
+
+#ifndef GLOBAL_STACK_SIZE
+#ifdef FIXED_POINT
+#define GLOBAL_STACK_SIZE 100000
+#else
+#define GLOBAL_STACK_SIZE 100000
+#endif
+#endif
+
+#endif /* ARCH_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/bands.c b/lib/rbcodec/codecs/libopus/celt/bands.c
new file mode 100644
index 0000000000..6e612980b6
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/bands.c
@@ -0,0 +1,1302 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2009 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include <math.h>
+#include "bands.h"
+#include "modes.h"
+#include "vq.h"
+#include "cwrs.h"
+#include "stack_alloc.h"
+#include "os_support.h"
+#include "mathops.h"
+#include "rate.h"
+
+opus_uint32 celt_lcg_rand(opus_uint32 seed)
+{
+   return 1664525 * seed + 1013904223;
+}
+
+/* This is a cos() approximation designed to be bit-exact on any platform. Bit exactness
+   with this approximation is important because it has an impact on the bit allocation */
+static opus_int16 bitexact_cos(opus_int16 x)
+{
+   opus_int32 tmp;
+   opus_int16 x2;
+   tmp = (4096+((opus_int32)(x)*(x)))>>13;
+   celt_assert(tmp<=32767);
+   x2 = tmp;
+   x2 = (32767-x2) + FRAC_MUL16(x2, (-7651 + FRAC_MUL16(x2, (8277 + FRAC_MUL16(-626, x2)))));
+   celt_assert(x2<=32766);
+   return 1+x2;
+}
+
+static int bitexact_log2tan(int isin,int icos)
+{
+   int lc;
+   int ls;
+   lc=EC_ILOG(icos);
+   ls=EC_ILOG(isin);
+   icos<<=15-lc;
+   isin<<=15-ls;
+   return (ls-lc)*(1<<11)
+         +FRAC_MUL16(isin, FRAC_MUL16(isin, -2597) + 7932)
+         -FRAC_MUL16(icos, FRAC_MUL16(icos, -2597) + 7932);
+}
+
+#ifdef FIXED_POINT
+/* Compute the amplitude (sqrt energy) in each of the bands */
+void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M)
+{
+   int i, c, N;
+   const opus_int16 *eBands = m->eBands;
+   N = M*m->shortMdctSize;
+   c=0; do {
+      for (i=0;i<end;i++)
+      {
+         int j;
+         opus_val32 maxval=0;
+         opus_val32 sum = 0;
+
+         j=M*eBands[i]; do {
+            maxval = MAX32(maxval, X[j+c*N]);
+            maxval = MAX32(maxval, -X[j+c*N]);
+         } while (++j<M*eBands[i+1]);
+
+         if (maxval > 0)
+         {
+            int shift = celt_ilog2(maxval)-10;
+            j=M*eBands[i]; do {
+               sum = MAC16_16(sum, EXTRACT16(VSHR32(X[j+c*N],shift)),
+                                   EXTRACT16(VSHR32(X[j+c*N],shift)));
+            } while (++j<M*eBands[i+1]);
+            /* We're adding one here to ensure the normalized band isn't larger than unity norm */
+            bandE[i+c*m->nbEBands] = EPSILON+VSHR32(EXTEND32(celt_sqrt(sum)),-shift);
+         } else {
+            bandE[i+c*m->nbEBands] = EPSILON;
+         }
+         /*printf ("%f ", bandE[i+c*m->nbEBands]);*/
+      }
+   } while (++c<C);
+   /*printf ("\n");*/
+}
+
+/* Normalise each band such that the energy is one. */
+void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M)
+{
+   int i, c, N;
+   const opus_int16 *eBands = m->eBands;
+   N = M*m->shortMdctSize;
+   c=0; do {
+      i=0; do {
+         opus_val16 g;
+         int j,shift;
+         opus_val16 E;
+         shift = celt_zlog2(bandE[i+c*m->nbEBands])-13;
+         E = VSHR32(bandE[i+c*m->nbEBands], shift);
+         g = EXTRACT16(celt_rcp(SHL32(E,3)));
+         j=M*eBands[i]; do {
+            X[j+c*N] = MULT16_16_Q15(VSHR32(freq[j+c*N],shift-1),g);
+         } while (++j<M*eBands[i+1]);
+      } while (++i<end);
+   } while (++c<C);
+}
+
+#else /* FIXED_POINT */
+/* Compute the amplitude (sqrt energy) in each of the bands */
+void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M)
+{
+   int i, c, N;
+   const opus_int16 *eBands = m->eBands;
+   N = M*m->shortMdctSize;
+   c=0; do {
+      for (i=0;i<end;i++)
+      {
+         int j;
+         opus_val32 sum = 1e-27f;
+         for (j=M*eBands[i];j<M*eBands[i+1];j++)
+            sum += X[j+c*N]*X[j+c*N];
+         bandE[i+c*m->nbEBands] = celt_sqrt(sum);
+         /*printf ("%f ", bandE[i+c*m->nbEBands]);*/
+      }
+   } while (++c<C);
+   /*printf ("\n");*/
+}
+
+/* Normalise each band such that the energy is one. */
+void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M)
+{
+   int i, c, N;
+   const opus_int16 *eBands = m->eBands;
+   N = M*m->shortMdctSize;
+   c=0; do {
+      for (i=0;i<end;i++)
+      {
+         int j;
+         opus_val16 g = 1.f/(1e-27f+bandE[i+c*m->nbEBands]);
+         for (j=M*eBands[i];j<M*eBands[i+1];j++)
+            X[j+c*N] = freq[j+c*N]*g;
+      }
+   } while (++c<C);
+}
+
+#endif /* FIXED_POINT */
+
+/* De-normalise the energy to produce the synthesis from the unit-energy bands */
+void denormalise_bands(const CELTMode *m, const celt_norm * OPUS_RESTRICT X, celt_sig * OPUS_RESTRICT freq, const celt_ener *bandE, int end, int C, int M)
+{
+   int i, c, N;
+   const opus_int16 *eBands = m->eBands;
+   N = M*m->shortMdctSize;
+   celt_assert2(C<=2, "denormalise_bands() not implemented for >2 channels");
+   c=0; do {
+      celt_sig * OPUS_RESTRICT f;
+      const celt_norm * OPUS_RESTRICT x;
+      f = freq+c*N;
+      x = X+c*N;
+      for (i=0;i<end;i++)
+      {
+         int j, band_end;
+         opus_val32 g = SHR32(bandE[i+c*m->nbEBands],1);
+         j=M*eBands[i];
+         band_end = M*eBands[i+1];
+         do {
+            *f++ = SHL32(MULT16_32_Q15(*x, g),2);
+            x++;
+         } while (++j<band_end);
+      }
+      for (i=M*eBands[end];i<N;i++)
+         *f++ = 0;
+   } while (++c<C);
+}
+
+/* This prevents energy collapse for transients with multiple short MDCTs */
+void anti_collapse(const CELTMode *m, celt_norm *X_, unsigned char *collapse_masks, int LM, int C, int size,
+      int start, int end, opus_val16 *logE, opus_val16 *prev1logE,
+      opus_val16 *prev2logE, int *pulses, opus_uint32 seed)
+{
+   int c, i, j, k;
+   for (i=start;i<end;i++)
+   {
+      int N0;
+      opus_val16 thresh, sqrt_1;
+      int depth;
+#ifdef FIXED_POINT
+      int shift;
+      opus_val32 thresh32;
+#endif
+
+      N0 = m->eBands[i+1]-m->eBands[i];
+      /* depth in 1/8 bits */
+      depth = (1+pulses[i])/((m->eBands[i+1]-m->eBands[i])<<LM);
+
+#ifdef FIXED_POINT
+      thresh32 = SHR32(celt_exp2(-SHL16(depth, 10-BITRES)),1);
+      thresh = MULT16_32_Q15(QCONST16(0.5f, 15), MIN32(32767,thresh32));
+      {
+         opus_val32 t;
+         t = N0<<LM;
+         shift = celt_ilog2(t)>>1;
+         t = SHL32(t, (7-shift)<<1);
+         sqrt_1 = celt_rsqrt_norm(t);
+      }
+#else
+      thresh = .5f*celt_exp2(-.125f*depth);
+      sqrt_1 = celt_rsqrt(N0<<LM);
+#endif
+
+      c=0; do
+      {
+         celt_norm *X;
+         opus_val16 prev1;
+         opus_val16 prev2;
+         opus_val32 Ediff;
+         opus_val16 r;
+         int renormalize=0;
+         prev1 = prev1logE[c*m->nbEBands+i];
+         prev2 = prev2logE[c*m->nbEBands+i];
+         if (C==1)
+         {
+            prev1 = MAX16(prev1,prev1logE[m->nbEBands+i]);
+            prev2 = MAX16(prev2,prev2logE[m->nbEBands+i]);
+         }
+         Ediff = EXTEND32(logE[c*m->nbEBands+i])-EXTEND32(MIN16(prev1,prev2));
+         Ediff = MAX32(0, Ediff);
+
+#ifdef FIXED_POINT
+         if (Ediff < 16384)
+         {
+            opus_val32 r32 = SHR32(celt_exp2(-EXTRACT16(Ediff)),1);
+            r = 2*MIN16(16383,r32);
+         } else {
+            r = 0;
+         }
+         if (LM==3)
+            r = MULT16_16_Q14(23170, MIN32(23169, r));
+         r = SHR16(MIN16(thresh, r),1);
+         r = SHR32(MULT16_16_Q15(sqrt_1, r),shift);
+#else
+         /* r needs to be multiplied by 2 or 2*sqrt(2) depending on LM because
+            short blocks don't have the same energy as long */
+         r = 2.f*celt_exp2(-Ediff);
+         if (LM==3)
+            r *= 1.41421356f;
+         r = MIN16(thresh, r);
+         r = r*sqrt_1;
+#endif
+         X = X_+c*size+(m->eBands[i]<<LM);
+         for (k=0;k<1<<LM;k++)
+         {
+            /* Detect collapse */
+            if (!(collapse_masks[i*C+c]&1<<k))
+            {
+               /* Fill with noise */
+               for (j=0;j<N0;j++)
+               {
+                  seed = celt_lcg_rand(seed);
+                  X[(j<<LM)+k] = (seed&0x8000 ? r : -r);
+               }
+               renormalize = 1;
+            }
+         }
+         /* We just added some energy, so we need to renormalise */
+         if (renormalize)
+            renormalise_vector(X, N0<<LM, Q15ONE);
+      } while (++c<C);
+   }
+}
+
+static void intensity_stereo(const CELTMode *m, celt_norm *X, celt_norm *Y, const celt_ener *bandE, int bandID, int N)
+{
+   int i = bandID;
+   int j;
+   opus_val16 a1, a2;
+   opus_val16 left, right;
+   opus_val16 norm;
+#ifdef FIXED_POINT
+   int shift = celt_zlog2(MAX32(bandE[i], bandE[i+m->nbEBands]))-13;
+#endif
+   left = VSHR32(bandE[i],shift);
+   right = VSHR32(bandE[i+m->nbEBands],shift);
+   norm = EPSILON + celt_sqrt(EPSILON+MULT16_16(left,left)+MULT16_16(right,right));
+   a1 = DIV32_16(SHL32(EXTEND32(left),14),norm);
+   a2 = DIV32_16(SHL32(EXTEND32(right),14),norm);
+   for (j=0;j<N;j++)
+   {
+      celt_norm r, l;
+      l = X[j];
+      r = Y[j];
+      X[j] = MULT16_16_Q14(a1,l) + MULT16_16_Q14(a2,r);
+      /* Side is not encoded, no need to calculate */
+   }
+}
+
+static void stereo_split(celt_norm *X, celt_norm *Y, int N)
+{
+   int j;
+   for (j=0;j<N;j++)
+   {
+      celt_norm r, l;
+      l = MULT16_16_Q15(QCONST16(.70710678f,15), X[j]);
+      r = MULT16_16_Q15(QCONST16(.70710678f,15), Y[j]);
+      X[j] = l+r;
+      Y[j] = r-l;
+   }
+}
+
+static void stereo_merge(celt_norm *X, celt_norm *Y, opus_val16 mid, int N)
+{
+   int j;
+   opus_val32 xp=0, side=0;
+   opus_val32 El, Er;
+   opus_val16 mid2;
+#ifdef FIXED_POINT
+   int kl, kr;
+#endif
+   opus_val32 t, lgain, rgain;
+
+   /* Compute the norm of X+Y and X-Y as |X|^2 + |Y|^2 +/- sum(xy) */
+   for (j=0;j<N;j++)
+   {
+      xp = MAC16_16(xp, X[j], Y[j]);
+      side = MAC16_16(side, Y[j], Y[j]);
+   }
+   /* Compensating for the mid normalization */
+   xp = MULT16_32_Q15(mid, xp);
+   /* mid and side are in Q15, not Q14 like X and Y */
+   mid2 = SHR32(mid, 1);
+   El = MULT16_16(mid2, mid2) + side - 2*xp;
+   Er = MULT16_16(mid2, mid2) + side + 2*xp;
+   if (Er < QCONST32(6e-4f, 28) || El < QCONST32(6e-4f, 28))
+   {
+      for (j=0;j<N;j++)
+         Y[j] = X[j];
+      return;
+   }
+
+#ifdef FIXED_POINT
+   kl = celt_ilog2(El)>>1;
+   kr = celt_ilog2(Er)>>1;
+#endif
+   t = VSHR32(El, (kl-7)<<1);
+   lgain = celt_rsqrt_norm(t);
+   t = VSHR32(Er, (kr-7)<<1);
+   rgain = celt_rsqrt_norm(t);
+
+#ifdef FIXED_POINT
+   if (kl < 7)
+      kl = 7;
+   if (kr < 7)
+      kr = 7;
+#endif
+
+   for (j=0;j<N;j++)
+   {
+      celt_norm r, l;
+      /* Apply mid scaling (side is already scaled) */
+      l = MULT16_16_Q15(mid, X[j]);
+      r = Y[j];
+      X[j] = EXTRACT16(PSHR32(MULT16_16(lgain, SUB16(l,r)), kl+1));
+      Y[j] = EXTRACT16(PSHR32(MULT16_16(rgain, ADD16(l,r)), kr+1));
+   }
+}
+
+/* Decide whether we should spread the pulses in the current frame */
+int spreading_decision(const CELTMode *m, celt_norm *X, int *average,
+      int last_decision, int *hf_average, int *tapset_decision, int update_hf,
+      int end, int C, int M)
+{
+   int i, c, N0;
+   int sum = 0, nbBands=0;
+   const opus_int16 * OPUS_RESTRICT eBands = m->eBands;
+   int decision;
+   int hf_sum=0;
+
+   celt_assert(end>0);
+
+   N0 = M*m->shortMdctSize;
+
+   if (M*(eBands[end]-eBands[end-1]) <= 8)
+      return SPREAD_NONE;
+   c=0; do {
+      for (i=0;i<end;i++)
+      {
+         int j, N, tmp=0;
+         int tcount[3] = {0,0,0};
+         celt_norm * OPUS_RESTRICT x = X+M*eBands[i]+c*N0;
+         N = M*(eBands[i+1]-eBands[i]);
+         if (N<=8)
+            continue;
+         /* Compute rough CDF of |x[j]| */
+         for (j=0;j<N;j++)
+         {
+            opus_val32 x2N; /* Q13 */
+
+            x2N = MULT16_16(MULT16_16_Q15(x[j], x[j]), N);
+            if (x2N < QCONST16(0.25f,13))
+               tcount[0]++;
+            if (x2N < QCONST16(0.0625f,13))
+               tcount[1]++;
+            if (x2N < QCONST16(0.015625f,13))
+               tcount[2]++;
+         }
+
+         /* Only include four last bands (8 kHz and up) */
+         if (i>m->nbEBands-4)
+            hf_sum += 32*(tcount[1]+tcount[0])/N;
+         tmp = (2*tcount[2] >= N) + (2*tcount[1] >= N) + (2*tcount[0] >= N);
+         sum += tmp*256;
+         nbBands++;
+      }
+   } while (++c<C);
+
+   if (update_hf)
+   {
+      if (hf_sum)
+         hf_sum /= C*(4-m->nbEBands+end);
+      *hf_average = (*hf_average+hf_sum)>>1;
+      hf_sum = *hf_average;
+      if (*tapset_decision==2)
+         hf_sum += 4;
+      else if (*tapset_decision==0)
+         hf_sum -= 4;
+      if (hf_sum > 22)
+         *tapset_decision=2;
+      else if (hf_sum > 18)
+         *tapset_decision=1;
+      else
+         *tapset_decision=0;
+   }
+   /*printf("%d %d %d\n", hf_sum, *hf_average, *tapset_decision);*/
+   celt_assert(nbBands>0); /*M*(eBands[end]-eBands[end-1]) <= 8 assures this*/
+   sum /= nbBands;
+   /* Recursive averaging */
+   sum = (sum+*average)>>1;
+   *average = sum;
+   /* Hysteresis */
+   sum = (3*sum + (((3-last_decision)<<7) + 64) + 2)>>2;
+   if (sum < 80)
+   {
+      decision = SPREAD_AGGRESSIVE;
+   } else if (sum < 256)
+   {
+      decision = SPREAD_NORMAL;
+   } else if (sum < 384)
+   {
+      decision = SPREAD_LIGHT;
+   } else {
+      decision = SPREAD_NONE;
+   }
+#ifdef FUZZING
+   decision = rand()&0x3;
+   *tapset_decision=rand()%3;
+#endif
+   return decision;
+}
+
+#ifdef MEASURE_NORM_MSE
+
+float MSE[30] = {0};
+int nbMSEBands = 0;
+int MSECount[30] = {0};
+
+void dump_norm_mse(void)
+{
+   int i;
+   for (i=0;i<nbMSEBands;i++)
+   {
+      printf ("%g ", MSE[i]/MSECount[i]);
+   }
+   printf ("\n");
+}
+
+void measure_norm_mse(const CELTMode *m, float *X, float *X0, float *bandE, float *bandE0, int M, int N, int C)
+{
+   static int init = 0;
+   int i;
+   if (!init)
+   {
+      atexit(dump_norm_mse);
+      init = 1;
+   }
+   for (i=0;i<m->nbEBands;i++)
+   {
+      int j;
+      int c;
+      float g;
+      if (bandE0[i]<10 || (C==2 && bandE0[i+m->nbEBands]<1))
+         continue;
+      c=0; do {
+         g = bandE[i+c*m->nbEBands]/(1e-15+bandE0[i+c*m->nbEBands]);
+         for (j=M*m->eBands[i];j<M*m->eBands[i+1];j++)
+            MSE[i] += (g*X[j+c*N]-X0[j+c*N])*(g*X[j+c*N]-X0[j+c*N]);
+      } while (++c<C);
+      MSECount[i]+=C;
+   }
+   nbMSEBands = m->nbEBands;
+}
+
+#endif
+
+/* Indexing table for converting from natural Hadamard to ordery Hadamard
+   This is essentially a bit-reversed Gray, on top of which we've added
+   an inversion of the order because we want the DC at the end rather than
+   the beginning. The lines are for N=2, 4, 8, 16 */
+static const int ordery_table[] = {
+       1,  0,
+       3,  0,  2,  1,
+       7,  0,  4,  3,  6,  1,  5,  2,
+      15,  0,  8,  7, 12,  3, 11,  4, 14,  1,  9,  6, 13,  2, 10,  5,
+};
+
+static void deinterleave_hadamard(celt_norm *X, int N0, int stride, int hadamard)
+{
+   int i,j;
+   VARDECL(celt_norm, tmp);
+   int N;
+   SAVE_STACK;
+   N = N0*stride;
+   ALLOC(tmp, N, celt_norm);
+   celt_assert(stride>0);
+   if (hadamard)
+   {
+      const int *ordery = ordery_table+stride-2;
+      for (i=0;i<stride;i++)
+      {
+         for (j=0;j<N0;j++)
+            tmp[ordery[i]*N0+j] = X[j*stride+i];
+      }
+   } else {
+      for (i=0;i<stride;i++)
+         for (j=0;j<N0;j++)
+            tmp[i*N0+j] = X[j*stride+i];
+   }
+   for (j=0;j<N;j++)
+      X[j] = tmp[j];
+   RESTORE_STACK;
+}
+
+static void interleave_hadamard(celt_norm *X, int N0, int stride, int hadamard)
+{
+   int i,j;
+   VARDECL(celt_norm, tmp);
+   int N;
+   SAVE_STACK;
+   N = N0*stride;
+   ALLOC(tmp, N, celt_norm);
+   if (hadamard)
+   {
+      const int *ordery = ordery_table+stride-2;
+      for (i=0;i<stride;i++)
+         for (j=0;j<N0;j++)
+            tmp[j*stride+i] = X[ordery[i]*N0+j];
+   } else {
+      for (i=0;i<stride;i++)
+         for (j=0;j<N0;j++)
+            tmp[j*stride+i] = X[i*N0+j];
+   }
+   for (j=0;j<N;j++)
+      X[j] = tmp[j];
+   RESTORE_STACK;
+}
+
+void haar1(celt_norm *X, int N0, int stride)
+{
+   int i, j;
+   N0 >>= 1;
+   for (i=0;i<stride;i++)
+      for (j=0;j<N0;j++)
+      {
+         celt_norm tmp1, tmp2;
+         tmp1 = MULT16_16_Q15(QCONST16(.70710678f,15), X[stride*2*j+i]);
+         tmp2 = MULT16_16_Q15(QCONST16(.70710678f,15), X[stride*(2*j+1)+i]);
+         X[stride*2*j+i] = tmp1 + tmp2;
+         X[stride*(2*j+1)+i] = tmp1 - tmp2;
+      }
+}
+
+static int compute_qn(int N, int b, int offset, int pulse_cap, int stereo)
+{
+   static const opus_int16 exp2_table8[8] =
+      {16384, 17866, 19483, 21247, 23170, 25267, 27554, 30048};
+   int qn, qb;
+   int N2 = 2*N-1;
+   if (stereo && N==2)
+      N2--;
+   /* The upper limit ensures that in a stereo split with itheta==16384, we'll
+       always have enough bits left over to code at least one pulse in the
+       side; otherwise it would collapse, since it doesn't get folded. */
+   qb = IMIN(b-pulse_cap-(4<<BITRES), (b+N2*offset)/N2);
+
+   qb = IMIN(8<<BITRES, qb);
+
+   if (qb<(1<<BITRES>>1)) {
+      qn = 1;
+   } else {
+      qn = exp2_table8[qb&0x7]>>(14-(qb>>BITRES));
+      qn = (qn+1)>>1<<1;
+   }
+   celt_assert(qn <= 256);
+   return qn;
+}
+
+/* This function is responsible for encoding and decoding a band for both
+   the mono and stereo case. Even in the mono case, it can split the band
+   in two and transmit the energy difference with the two half-bands. It
+   can be called recursively so bands can end up being split in 8 parts. */
+static unsigned quant_band(int encode, const CELTMode *m, int i, celt_norm *X, celt_norm *Y,
+      int N, int b, int spread, int B, int intensity, int tf_change, celt_norm *lowband, ec_ctx *ec,
+      opus_int32 *remaining_bits, int LM, celt_norm *lowband_out, const celt_ener *bandE, int level,
+      opus_uint32 *seed, opus_val16 gain, celt_norm *lowband_scratch, int fill)
+{
+   const unsigned char *cache;
+   int q;
+   int curr_bits;
+   int stereo, split;
+   int imid=0, iside=0;
+   int N0=N;
+   int N_B=N;
+   int N_B0;
+   int B0=B;
+   int time_divide=0;
+   int recombine=0;
+   int inv = 0;
+   opus_val16 mid=0, side=0;
+   int longBlocks;
+   unsigned cm=0;
+#ifdef RESYNTH
+   int resynth = 1;
+#else
+   int resynth = !encode;
+#endif
+
+   longBlocks = B0==1;
+
+   N_B /= B;
+   N_B0 = N_B;
+
+   split = stereo = Y != NULL;
+
+   /* Special case for one sample */
+   if (N==1)
+   {
+      int c;
+      celt_norm *x = X;
+      c=0; do {
+         int sign=0;
+         if (*remaining_bits>=1<<BITRES)
+         {
+            if (encode)
+            {
+               sign = x[0]<0;
+               ec_enc_bits(ec, sign, 1);
+            } else {
+               sign = ec_dec_bits(ec, 1);
+            }
+            *remaining_bits -= 1<<BITRES;
+            b-=1<<BITRES;
+         }
+         if (resynth)
+            x[0] = sign ? -NORM_SCALING : NORM_SCALING;
+         x = Y;
+      } while (++c<1+stereo);
+      if (lowband_out)
+         lowband_out[0] = SHR16(X[0],4);
+      return 1;
+   }
+
+   if (!stereo && level == 0)
+   {
+      int k;
+      if (tf_change>0)
+         recombine = tf_change;
+      /* Band recombining to increase frequency resolution */
+
+      if (lowband && (recombine || ((N_B&1) == 0 && tf_change<0) || B0>1))
+      {
+         int j;
+         for (j=0;j<N;j++)
+            lowband_scratch[j] = lowband[j];
+         lowband = lowband_scratch;
+      }
+
+      for (k=0;k<recombine;k++)
+      {
+         static const unsigned char bit_interleave_table[16]={
+           0,1,1,1,2,3,3,3,2,3,3,3,2,3,3,3
+         };
+         if (encode)
+            haar1(X, N>>k, 1<<k);
+         if (lowband)
+            haar1(lowband, N>>k, 1<<k);
+         fill = bit_interleave_table[fill&0xF]|bit_interleave_table[fill>>4]<<2;
+      }
+      B>>=recombine;
+      N_B<<=recombine;
+
+      /* Increasing the time resolution */
+      while ((N_B&1) == 0 && tf_change<0)
+      {
+         if (encode)
+            haar1(X, N_B, B);
+         if (lowband)
+            haar1(lowband, N_B, B);
+         fill |= fill<<B;
+         B <<= 1;
+         N_B >>= 1;
+         time_divide++;
+         tf_change++;
+      }
+      B0=B;
+      N_B0 = N_B;
+
+      /* Reorganize the samples in time order instead of frequency order */
+      if (B0>1)
+      {
+         if (encode)
+            deinterleave_hadamard(X, N_B>>recombine, B0<<recombine, longBlocks);
+         if (lowband)
+            deinterleave_hadamard(lowband, N_B>>recombine, B0<<recombine, longBlocks);
+      }
+   }
+
+   /* If we need 1.5 more bit than we can produce, split the band in two. */
+   cache = m->cache.bits + m->cache.index[(LM+1)*m->nbEBands+i];
+   if (!stereo && LM != -1 && b > cache[cache[0]]+12 && N>2)
+   {
+      N >>= 1;
+      Y = X+N;
+      split = 1;
+      LM -= 1;
+      if (B==1)
+         fill = (fill&1)|(fill<<1);
+      B = (B+1)>>1;
+   }
+
+   if (split)
+   {
+      int qn;
+      int itheta=0;
+      int mbits, sbits, delta;
+      int qalloc;
+      int pulse_cap;
+      int offset;
+      int orig_fill;
+      opus_int32 tell;
+
+      /* Decide on the resolution to give to the split parameter theta */
+      pulse_cap = m->logN[i]+LM*(1<<BITRES);
+      offset = (pulse_cap>>1) - (stereo&&N==2 ? QTHETA_OFFSET_TWOPHASE : QTHETA_OFFSET);
+      qn = compute_qn(N, b, offset, pulse_cap, stereo);
+      if (stereo && i>=intensity)
+         qn = 1;
+      if (encode)
+      {
+         /* theta is the atan() of the ratio between the (normalized)
+            side and mid. With just that parameter, we can re-scale both
+            mid and side because we know that 1) they have unit norm and
+            2) they are orthogonal. */
+         itheta = stereo_itheta(X, Y, stereo, N);
+      }
+      tell = ec_tell_frac(ec);
+      if (qn!=1)
+      {
+         if (encode)
+            itheta = (itheta*qn+8192)>>14;
+
+         /* Entropy coding of the angle. We use a uniform pdf for the
+            time split, a step for stereo, and a triangular one for the rest. */
+         if (stereo && N>2)
+         {
+            int p0 = 3;
+            int x = itheta;
+            int x0 = qn/2;
+            int ft = p0*(x0+1) + x0;
+            /* Use a probability of p0 up to itheta=8192 and then use 1 after */
+            if (encode)
+            {
+               ec_encode(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft);
+            } else {
+               int fs;
+               fs=ec_decode(ec,ft);
+               if (fs<(x0+1)*p0)
+                  x=fs/p0;
+               else
+                  x=x0+1+(fs-(x0+1)*p0);
+               ec_dec_update(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft);
+               itheta = x;
+            }
+         } else if (B0>1 || stereo) {
+            /* Uniform pdf */
+            if (encode)
+               ec_enc_uint(ec, itheta, qn+1);
+            else
+               itheta = ec_dec_uint(ec, qn+1);
+         } else {
+            int fs=1, ft;
+            ft = ((qn>>1)+1)*((qn>>1)+1);
+            if (encode)
+            {
+               int fl;
+
+               fs = itheta <= (qn>>1) ? itheta + 1 : qn + 1 - itheta;
+               fl = itheta <= (qn>>1) ? itheta*(itheta + 1)>>1 :
+                ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1);
+
+               ec_encode(ec, fl, fl+fs, ft);
+            } else {
+               /* Triangular pdf */
+               int fl=0;
+               int fm;
+               fm = ec_decode(ec, ft);
+
+               if (fm < ((qn>>1)*((qn>>1) + 1)>>1))
+               {
+                  itheta = (isqrt32(8*(opus_uint32)fm + 1) - 1)>>1;
+                  fs = itheta + 1;
+                  fl = itheta*(itheta + 1)>>1;
+               }
+               else
+               {
+                  itheta = (2*(qn + 1)
+                   - isqrt32(8*(opus_uint32)(ft - fm - 1) + 1))>>1;
+                  fs = qn + 1 - itheta;
+                  fl = ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1);
+               }
+
+               ec_dec_update(ec, fl, fl+fs, ft);
+            }
+         }
+         itheta = (opus_int32)itheta*16384/qn;
+         if (encode && stereo)
+         {
+            if (itheta==0)
+               intensity_stereo(m, X, Y, bandE, i, N);
+            else
+               stereo_split(X, Y, N);
+         }
+         /* NOTE: Renormalising X and Y *may* help fixed-point a bit at very high rate.
+                  Let's do that at higher complexity */
+      } else if (stereo) {
+         if (encode)
+         {
+            inv = itheta > 8192;
+            if (inv)
+            {
+               int j;
+               for (j=0;j<N;j++)
+                  Y[j] = -Y[j];
+            }
+            intensity_stereo(m, X, Y, bandE, i, N);
+         }
+         if (b>2<<BITRES && *remaining_bits > 2<<BITRES)
+         {
+            if (encode)
+               ec_enc_bit_logp(ec, inv, 2);
+            else
+               inv = ec_dec_bit_logp(ec, 2);
+         } else
+            inv = 0;
+         itheta = 0;
+      }
+      qalloc = ec_tell_frac(ec) - tell;
+      b -= qalloc;
+
+      orig_fill = fill;
+      if (itheta == 0)
+      {
+         imid = 32767;
+         iside = 0;
+         fill &= (1<<B)-1;
+         delta = -16384;
+      } else if (itheta == 16384)
+      {
+         imid = 0;
+         iside = 32767;
+         fill &= ((1<<B)-1)<<B;
+         delta = 16384;
+      } else {
+         imid = bitexact_cos(itheta);
+         iside = bitexact_cos(16384-itheta);
+         /* This is the mid vs side allocation that minimizes squared error
+            in that band. */
+         delta = FRAC_MUL16((N-1)<<7,bitexact_log2tan(iside,imid));
+      }
+
+#ifdef FIXED_POINT
+      mid = imid;
+      side = iside;
+#else
+      mid = (1.f/32768)*imid;
+      side = (1.f/32768)*iside;
+#endif
+
+      /* This is a special case for N=2 that only works for stereo and takes
+         advantage of the fact that mid and side are orthogonal to encode
+         the side with just one bit. */
+      if (N==2 && stereo)
+      {
+         int c;
+         int sign=0;
+         celt_norm *x2, *y2;
+         mbits = b;
+         sbits = 0;
+         /* Only need one bit for the side */
+         if (itheta != 0 && itheta != 16384)
+            sbits = 1<<BITRES;
+         mbits -= sbits;
+         c = itheta > 8192;
+         *remaining_bits -= qalloc+sbits;
+
+         x2 = c ? Y : X;
+         y2 = c ? X : Y;
+         if (sbits)
+         {
+            if (encode)
+            {
+               /* Here we only need to encode a sign for the side */
+               sign = x2[0]*y2[1] - x2[1]*y2[0] < 0;
+               ec_enc_bits(ec, sign, 1);
+            } else {
+               sign = ec_dec_bits(ec, 1);
+            }
+         }
+         sign = 1-2*sign;
+         /* We use orig_fill here because we want to fold the side, but if
+             itheta==16384, we'll have cleared the low bits of fill. */
+         cm = quant_band(encode, m, i, x2, NULL, N, mbits, spread, B, intensity, tf_change, lowband, ec, remaining_bits, LM, lowband_out, NULL, level, seed, gain, lowband_scratch, orig_fill);
+         /* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse),
+             and there's no need to worry about mixing with the other channel. */
+         y2[0] = -sign*x2[1];
+         y2[1] = sign*x2[0];
+         if (resynth)
+         {
+            celt_norm tmp;
+            X[0] = MULT16_16_Q15(mid, X[0]);
+            X[1] = MULT16_16_Q15(mid, X[1]);
+            Y[0] = MULT16_16_Q15(side, Y[0]);
+            Y[1] = MULT16_16_Q15(side, Y[1]);
+            tmp = X[0];
+            X[0] = SUB16(tmp,Y[0]);
+            Y[0] = ADD16(tmp,Y[0]);
+            tmp = X[1];
+            X[1] = SUB16(tmp,Y[1]);
+            Y[1] = ADD16(tmp,Y[1]);
+         }
+      } else {
+         /* "Normal" split code */
+         celt_norm *next_lowband2=NULL;
+         celt_norm *next_lowband_out1=NULL;
+         int next_level=0;
+         opus_int32 rebalance;
+
+         /* Give more bits to low-energy MDCTs than they would otherwise deserve */
+         if (B0>1 && !stereo && (itheta&0x3fff))
+         {
+            if (itheta > 8192)
+               /* Rough approximation for pre-echo masking */
+               delta -= delta>>(4-LM);
+            else
+               /* Corresponds to a forward-masking slope of 1.5 dB per 10 ms */
+               delta = IMIN(0, delta + (N<<BITRES>>(5-LM)));
+         }
+         mbits = IMAX(0, IMIN(b, (b-delta)/2));
+         sbits = b-mbits;
+         *remaining_bits -= qalloc;
+
+         if (lowband && !stereo)
+            next_lowband2 = lowband+N; /* >32-bit split case */
+
+         /* Only stereo needs to pass on lowband_out. Otherwise, it's
+            handled at the end */
+         if (stereo)
+            next_lowband_out1 = lowband_out;
+         else
+            next_level = level+1;
+
+         rebalance = *remaining_bits;
+         if (mbits >= sbits)
+         {
+            /* In stereo mode, we do not apply a scaling to the mid because we need the normalized
+               mid for folding later */
+            cm = quant_band(encode, m, i, X, NULL, N, mbits, spread, B, intensity, tf_change,
+                  lowband, ec, remaining_bits, LM, next_lowband_out1,
+                  NULL, next_level, seed, stereo ? Q15ONE : MULT16_16_P15(gain,mid), lowband_scratch, fill);
+            rebalance = mbits - (rebalance-*remaining_bits);
+            if (rebalance > 3<<BITRES && itheta!=0)
+               sbits += rebalance - (3<<BITRES);
+
+            /* For a stereo split, the high bits of fill are always zero, so no
+               folding will be done to the side. */
+            cm |= quant_band(encode, m, i, Y, NULL, N, sbits, spread, B, intensity, tf_change,
+                  next_lowband2, ec, remaining_bits, LM, NULL,
+                  NULL, next_level, seed, MULT16_16_P15(gain,side), NULL, fill>>B)<<((B0>>1)&(stereo-1));
+         } else {
+            /* For a stereo split, the high bits of fill are always zero, so no
+               folding will be done to the side. */
+            cm = quant_band(encode, m, i, Y, NULL, N, sbits, spread, B, intensity, tf_change,
+                  next_lowband2, ec, remaining_bits, LM, NULL,
+                  NULL, next_level, seed, MULT16_16_P15(gain,side), NULL, fill>>B)<<((B0>>1)&(stereo-1));
+            rebalance = sbits - (rebalance-*remaining_bits);
+            if (rebalance > 3<<BITRES && itheta!=16384)
+               mbits += rebalance - (3<<BITRES);
+            /* In stereo mode, we do not apply a scaling to the mid because we need the normalized
+               mid for folding later */
+            cm |= quant_band(encode, m, i, X, NULL, N, mbits, spread, B, intensity, tf_change,
+                  lowband, ec, remaining_bits, LM, next_lowband_out1,
+                  NULL, next_level, seed, stereo ? Q15ONE : MULT16_16_P15(gain,mid), lowband_scratch, fill);
+         }
+      }
+
+   } else {
+      /* This is the basic no-split case */
+      q = bits2pulses(m, i, LM, b);
+      curr_bits = pulses2bits(m, i, LM, q);
+      *remaining_bits -= curr_bits;
+
+      /* Ensures we can never bust the budget */
+      while (*remaining_bits < 0 && q > 0)
+      {
+         *remaining_bits += curr_bits;
+         q--;
+         curr_bits = pulses2bits(m, i, LM, q);
+         *remaining_bits -= curr_bits;
+      }
+
+      if (q!=0)
+      {
+         int K = get_pulses(q);
+
+         /* Finally do the actual quantization */
+         if (encode)
+         {
+            cm = alg_quant(X, N, K, spread, B, ec
+#ifdef RESYNTH
+                 , gain
+#endif
+                 );
+         } else {
+            cm = alg_unquant(X, N, K, spread, B, ec, gain);
+         }
+      } else {
+         /* If there's no pulse, fill the band anyway */
+         int j;
+         if (resynth)
+         {
+            unsigned cm_mask;
+            /*B can be as large as 16, so this shift might overflow an int on a
+               16-bit platform; use a long to get defined behavior.*/
+            cm_mask = (unsigned)(1UL<<B)-1;
+            fill &= cm_mask;
+            if (!fill)
+            {
+               for (j=0;j<N;j++)
+                  X[j] = 0;
+            } else {
+               if (lowband == NULL)
+               {
+                  /* Noise */
+                  for (j=0;j<N;j++)
+                  {
+                     *seed = celt_lcg_rand(*seed);
+                     X[j] = (celt_norm)((opus_int32)*seed>>20);
+                  }
+                  cm = cm_mask;
+               } else {
+                  /* Folded spectrum */
+                  for (j=0;j<N;j++)
+                  {
+                     opus_val16 tmp;
+                     *seed = celt_lcg_rand(*seed);
+                     /* About 48 dB below the "normal" folding level */
+                     tmp = QCONST16(1.0f/256, 10);
+                     tmp = (*seed)&0x8000 ? tmp : -tmp;
+                     X[j] = lowband[j]+tmp;
+                  }
+                  cm = fill;
+               }
+               renormalise_vector(X, N, gain);
+            }
+         }
+      }
+   }
+
+   /* This code is used by the decoder and by the resynthesis-enabled encoder */
+   if (resynth)
+   {
+      if (stereo)
+      {
+         if (N!=2)
+            stereo_merge(X, Y, mid, N);
+         if (inv)
+         {
+            int j;
+            for (j=0;j<N;j++)
+               Y[j] = -Y[j];
+         }
+      } else if (level == 0)
+      {
+         int k;
+
+         /* Undo the sample reorganization going from time order to frequency order */
+         if (B0>1)
+            interleave_hadamard(X, N_B>>recombine, B0<<recombine, longBlocks);
+
+         /* Undo time-freq changes that we did earlier */
+         N_B = N_B0;
+         B = B0;
+         for (k=0;k<time_divide;k++)
+         {
+            B >>= 1;
+            N_B <<= 1;
+            cm |= cm>>B;
+            haar1(X, N_B, B);
+         }
+
+         for (k=0;k<recombine;k++)
+         {
+            static const unsigned char bit_deinterleave_table[16]={
+              0x00,0x03,0x0C,0x0F,0x30,0x33,0x3C,0x3F,
+              0xC0,0xC3,0xCC,0xCF,0xF0,0xF3,0xFC,0xFF
+            };
+            cm = bit_deinterleave_table[cm];
+            haar1(X, N0>>k, 1<<k);
+         }
+         B<<=recombine;
+
+         /* Scale output for later folding */
+         if (lowband_out)
+         {
+            int j;
+            opus_val16 n;
+            n = celt_sqrt(SHL32(EXTEND32(N0),22));
+            for (j=0;j<N0;j++)
+               lowband_out[j] = MULT16_16_Q15(n,X[j]);
+         }
+         cm &= (1<<B)-1;
+      }
+   }
+   return cm;
+}
+
+void quant_all_bands(int encode, const CELTMode *m, int start, int end,
+      celt_norm *X_, celt_norm *Y_, unsigned char *collapse_masks, const celt_ener *bandE, int *pulses,
+      int shortBlocks, int spread, int dual_stereo, int intensity, int *tf_res,
+      opus_int32 total_bits, opus_int32 balance, ec_ctx *ec, int LM, int codedBands, opus_uint32 *seed)
+{
+   int i;
+   opus_int32 remaining_bits;
+   const opus_int16 * OPUS_RESTRICT eBands = m->eBands;
+   celt_norm * OPUS_RESTRICT norm, * OPUS_RESTRICT norm2;
+   VARDECL(celt_norm, _norm);
+   VARDECL(celt_norm, lowband_scratch);
+   int B;
+   int M;
+   int lowband_offset;
+   int update_lowband = 1;
+   int C = Y_ != NULL ? 2 : 1;
+#ifdef RESYNTH
+   int resynth = 1;
+#else
+   int resynth = !encode;
+#endif
+   SAVE_STACK;
+
+   M = 1<<LM;
+   B = shortBlocks ? M : 1;
+   ALLOC(_norm, C*M*eBands[m->nbEBands], celt_norm);
+   ALLOC(lowband_scratch, M*(eBands[m->nbEBands]-eBands[m->nbEBands-1]), celt_norm);
+   norm = _norm;
+   norm2 = norm + M*eBands[m->nbEBands];
+
+   lowband_offset = 0;
+   for (i=start;i<end;i++)
+   {
+      opus_int32 tell;
+      int b;
+      int N;
+      opus_int32 curr_balance;
+      int effective_lowband=-1;
+      celt_norm * OPUS_RESTRICT X, * OPUS_RESTRICT Y;
+      int tf_change=0;
+      unsigned x_cm;
+      unsigned y_cm;
+
+      X = X_+M*eBands[i];
+      if (Y_!=NULL)
+         Y = Y_+M*eBands[i];
+      else
+         Y = NULL;
+      N = M*eBands[i+1]-M*eBands[i];
+      tell = ec_tell_frac(ec);
+
+      /* Compute how many bits we want to allocate to this band */
+      if (i != start)
+         balance -= tell;
+      remaining_bits = total_bits-tell-1;
+      if (i <= codedBands-1)
+      {
+         curr_balance = balance / IMIN(3, codedBands-i);
+         b = IMAX(0, IMIN(16383, IMIN(remaining_bits+1,pulses[i]+curr_balance)));
+      } else {
+         b = 0;
+      }
+
+      if (resynth && M*eBands[i]-N >= M*eBands[start] && (update_lowband || lowband_offset==0))
+            lowband_offset = i;
+
+      tf_change = tf_res[i];
+      if (i>=m->effEBands)
+      {
+         X=norm;
+         if (Y_!=NULL)
+            Y = norm;
+      }
+
+      /* Get a conservative estimate of the collapse_mask's for the bands we're
+          going to be folding from. */
+      if (lowband_offset != 0 && (spread!=SPREAD_AGGRESSIVE || B>1 || tf_change<0))
+      {
+         int fold_start;
+         int fold_end;
+         int fold_i;
+         /* This ensures we never repeat spectral content within one band */
+         effective_lowband = IMAX(M*eBands[start], M*eBands[lowband_offset]-N);
+         fold_start = lowband_offset;
+         while(M*eBands[--fold_start] > effective_lowband);
+         fold_end = lowband_offset-1;
+         while(M*eBands[++fold_end] < effective_lowband+N);
+         x_cm = y_cm = 0;
+         fold_i = fold_start; do {
+           x_cm |= collapse_masks[fold_i*C+0];
+           y_cm |= collapse_masks[fold_i*C+C-1];
+         } while (++fold_i<fold_end);
+      }
+      /* Otherwise, we'll be using the LCG to fold, so all blocks will (almost
+          always) be non-zero.*/
+      else
+         x_cm = y_cm = (1<<B)-1;
+
+      if (dual_stereo && i==intensity)
+      {
+         int j;
+
+         /* Switch off dual stereo to do intensity */
+         dual_stereo = 0;
+         if (resynth)
+            for (j=M*eBands[start];j<M*eBands[i];j++)
+               norm[j] = HALF32(norm[j]+norm2[j]);
+      }
+      if (dual_stereo)
+      {
+         x_cm = quant_band(encode, m, i, X, NULL, N, b/2, spread, B, intensity, tf_change,
+               effective_lowband != -1 ? norm+effective_lowband : NULL, ec, &remaining_bits, LM,
+               norm+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, x_cm);
+         y_cm = quant_band(encode, m, i, Y, NULL, N, b/2, spread, B, intensity, tf_change,
+               effective_lowband != -1 ? norm2+effective_lowband : NULL, ec, &remaining_bits, LM,
+               norm2+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, y_cm);
+      } else {
+         x_cm = quant_band(encode, m, i, X, Y, N, b, spread, B, intensity, tf_change,
+               effective_lowband != -1 ? norm+effective_lowband : NULL, ec, &remaining_bits, LM,
+               norm+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, x_cm|y_cm);
+         y_cm = x_cm;
+      }
+      collapse_masks[i*C+0] = (unsigned char)x_cm;
+      collapse_masks[i*C+C-1] = (unsigned char)y_cm;
+      balance += pulses[i] + tell;
+
+      /* Update the folding position only as long as we have 1 bit/sample depth */
+      update_lowband = b>(N<<BITRES);
+   }
+   RESTORE_STACK;
+}
+
diff --git a/lib/rbcodec/codecs/libopus/celt/bands.h b/lib/rbcodec/codecs/libopus/celt/bands.h
new file mode 100644
index 0000000000..9ff8ffd7ba
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/bands.h
@@ -0,0 +1,95 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2009 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef BANDS_H
+#define BANDS_H
+
+#include "arch.h"
+#include "modes.h"
+#include "entenc.h"
+#include "entdec.h"
+#include "rate.h"
+
+/** Compute the amplitude (sqrt energy) in each of the bands
+ * @param m Mode data
+ * @param X Spectrum
+ * @param bands Square root of the energy for each band (returned)
+ */
+void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M);
+
+/*void compute_noise_energies(const CELTMode *m, const celt_sig *X, const opus_val16 *tonality, celt_ener *bandE);*/
+
+/** Normalise each band of X such that the energy in each band is
+    equal to 1
+ * @param m Mode data
+ * @param X Spectrum (returned normalised)
+ * @param bands Square root of the energy for each band
+ */
+void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M);
+
+/** Denormalise each band of X to restore full amplitude
+ * @param m Mode data
+ * @param X Spectrum (returned de-normalised)
+ * @param bands Square root of the energy for each band
+ */
+void denormalise_bands(const CELTMode *m, const celt_norm * OPUS_RESTRICT X, celt_sig * OPUS_RESTRICT freq, const celt_ener *bandE, int end, int C, int M);
+
+#define SPREAD_NONE       (0)
+#define SPREAD_LIGHT      (1)
+#define SPREAD_NORMAL     (2)
+#define SPREAD_AGGRESSIVE (3)
+
+int spreading_decision(const CELTMode *m, celt_norm *X, int *average,
+      int last_decision, int *hf_average, int *tapset_decision, int update_hf,
+      int end, int C, int M);
+
+#ifdef MEASURE_NORM_MSE
+void measure_norm_mse(const CELTMode *m, float *X, float *X0, float *bandE, float *bandE0, int M, int N, int C);
+#endif
+
+void haar1(celt_norm *X, int N0, int stride);
+
+/** Quantisation/encoding of the residual spectrum
+ * @param m Mode data
+ * @param X Residual (normalised)
+ * @param total_bits Total number of bits that can be used for the frame (including the ones already spent)
+ * @param enc Entropy encoder
+ */
+void quant_all_bands(int encode, const CELTMode *m, int start, int end,
+      celt_norm * X, celt_norm * Y, unsigned char *collapse_masks, const celt_ener *bandE, int *pulses,
+      int time_domain, int fold, int dual_stereo, int intensity, int *tf_res,
+      opus_int32 total_bits, opus_int32 balance, ec_ctx *ec, int M, int codedBands, opus_uint32 *seed);
+
+void anti_collapse(const CELTMode *m, celt_norm *X_, unsigned char *collapse_masks, int LM, int C, int size,
+      int start, int end, opus_val16 *logE, opus_val16 *prev1logE,
+      opus_val16 *prev2logE, int *pulses, opus_uint32 seed);
+
+opus_uint32 celt_lcg_rand(opus_uint32 seed);
+
+#endif /* BANDS_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/celt.c b/lib/rbcodec/codecs/libopus/celt/celt.c
new file mode 100644
index 0000000000..8d42cc95b3
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/celt.c
@@ -0,0 +1,2870 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2010 Xiph.Org Foundation
+   Copyright (c) 2008 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#define CELT_C
+
+#include "os_support.h"
+#include "mdct.h"
+#include <math.h>
+#include "celt.h"
+#include "pitch.h"
+#include "bands.h"
+#include "modes.h"
+#include "entcode.h"
+#include "quant_bands.h"
+#include "rate.h"
+#include "stack_alloc.h"
+#include "mathops.h"
+#include "float_cast.h"
+#include <stdarg.h>
+#include "celt_lpc.h"
+#include "vq.h"
+
+#ifndef OPUS_VERSION
+#define OPUS_VERSION "unknown"
+#endif
+
+#ifdef CUSTOM_MODES
+#define OPUS_CUSTOM_NOSTATIC
+#else
+#define OPUS_CUSTOM_NOSTATIC static inline
+#endif
+
+static const unsigned char trim_icdf[11] = {126, 124, 119, 109, 87, 41, 19, 9, 4, 2, 0};
+/* Probs: NONE: 21.875%, LIGHT: 6.25%, NORMAL: 65.625%, AGGRESSIVE: 6.25% */
+static const unsigned char spread_icdf[4] = {25, 23, 2, 0};
+
+static const unsigned char tapset_icdf[3]={2,1,0};
+
+#ifdef CUSTOM_MODES
+static const unsigned char toOpusTable[20] = {
+      0xE0, 0xE8, 0xF0, 0xF8,
+      0xC0, 0xC8, 0xD0, 0xD8,
+      0xA0, 0xA8, 0xB0, 0xB8,
+      0x00, 0x00, 0x00, 0x00,
+      0x80, 0x88, 0x90, 0x98,
+};
+
+static const unsigned char fromOpusTable[16] = {
+      0x80, 0x88, 0x90, 0x98,
+      0x40, 0x48, 0x50, 0x58,
+      0x20, 0x28, 0x30, 0x38,
+      0x00, 0x08, 0x10, 0x18
+};
+
+static inline int toOpus(unsigned char c)
+{
+   int ret=0;
+   if (c<0xA0)
+      ret = toOpusTable[c>>3];
+   if (ret == 0)
+      return -1;
+   else
+      return ret|(c&0x7);
+}
+
+static inline int fromOpus(unsigned char c)
+{
+   if (c<0x80)
+      return -1;
+   else
+      return fromOpusTable[(c>>3)-16] | (c&0x7);
+}
+#endif /* CUSTOM_MODES */
+
+#define COMBFILTER_MAXPERIOD 1024
+#define COMBFILTER_MINPERIOD 15
+
+static int resampling_factor(opus_int32 rate)
+{
+   int ret;
+   switch (rate)
+   {
+   case 48000:
+      ret = 1;
+      break;
+   case 24000:
+      ret = 2;
+      break;
+   case 16000:
+      ret = 3;
+      break;
+   case 12000:
+      ret = 4;
+      break;
+   case 8000:
+      ret = 6;
+      break;
+   default:
+#ifndef CUSTOM_MODES
+      celt_assert(0);
+#endif
+      ret = 0;
+      break;
+   }
+   return ret;
+}
+
+/** Encoder state
+ @brief Encoder state
+ */
+struct OpusCustomEncoder {
+   const OpusCustomMode *mode;     /**< Mode used by the encoder */
+   int overlap;
+   int channels;
+   int stream_channels;
+
+   int force_intra;
+   int clip;
+   int disable_pf;
+   int complexity;
+   int upsample;
+   int start, end;
+
+   opus_int32 bitrate;
+   int vbr;
+   int signalling;
+   int constrained_vbr;      /* If zero, VBR can do whatever it likes with the rate */
+   int loss_rate;
+   int lsb_depth;
+
+   /* Everything beyond this point gets cleared on a reset */
+#define ENCODER_RESET_START rng
+
+   opus_uint32 rng;
+   int spread_decision;
+   opus_val32 delayedIntra;
+   int tonal_average;
+   int lastCodedBands;
+   int hf_average;
+   int tapset_decision;
+
+   int prefilter_period;
+   opus_val16 prefilter_gain;
+   int prefilter_tapset;
+#ifdef RESYNTH
+   int prefilter_period_old;
+   opus_val16 prefilter_gain_old;
+   int prefilter_tapset_old;
+#endif
+   int consec_transient;
+
+   opus_val32 preemph_memE[2];
+   opus_val32 preemph_memD[2];
+
+   /* VBR-related parameters */
+   opus_int32 vbr_reservoir;
+   opus_int32 vbr_drift;
+   opus_int32 vbr_offset;
+   opus_int32 vbr_count;
+
+#ifdef RESYNTH
+   celt_sig syn_mem[2][2*MAX_PERIOD];
+#endif
+
+   celt_sig in_mem[1]; /* Size = channels*mode->overlap */
+   /* celt_sig prefilter_mem[],  Size = channels*COMBFILTER_PERIOD */
+   /* celt_sig overlap_mem[],  Size = channels*mode->overlap */
+   /* opus_val16 oldEBands[], Size = 2*channels*mode->nbEBands */
+};
+
+int celt_encoder_get_size(int channels)
+{
+   CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
+   return opus_custom_encoder_get_size(mode, channels);
+}
+
+OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_get_size(const CELTMode *mode, int channels)
+{
+   int size = sizeof(struct CELTEncoder)
+         + (2*channels*mode->overlap-1)*sizeof(celt_sig)
+         + channels*COMBFILTER_MAXPERIOD*sizeof(celt_sig)
+         + 3*channels*mode->nbEBands*sizeof(opus_val16);
+   return size;
+}
+
+#ifdef CUSTOM_MODES
+CELTEncoder *opus_custom_encoder_create(const CELTMode *mode, int channels, int *error)
+{
+   int ret;
+   CELTEncoder *st = (CELTEncoder *)opus_alloc(opus_custom_encoder_get_size(mode, channels));
+   /* init will handle the NULL case */
+   ret = opus_custom_encoder_init(st, mode, channels);
+   if (ret != OPUS_OK)
+   {
+      opus_custom_encoder_destroy(st);
+      st = NULL;
+   }
+   if (error)
+      *error = ret;
+   return st;
+}
+#endif /* CUSTOM_MODES */
+
+int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels)
+{
+   int ret;
+   ret = opus_custom_encoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels);
+   if (ret != OPUS_OK)
+      return ret;
+   st->upsample = resampling_factor(sampling_rate);
+   return OPUS_OK;
+}
+
+OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_init(CELTEncoder *st, const CELTMode *mode, int channels)
+{
+   if (channels < 0 || channels > 2)
+      return OPUS_BAD_ARG;
+
+   if (st==NULL || mode==NULL)
+      return OPUS_ALLOC_FAIL;
+
+   OPUS_CLEAR((char*)st, opus_custom_encoder_get_size(mode, channels));
+
+   st->mode = mode;
+   st->overlap = mode->overlap;
+   st->stream_channels = st->channels = channels;
+
+   st->upsample = 1;
+   st->start = 0;
+   st->end = st->mode->effEBands;
+   st->signalling = 1;
+
+   st->constrained_vbr = 1;
+   st->clip = 1;
+
+   st->bitrate = OPUS_BITRATE_MAX;
+   st->vbr = 0;
+   st->force_intra  = 0;
+   st->complexity = 5;
+   st->lsb_depth=24;
+
+   opus_custom_encoder_ctl(st, OPUS_RESET_STATE);
+
+   return OPUS_OK;
+}
+
+#ifdef CUSTOM_MODES
+void opus_custom_encoder_destroy(CELTEncoder *st)
+{
+   opus_free(st);
+}
+#endif /* CUSTOM_MODES */
+
+static inline opus_val16 SIG2WORD16(celt_sig x)
+{
+#ifdef FIXED_POINT
+   x = PSHR32(x, SIG_SHIFT);
+   x = MAX32(x, -32768);
+   x = MIN32(x, 32767);
+   return EXTRACT16(x);
+#else
+   return (opus_val16)x;
+#endif
+}
+
+static int transient_analysis(const opus_val32 * OPUS_RESTRICT in, int len, int C,
+                              int overlap)
+{
+   int i;
+   VARDECL(opus_val16, tmp);
+   opus_val32 mem0=0,mem1=0;
+   int is_transient = 0;
+   int block;
+   int N;
+   VARDECL(opus_val16, bins);
+   SAVE_STACK;
+   ALLOC(tmp, len, opus_val16);
+
+   block = overlap/2;
+   N=len/block;
+   ALLOC(bins, N, opus_val16);
+   if (C==1)
+   {
+      for (i=0;i<len;i++)
+         tmp[i] = SHR32(in[i],SIG_SHIFT);
+   } else {
+      for (i=0;i<len;i++)
+         tmp[i] = SHR32(ADD32(in[i],in[i+len]), SIG_SHIFT+1);
+   }
+
+   /* High-pass filter: (1 - 2*z^-1 + z^-2) / (1 - z^-1 + .5*z^-2) */
+   for (i=0;i<len;i++)
+   {
+      opus_val32 x,y;
+      x = tmp[i];
+      y = ADD32(mem0, x);
+#ifdef FIXED_POINT
+      mem0 = mem1 + y - SHL32(x,1);
+      mem1 = x - SHR32(y,1);
+#else
+      mem0 = mem1 + y - 2*x;
+      mem1 = x - .5f*y;
+#endif
+      tmp[i] = EXTRACT16(SHR32(y,2));
+   }
+   /* First few samples are bad because we don't propagate the memory */
+   for (i=0;i<12;i++)
+      tmp[i] = 0;
+
+   for (i=0;i<N;i++)
+   {
+      int j;
+      opus_val16 max_abs=0;
+      for (j=0;j<block;j++)
+         max_abs = MAX16(max_abs, ABS16(tmp[i*block+j]));
+      bins[i] = max_abs;
+   }
+   for (i=0;i<N;i++)
+   {
+      int j;
+      int conseq=0;
+      opus_val16 t1, t2, t3;
+
+      t1 = MULT16_16_Q15(QCONST16(.15f, 15), bins[i]);
+      t2 = MULT16_16_Q15(QCONST16(.4f, 15), bins[i]);
+      t3 = MULT16_16_Q15(QCONST16(.15f, 15), bins[i]);
+      for (j=0;j<i;j++)
+      {
+         if (bins[j] < t1)
+            conseq++;
+         if (bins[j] < t2)
+            conseq++;
+         else
+            conseq = 0;
+      }
+      if (conseq>=3)
+         is_transient=1;
+      conseq = 0;
+      for (j=i+1;j<N;j++)
+      {
+         if (bins[j] < t3)
+            conseq++;
+         else
+            conseq = 0;
+      }
+      if (conseq>=7)
+         is_transient=1;
+   }
+   RESTORE_STACK;
+#ifdef FUZZING
+   is_transient = rand()&0x1;
+#endif
+   return is_transient;
+}
+
+/** Apply window and compute the MDCT for all sub-frames and
+    all channels in a frame */
+static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, celt_sig * OPUS_RESTRICT out, int C, int LM)
+{
+   if (C==1 && !shortBlocks)
+   {
+      const int overlap = OVERLAP(mode);
+      clt_mdct_forward(&mode->mdct, in, out, mode->window, overlap, mode->maxLM-LM, 1);
+   } else {
+      const int overlap = OVERLAP(mode);
+      int N = mode->shortMdctSize<<LM;
+      int B = 1;
+      int b, c;
+      if (shortBlocks)
+      {
+         N = mode->shortMdctSize;
+         B = shortBlocks;
+      }
+      c=0; do {
+         for (b=0;b<B;b++)
+         {
+            /* Interleaving the sub-frames while doing the MDCTs */
+            clt_mdct_forward(&mode->mdct, in+c*(B*N+overlap)+b*N, &out[b+c*N*B], mode->window, overlap, shortBlocks ? mode->maxLM : mode->maxLM-LM, B);
+         }
+      } while (++c<C);
+   }
+}
+
+/** Compute the IMDCT and apply window for all sub-frames and
+    all channels in a frame */
+static void compute_inv_mdcts(const CELTMode *mode, int shortBlocks, celt_sig *X,
+      celt_sig * OPUS_RESTRICT out_mem[],
+      celt_sig * OPUS_RESTRICT overlap_mem[], int C, int LM)
+{
+   int c;
+   const int N = mode->shortMdctSize<<LM;
+   const int overlap = OVERLAP(mode);
+   VARDECL(opus_val32, x);
+   SAVE_STACK;
+
+   ALLOC(x, N+overlap, opus_val32);
+   c=0; do {
+      int j;
+      int b;
+      int N2 = N;
+      int B = 1;
+
+      if (shortBlocks)
+      {
+         N2 = mode->shortMdctSize;
+         B = shortBlocks;
+      }
+      /* Prevents problems from the imdct doing the overlap-add */
+      OPUS_CLEAR(x, overlap);
+
+      for (b=0;b<B;b++)
+      {
+         /* IMDCT on the interleaved the sub-frames */
+         clt_mdct_backward(&mode->mdct, &X[b+c*N2*B], x+N2*b, mode->window, overlap, shortBlocks ? mode->maxLM : mode->maxLM-LM, B);
+      }
+
+      for (j=0;j<overlap;j++)
+         out_mem[c][j] = x[j] + overlap_mem[c][j];
+      for (;j<N;j++)
+         out_mem[c][j] = x[j];
+      for (j=0;j<overlap;j++)
+         overlap_mem[c][j] = x[N+j];
+   } while (++c<C);
+   RESTORE_STACK;
+}
+
+static void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, const opus_val16 *coef, celt_sig *mem)
+{
+   int c;
+   int count=0;
+   c=0; do {
+      int j;
+      celt_sig * OPUS_RESTRICT x;
+      opus_val16  * OPUS_RESTRICT y;
+      celt_sig m = mem[c];
+      x =in[c];
+      y = pcm+c;
+      for (j=0;j<N;j++)
+      {
+         celt_sig tmp = *x + m;
+         m = MULT16_32_Q15(coef[0], tmp)
+           - MULT16_32_Q15(coef[1], *x);
+         tmp = SHL32(MULT16_32_Q15(coef[3], tmp), 2);
+         x++;
+         /* Technically the store could be moved outside of the if because
+            the stores we don't want will just be overwritten */
+         if (count==0)
+            *y = SCALEOUT(SIG2WORD16(tmp));
+         if (++count==downsample)
+         {
+            y+=C;
+            count=0;
+         }
+      }
+      mem[c] = m;
+   } while (++c<C);
+}
+
+static void comb_filter(opus_val32 *y, opus_val32 *x, int T0, int T1, int N,
+      opus_val16 g0, opus_val16 g1, int tapset0, int tapset1,
+      const opus_val16 *window, int overlap)
+{
+   int i;
+   /* printf ("%d %d %f %f\n", T0, T1, g0, g1); */
+   opus_val16 g00, g01, g02, g10, g11, g12;
+   static const opus_val16 gains[3][3] = {
+         {QCONST16(0.3066406250f, 15), QCONST16(0.2170410156f, 15), QCONST16(0.1296386719f, 15)},
+         {QCONST16(0.4638671875f, 15), QCONST16(0.2680664062f, 15), QCONST16(0.f, 15)},
+         {QCONST16(0.7998046875f, 15), QCONST16(0.1000976562f, 15), QCONST16(0.f, 15)}};
+   g00 = MULT16_16_Q15(g0, gains[tapset0][0]);
+   g01 = MULT16_16_Q15(g0, gains[tapset0][1]);
+   g02 = MULT16_16_Q15(g0, gains[tapset0][2]);
+   g10 = MULT16_16_Q15(g1, gains[tapset1][0]);
+   g11 = MULT16_16_Q15(g1, gains[tapset1][1]);
+   g12 = MULT16_16_Q15(g1, gains[tapset1][2]);
+   for (i=0;i<overlap;i++)
+   {
+      opus_val16 f;
+      f = MULT16_16_Q15(window[i],window[i]);
+      y[i] = x[i]
+               + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g00),x[i-T0])
+               + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g01),x[i-T0-1])
+               + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g01),x[i-T0+1])
+               + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g02),x[i-T0-2])
+               + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g02),x[i-T0+2])
+               + MULT16_32_Q15(MULT16_16_Q15(f,g10),x[i-T1])
+               + MULT16_32_Q15(MULT16_16_Q15(f,g11),x[i-T1-1])
+               + MULT16_32_Q15(MULT16_16_Q15(f,g11),x[i-T1+1])
+               + MULT16_32_Q15(MULT16_16_Q15(f,g12),x[i-T1-2])
+               + MULT16_32_Q15(MULT16_16_Q15(f,g12),x[i-T1+2]);
+
+   }
+   for (i=overlap;i<N;i++)
+      y[i] = x[i]
+               + MULT16_32_Q15(g10,x[i-T1])
+               + MULT16_32_Q15(g11,x[i-T1-1])
+               + MULT16_32_Q15(g11,x[i-T1+1])
+               + MULT16_32_Q15(g12,x[i-T1-2])
+               + MULT16_32_Q15(g12,x[i-T1+2]);
+}
+
+static const signed char tf_select_table[4][8] = {
+      {0, -1, 0, -1,    0,-1, 0,-1},
+      {0, -1, 0, -2,    1, 0, 1,-1},
+      {0, -2, 0, -3,    2, 0, 1,-1},
+      {0, -2, 0, -3,    3, 0, 1,-1},
+};
+
+static opus_val32 l1_metric(const celt_norm *tmp, int N, int LM, int width)
+{
+   int i, j;
+   static const opus_val16 sqrtM_1[4] = {Q15ONE, QCONST16(.70710678f,15), QCONST16(0.5f,15), QCONST16(0.35355339f,15)};
+   opus_val32 L1;
+   opus_val16 bias;
+   L1=0;
+   for (i=0;i<1<<LM;i++)
+   {
+      opus_val32 L2 = 0;
+      for (j=0;j<N>>LM;j++)
+         L2 = MAC16_16(L2, tmp[(j<<LM)+i], tmp[(j<<LM)+i]);
+      L1 += celt_sqrt(L2);
+   }
+   L1 = MULT16_32_Q15(sqrtM_1[LM], L1);
+   if (width==1)
+      bias = QCONST16(.12f,15)*LM;
+   else if (width==2)
+      bias = QCONST16(.05f,15)*LM;
+   else
+      bias = QCONST16(.02f,15)*LM;
+   L1 = MAC16_32_Q15(L1, bias, L1);
+   return L1;
+}
+
+static int tf_analysis(const CELTMode *m, int len, int C, int isTransient,
+      int *tf_res, int nbCompressedBytes, celt_norm *X, int N0, int LM,
+      int *tf_sum)
+{
+   int i;
+   VARDECL(int, metric);
+   int cost0;
+   int cost1;
+   VARDECL(int, path0);
+   VARDECL(int, path1);
+   VARDECL(celt_norm, tmp);
+   int lambda;
+   int tf_select=0;
+   SAVE_STACK;
+
+   if (nbCompressedBytes<15*C)
+   {
+      *tf_sum = 0;
+      for (i=0;i<len;i++)
+         tf_res[i] = isTransient;
+      return 0;
+   }
+   if (nbCompressedBytes<40)
+      lambda = 12;
+   else if (nbCompressedBytes<60)
+      lambda = 6;
+   else if (nbCompressedBytes<100)
+      lambda = 4;
+   else
+      lambda = 3;
+
+   ALLOC(metric, len, int);
+   ALLOC(tmp, (m->eBands[len]-m->eBands[len-1])<<LM, celt_norm);
+   ALLOC(path0, len, int);
+   ALLOC(path1, len, int);
+
+   *tf_sum = 0;
+   for (i=0;i<len;i++)
+   {
+      int j, k, N;
+      opus_val32 L1, best_L1;
+      int best_level=0;
+      N = (m->eBands[i+1]-m->eBands[i])<<LM;
+      for (j=0;j<N;j++)
+         tmp[j] = X[j+(m->eBands[i]<<LM)];
+      /* Just add the right channel if we're in stereo */
+      if (C==2)
+         for (j=0;j<N;j++)
+            tmp[j] = ADD16(SHR16(tmp[j], 1),SHR16(X[N0+j+(m->eBands[i]<<LM)], 1));
+      L1 = l1_metric(tmp, N, isTransient ? LM : 0, N>>LM);
+      best_L1 = L1;
+      /*printf ("%f ", L1);*/
+      for (k=0;k<LM;k++)
+      {
+         int B;
+
+         if (isTransient)
+            B = (LM-k-1);
+         else
+            B = k+1;
+
+         if (isTransient)
+            haar1(tmp, N>>(LM-k), 1<<(LM-k));
+         else
+            haar1(tmp, N>>k, 1<<k);
+
+         L1 = l1_metric(tmp, N, B, N>>LM);
+
+         if (L1 < best_L1)
+         {
+            best_L1 = L1;
+            best_level = k+1;
+         }
+      }
+      /*printf ("%d ", isTransient ? LM-best_level : best_level);*/
+      if (isTransient)
+         metric[i] = best_level;
+      else
+         metric[i] = -best_level;
+      *tf_sum += metric[i];
+   }
+   /*printf("\n");*/
+   /* NOTE: Future optimized implementations could detect extreme transients and set
+      tf_select = 1 but so far we have not found a reliable way of making this useful */
+   tf_select = 0;
+
+   cost0 = 0;
+   cost1 = isTransient ? 0 : lambda;
+   /* Viterbi forward pass */
+   for (i=1;i<len;i++)
+   {
+      int curr0, curr1;
+      int from0, from1;
+
+      from0 = cost0;
+      from1 = cost1 + lambda;
+      if (from0 < from1)
+      {
+         curr0 = from0;
+         path0[i]= 0;
+      } else {
+         curr0 = from1;
+         path0[i]= 1;
+      }
+
+      from0 = cost0 + lambda;
+      from1 = cost1;
+      if (from0 < from1)
+      {
+         curr1 = from0;
+         path1[i]= 0;
+      } else {
+         curr1 = from1;
+         path1[i]= 1;
+      }
+      cost0 = curr0 + abs(metric[i]-tf_select_table[LM][4*isTransient+2*tf_select+0]);
+      cost1 = curr1 + abs(metric[i]-tf_select_table[LM][4*isTransient+2*tf_select+1]);
+   }
+   tf_res[len-1] = cost0 < cost1 ? 0 : 1;
+   /* Viterbi backward pass to check the decisions */
+   for (i=len-2;i>=0;i--)
+   {
+      if (tf_res[i+1] == 1)
+         tf_res[i] = path1[i+1];
+      else
+         tf_res[i] = path0[i+1];
+   }
+   RESTORE_STACK;
+#ifdef FUZZING
+   tf_select = rand()&0x1;
+   tf_res[0] = rand()&0x1;
+   for (i=1;i<len;i++)
+      tf_res[i] = tf_res[i-1] ^ ((rand()&0xF) == 0);
+#endif
+   return tf_select;
+}
+
+static void tf_encode(int start, int end, int isTransient, int *tf_res, int LM, int tf_select, ec_enc *enc)
+{
+   int curr, i;
+   int tf_select_rsv;
+   int tf_changed;
+   int logp;
+   opus_uint32 budget;
+   opus_uint32 tell;
+   budget = enc->storage*8;
+   tell = ec_tell(enc);
+   logp = isTransient ? 2 : 4;
+   /* Reserve space to code the tf_select decision. */
+   tf_select_rsv = LM>0 && tell+logp+1 <= budget;
+   budget -= tf_select_rsv;
+   curr = tf_changed = 0;
+   for (i=start;i<end;i++)
+   {
+      if (tell+logp<=budget)
+      {
+         ec_enc_bit_logp(enc, tf_res[i] ^ curr, logp);
+         tell = ec_tell(enc);
+         curr = tf_res[i];
+         tf_changed |= curr;
+      }
+      else
+         tf_res[i] = curr;
+      logp = isTransient ? 4 : 5;
+   }
+   /* Only code tf_select if it would actually make a difference. */
+   if (tf_select_rsv &&
+         tf_select_table[LM][4*isTransient+0+tf_changed]!=
+         tf_select_table[LM][4*isTransient+2+tf_changed])
+      ec_enc_bit_logp(enc, tf_select, 1);
+   else
+      tf_select = 0;
+   for (i=start;i<end;i++)
+      tf_res[i] = tf_select_table[LM][4*isTransient+2*tf_select+tf_res[i]];
+   /*printf("%d %d ", isTransient, tf_select); for(i=0;i<end;i++)printf("%d ", tf_res[i]);printf("\n");*/
+}
+
+static void tf_decode(int start, int end, int isTransient, int *tf_res, int LM, ec_dec *dec)
+{
+   int i, curr, tf_select;
+   int tf_select_rsv;
+   int tf_changed;
+   int logp;
+   opus_uint32 budget;
+   opus_uint32 tell;
+
+   budget = dec->storage*8;
+   tell = ec_tell(dec);
+   logp = isTransient ? 2 : 4;
+   tf_select_rsv = LM>0 && tell+logp+1<=budget;
+   budget -= tf_select_rsv;
+   tf_changed = curr = 0;
+   for (i=start;i<end;i++)
+   {
+      if (tell+logp<=budget)
+      {
+         curr ^= ec_dec_bit_logp(dec, logp);
+         tell = ec_tell(dec);
+         tf_changed |= curr;
+      }
+      tf_res[i] = curr;
+      logp = isTransient ? 4 : 5;
+   }
+   tf_select = 0;
+   if (tf_select_rsv &&
+     tf_select_table[LM][4*isTransient+0+tf_changed] !=
+     tf_select_table[LM][4*isTransient+2+tf_changed])
+   {
+      tf_select = ec_dec_bit_logp(dec, 1);
+   }
+   for (i=start;i<end;i++)
+   {
+      tf_res[i] = tf_select_table[LM][4*isTransient+2*tf_select+tf_res[i]];
+   }
+}
+
+static void init_caps(const CELTMode *m,int *cap,int LM,int C)
+{
+   int i;
+   for (i=0;i<m->nbEBands;i++)
+   {
+      int N;
+      N=(m->eBands[i+1]-m->eBands[i])<<LM;
+      cap[i] = (m->cache.caps[m->nbEBands*(2*LM+C-1)+i]+64)*C*N>>2;
+   }
+}
+
+static int alloc_trim_analysis(const CELTMode *m, const celt_norm *X,
+      const opus_val16 *bandLogE, int end, int LM, int C, int N0)
+{
+   int i;
+   opus_val32 diff=0;
+   int c;
+   int trim_index = 5;
+   if (C==2)
+   {
+      opus_val16 sum = 0; /* Q10 */
+      /* Compute inter-channel correlation for low frequencies */
+      for (i=0;i<8;i++)
+      {
+         int j;
+         opus_val32 partial = 0;
+         for (j=m->eBands[i]<<LM;j<m->eBands[i+1]<<LM;j++)
+            partial = MAC16_16(partial, X[j], X[N0+j]);
+         sum = ADD16(sum, EXTRACT16(SHR32(partial, 18)));
+      }
+      sum = MULT16_16_Q15(QCONST16(1.f/8, 15), sum);
+      /*printf ("%f\n", sum);*/
+      if (sum > QCONST16(.995f,10))
+         trim_index-=4;
+      else if (sum > QCONST16(.92f,10))
+         trim_index-=3;
+      else if (sum > QCONST16(.85f,10))
+         trim_index-=2;
+      else if (sum > QCONST16(.8f,10))
+         trim_index-=1;
+   }
+
+   /* Estimate spectral tilt */
+   c=0; do {
+      for (i=0;i<end-1;i++)
+      {
+         diff += bandLogE[i+c*m->nbEBands]*(opus_int32)(2+2*i-m->nbEBands);
+      }
+   } while (++c<C);
+   /* We divide by two here to avoid making the tilt larger for stereo as a
+      result of a bug in the loop above */
+   diff /= 2*C*(end-1);
+   /*printf("%f\n", diff);*/
+   if (diff > QCONST16(2.f, DB_SHIFT))
+      trim_index--;
+   if (diff > QCONST16(8.f, DB_SHIFT))
+      trim_index--;
+   if (diff < -QCONST16(4.f, DB_SHIFT))
+      trim_index++;
+   if (diff < -QCONST16(10.f, DB_SHIFT))
+      trim_index++;
+
+   if (trim_index<0)
+      trim_index = 0;
+   if (trim_index>10)
+      trim_index = 10;
+#ifdef FUZZING
+   trim_index = rand()%11;
+#endif
+   return trim_index;
+}
+
+static int stereo_analysis(const CELTMode *m, const celt_norm *X,
+      int LM, int N0)
+{
+   int i;
+   int thetas;
+   opus_val32 sumLR = EPSILON, sumMS = EPSILON;
+
+   /* Use the L1 norm to model the entropy of the L/R signal vs the M/S signal */
+   for (i=0;i<13;i++)
+   {
+      int j;
+      for (j=m->eBands[i]<<LM;j<m->eBands[i+1]<<LM;j++)
+      {
+         opus_val32 L, R, M, S;
+         /* We cast to 32-bit first because of the -32768 case */
+         L = EXTEND32(X[j]);
+         R = EXTEND32(X[N0+j]);
+         M = ADD32(L, R);
+         S = SUB32(L, R);
+         sumLR = ADD32(sumLR, ADD32(ABS32(L), ABS32(R)));
+         sumMS = ADD32(sumMS, ADD32(ABS32(M), ABS32(S)));
+      }
+   }
+   sumMS = MULT16_32_Q15(QCONST16(0.707107f, 15), sumMS);
+   thetas = 13;
+   /* We don't need thetas for lower bands with LM<=1 */
+   if (LM<=1)
+      thetas -= 8;
+   return MULT16_32_Q15((m->eBands[13]<<(LM+1))+thetas, sumMS)
+         > MULT16_32_Q15(m->eBands[13]<<(LM+1), sumLR);
+}
+
+int celt_encode_with_ec(CELTEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc)
+{
+   int i, c, N;
+   opus_int32 bits;
+   ec_enc _enc;
+   VARDECL(celt_sig, in);
+   VARDECL(celt_sig, freq);
+   VARDECL(celt_norm, X);
+   VARDECL(celt_ener, bandE);
+   VARDECL(opus_val16, bandLogE);
+   VARDECL(int, fine_quant);
+   VARDECL(opus_val16, error);
+   VARDECL(int, pulses);
+   VARDECL(int, cap);
+   VARDECL(int, offsets);
+   VARDECL(int, fine_priority);
+   VARDECL(int, tf_res);
+   VARDECL(unsigned char, collapse_masks);
+   celt_sig *prefilter_mem;
+   opus_val16 *oldBandE, *oldLogE, *oldLogE2;
+   int shortBlocks=0;
+   int isTransient=0;
+   const int CC = st->channels;
+   const int C = st->stream_channels;
+   int LM, M;
+   int tf_select;
+   int nbFilledBytes, nbAvailableBytes;
+   int effEnd;
+   int codedBands;
+   int tf_sum;
+   int alloc_trim;
+   int pitch_index=COMBFILTER_MINPERIOD;
+   opus_val16 gain1 = 0;
+   int intensity=0;
+   int dual_stereo=0;
+   int effectiveBytes;
+   opus_val16 pf_threshold;
+   int dynalloc_logp;
+   opus_int32 vbr_rate;
+   opus_int32 total_bits;
+   opus_int32 total_boost;
+   opus_int32 balance;
+   opus_int32 tell;
+   int prefilter_tapset=0;
+   int pf_on;
+   int anti_collapse_rsv;
+   int anti_collapse_on=0;
+   int silence=0;
+   ALLOC_STACK;
+
+   if (nbCompressedBytes<2 || pcm==NULL)
+     return OPUS_BAD_ARG;
+
+   frame_size *= st->upsample;
+   for (LM=0;LM<=st->mode->maxLM;LM++)
+      if (st->mode->shortMdctSize<<LM==frame_size)
+         break;
+   if (LM>st->mode->maxLM)
+      return OPUS_BAD_ARG;
+   M=1<<LM;
+   N = M*st->mode->shortMdctSize;
+
+   prefilter_mem = st->in_mem+CC*(st->overlap);
+   oldBandE = (opus_val16*)(st->in_mem+CC*(2*st->overlap+COMBFILTER_MAXPERIOD));
+   oldLogE = oldBandE + CC*st->mode->nbEBands;
+   oldLogE2 = oldLogE + CC*st->mode->nbEBands;
+
+   if (enc==NULL)
+   {
+      tell=1;
+      nbFilledBytes=0;
+   } else {
+      tell=ec_tell(enc);
+      nbFilledBytes=(tell+4)>>3;
+   }
+
+#ifdef CUSTOM_MODES
+   if (st->signalling && enc==NULL)
+   {
+      int tmp = (st->mode->effEBands-st->end)>>1;
+      st->end = IMAX(1, st->mode->effEBands-tmp);
+      compressed[0] = tmp<<5;
+      compressed[0] |= LM<<3;
+      compressed[0] |= (C==2)<<2;
+      /* Convert "standard mode" to Opus header */
+      if (st->mode->Fs==48000 && st->mode->shortMdctSize==120)
+      {
+         int c0 = toOpus(compressed[0]);
+         if (c0<0)
+            return OPUS_BAD_ARG;
+         compressed[0] = c0;
+      }
+      compressed++;
+      nbCompressedBytes--;
+   }
+#else
+   celt_assert(st->signalling==0);
+#endif
+
+   /* Can't produce more than 1275 output bytes */
+   nbCompressedBytes = IMIN(nbCompressedBytes,1275);
+   nbAvailableBytes = nbCompressedBytes - nbFilledBytes;
+
+   if (st->vbr && st->bitrate!=OPUS_BITRATE_MAX)
+   {
+      opus_int32 den=st->mode->Fs>>BITRES;
+      vbr_rate=(st->bitrate*frame_size+(den>>1))/den;
+#ifdef CUSTOM_MODES
+      if (st->signalling)
+         vbr_rate -= 8<<BITRES;
+#endif
+      effectiveBytes = vbr_rate>>(3+BITRES);
+   } else {
+      opus_int32 tmp;
+      vbr_rate = 0;
+      tmp = st->bitrate*frame_size;
+      if (tell>1)
+         tmp += tell;
+      if (st->bitrate!=OPUS_BITRATE_MAX)
+         nbCompressedBytes = IMAX(2, IMIN(nbCompressedBytes,
+               (tmp+4*st->mode->Fs)/(8*st->mode->Fs)-!!st->signalling));
+      effectiveBytes = nbCompressedBytes;
+   }
+
+   if (enc==NULL)
+   {
+      ec_enc_init(&_enc, compressed, nbCompressedBytes);
+      enc = &_enc;
+   }
+
+   if (vbr_rate>0)
+   {
+      /* Computes the max bit-rate allowed in VBR mode to avoid violating the
+          target rate and buffering.
+         We must do this up front so that bust-prevention logic triggers
+          correctly if we don't have enough bits. */
+      if (st->constrained_vbr)
+      {
+         opus_int32 vbr_bound;
+         opus_int32 max_allowed;
+         /* We could use any multiple of vbr_rate as bound (depending on the
+             delay).
+            This is clamped to ensure we use at least two bytes if the encoder
+             was entirely empty, but to allow 0 in hybrid mode. */
+         vbr_bound = vbr_rate;
+         max_allowed = IMIN(IMAX(tell==1?2:0,
+               (vbr_rate+vbr_bound-st->vbr_reservoir)>>(BITRES+3)),
+               nbAvailableBytes);
+         if(max_allowed < nbAvailableBytes)
+         {
+            nbCompressedBytes = nbFilledBytes+max_allowed;
+            nbAvailableBytes = max_allowed;
+            ec_enc_shrink(enc, nbCompressedBytes);
+         }
+      }
+   }
+   total_bits = nbCompressedBytes*8;
+
+   effEnd = st->end;
+   if (effEnd > st->mode->effEBands)
+      effEnd = st->mode->effEBands;
+
+   ALLOC(in, CC*(N+st->overlap), celt_sig);
+
+   /* Find pitch period and gain */
+   {
+      VARDECL(celt_sig, _pre);
+      celt_sig *pre[2];
+      SAVE_STACK;
+      ALLOC(_pre, CC*(N+COMBFILTER_MAXPERIOD), celt_sig);
+
+      pre[0] = _pre;
+      pre[1] = _pre + (N+COMBFILTER_MAXPERIOD);
+
+      silence = 1;
+      c=0; do {
+         int count = 0;
+         const opus_val16 * OPUS_RESTRICT pcmp = pcm+c;
+         celt_sig * OPUS_RESTRICT inp = in+c*(N+st->overlap)+st->overlap;
+
+         for (i=0;i<N;i++)
+         {
+            celt_sig x, tmp;
+
+            x = SCALEIN(*pcmp);
+#ifndef FIXED_POINT
+            if (!(x==x))
+               x = 0;
+            if (st->clip)
+               x = MAX32(-65536.f, MIN32(65536.f,x));
+#endif
+            if (++count==st->upsample)
+            {
+               count=0;
+               pcmp+=CC;
+            } else {
+               x = 0;
+            }
+            /* Apply pre-emphasis */
+            tmp = MULT16_16(st->mode->preemph[2], x);
+            *inp = tmp + st->preemph_memE[c];
+            st->preemph_memE[c] = MULT16_32_Q15(st->mode->preemph[1], *inp)
+                                   - MULT16_32_Q15(st->mode->preemph[0], tmp);
+            silence = silence && *inp == 0;
+            inp++;
+         }
+         OPUS_COPY(pre[c], prefilter_mem+c*COMBFILTER_MAXPERIOD, COMBFILTER_MAXPERIOD);
+         OPUS_COPY(pre[c]+COMBFILTER_MAXPERIOD, in+c*(N+st->overlap)+st->overlap, N);
+      } while (++c<CC);
+
+#ifdef FUZZING
+      if ((rand()&0x3F)==0)
+         silence = 1;
+#endif
+      if (tell==1)
+         ec_enc_bit_logp(enc, silence, 15);
+      else
+         silence=0;
+      if (silence)
+      {
+         /*In VBR mode there is no need to send more than the minimum. */
+         if (vbr_rate>0)
+         {
+            effectiveBytes=nbCompressedBytes=IMIN(nbCompressedBytes, nbFilledBytes+2);
+            total_bits=nbCompressedBytes*8;
+            nbAvailableBytes=2;
+            ec_enc_shrink(enc, nbCompressedBytes);
+         }
+         /* Pretend we've filled all the remaining bits with zeros
+            (that's what the initialiser did anyway) */
+         tell = nbCompressedBytes*8;
+         enc->nbits_total+=tell-ec_tell(enc);
+      }
+      if (nbAvailableBytes>12*C && st->start==0 && !silence && !st->disable_pf && st->complexity >= 5)
+      {
+         VARDECL(opus_val16, pitch_buf);
+         ALLOC(pitch_buf, (COMBFILTER_MAXPERIOD+N)>>1, opus_val16);
+
+         pitch_downsample(pre, pitch_buf, COMBFILTER_MAXPERIOD+N, CC);
+         pitch_search(pitch_buf+(COMBFILTER_MAXPERIOD>>1), pitch_buf, N,
+               COMBFILTER_MAXPERIOD-COMBFILTER_MINPERIOD, &pitch_index);
+         pitch_index = COMBFILTER_MAXPERIOD-pitch_index;
+
+         gain1 = remove_doubling(pitch_buf, COMBFILTER_MAXPERIOD, COMBFILTER_MINPERIOD,
+               N, &pitch_index, st->prefilter_period, st->prefilter_gain);
+         if (pitch_index > COMBFILTER_MAXPERIOD-2)
+            pitch_index = COMBFILTER_MAXPERIOD-2;
+         gain1 = MULT16_16_Q15(QCONST16(.7f,15),gain1);
+         if (st->loss_rate>2)
+            gain1 = HALF32(gain1);
+         if (st->loss_rate>4)
+            gain1 = HALF32(gain1);
+         if (st->loss_rate>8)
+            gain1 = 0;
+         prefilter_tapset = st->tapset_decision;
+      } else {
+         gain1 = 0;
+      }
+
+      /* Gain threshold for enabling the prefilter/postfilter */
+      pf_threshold = QCONST16(.2f,15);
+
+      /* Adjusting the threshold based on rate and continuity */
+      if (abs(pitch_index-st->prefilter_period)*10>pitch_index)
+         pf_threshold += QCONST16(.2f,15);
+      if (nbAvailableBytes<25)
+         pf_threshold += QCONST16(.1f,15);
+      if (nbAvailableBytes<35)
+         pf_threshold += QCONST16(.1f,15);
+      if (st->prefilter_gain > QCONST16(.4f,15))
+         pf_threshold -= QCONST16(.1f,15);
+      if (st->prefilter_gain > QCONST16(.55f,15))
+         pf_threshold -= QCONST16(.1f,15);
+
+      /* Hard threshold at 0.2 */
+      pf_threshold = MAX16(pf_threshold, QCONST16(.2f,15));
+      if (gain1<pf_threshold)
+      {
+         if(st->start==0 && tell+16<=total_bits)
+            ec_enc_bit_logp(enc, 0, 1);
+         gain1 = 0;
+         pf_on = 0;
+      } else {
+         /*This block is not gated by a total bits check only because
+           of the nbAvailableBytes check above.*/
+         int qg;
+         int octave;
+
+         if (ABS16(gain1-st->prefilter_gain)<QCONST16(.1f,15))
+            gain1=st->prefilter_gain;
+
+#ifdef FIXED_POINT
+         qg = ((gain1+1536)>>10)/3-1;
+#else
+         qg = (int)floor(.5f+gain1*32/3)-1;
+#endif
+         qg = IMAX(0, IMIN(7, qg));
+         ec_enc_bit_logp(enc, 1, 1);
+         pitch_index += 1;
+         octave = EC_ILOG(pitch_index)-5;
+         ec_enc_uint(enc, octave, 6);
+         ec_enc_bits(enc, pitch_index-(16<<octave), 4+octave);
+         pitch_index -= 1;
+         ec_enc_bits(enc, qg, 3);
+         if (ec_tell(enc)+2<=total_bits)
+            ec_enc_icdf(enc, prefilter_tapset, tapset_icdf, 2);
+         else
+           prefilter_tapset = 0;
+         gain1 = QCONST16(0.09375f,15)*(qg+1);
+         pf_on = 1;
+      }
+      /*printf("%d %f\n", pitch_index, gain1);*/
+
+      c=0; do {
+         int offset = st->mode->shortMdctSize-st->mode->overlap;
+         st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD);
+         OPUS_COPY(in+c*(N+st->overlap), st->in_mem+c*(st->overlap), st->overlap);
+         if (offset)
+            comb_filter(in+c*(N+st->overlap)+st->overlap, pre[c]+COMBFILTER_MAXPERIOD,
+                  st->prefilter_period, st->prefilter_period, offset, -st->prefilter_gain, -st->prefilter_gain,
+                  st->prefilter_tapset, st->prefilter_tapset, NULL, 0);
+
+         comb_filter(in+c*(N+st->overlap)+st->overlap+offset, pre[c]+COMBFILTER_MAXPERIOD+offset,
+               st->prefilter_period, pitch_index, N-offset, -st->prefilter_gain, -gain1,
+               st->prefilter_tapset, prefilter_tapset, st->mode->window, st->mode->overlap);
+         OPUS_COPY(st->in_mem+c*(st->overlap), in+c*(N+st->overlap)+N, st->overlap);
+
+         if (N>COMBFILTER_MAXPERIOD)
+         {
+            OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD, pre[c]+N, COMBFILTER_MAXPERIOD);
+         } else {
+            OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD, prefilter_mem+c*COMBFILTER_MAXPERIOD+N, COMBFILTER_MAXPERIOD-N);
+            OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD+COMBFILTER_MAXPERIOD-N, pre[c]+COMBFILTER_MAXPERIOD, N);
+         }
+      } while (++c<CC);
+
+      RESTORE_STACK;
+   }
+
+   isTransient = 0;
+   shortBlocks = 0;
+   if (LM>0 && ec_tell(enc)+3<=total_bits)
+   {
+      if (st->complexity > 1)
+      {
+         isTransient = transient_analysis(in, N+st->overlap, CC,
+                  st->overlap);
+         if (isTransient)
+            shortBlocks = M;
+      }
+      ec_enc_bit_logp(enc, isTransient, 3);
+   }
+
+   ALLOC(freq, CC*N, celt_sig); /**< Interleaved signal MDCTs */
+   ALLOC(bandE,st->mode->nbEBands*CC, celt_ener);
+   ALLOC(bandLogE,st->mode->nbEBands*CC, opus_val16);
+   /* Compute MDCTs */
+   compute_mdcts(st->mode, shortBlocks, in, freq, CC, LM);
+
+   if (CC==2&&C==1)
+   {
+      for (i=0;i<N;i++)
+         freq[i] = ADD32(HALF32(freq[i]), HALF32(freq[N+i]));
+   }
+   if (st->upsample != 1)
+   {
+      c=0; do
+      {
+         int bound = N/st->upsample;
+         for (i=0;i<bound;i++)
+            freq[c*N+i] *= st->upsample;
+         for (;i<N;i++)
+            freq[c*N+i] = 0;
+      } while (++c<C);
+   }
+   ALLOC(X, C*N, celt_norm);         /**< Interleaved normalised MDCTs */
+
+   compute_band_energies(st->mode, freq, bandE, effEnd, C, M);
+
+   amp2Log2(st->mode, effEnd, st->end, bandE, bandLogE, C);
+
+   /* Band normalisation */
+   normalise_bands(st->mode, freq, X, bandE, effEnd, C, M);
+
+   ALLOC(tf_res, st->mode->nbEBands, int);
+   tf_select = tf_analysis(st->mode, effEnd, C, isTransient, tf_res, effectiveBytes, X, N, LM, &tf_sum);
+   for (i=effEnd;i<st->end;i++)
+      tf_res[i] = tf_res[effEnd-1];
+
+   ALLOC(error, C*st->mode->nbEBands, opus_val16);
+   quant_coarse_energy(st->mode, st->start, st->end, effEnd, bandLogE,
+         oldBandE, total_bits, error, enc,
+         C, LM, nbAvailableBytes, st->force_intra,
+         &st->delayedIntra, st->complexity >= 4, st->loss_rate);
+
+   tf_encode(st->start, st->end, isTransient, tf_res, LM, tf_select, enc);
+
+   st->spread_decision = SPREAD_NORMAL;
+   if (ec_tell(enc)+4<=total_bits)
+   {
+      if (shortBlocks || st->complexity < 3 || nbAvailableBytes < 10*C)
+      {
+         if (st->complexity == 0)
+            st->spread_decision = SPREAD_NONE;
+      } else {
+         st->spread_decision = spreading_decision(st->mode, X,
+               &st->tonal_average, st->spread_decision, &st->hf_average,
+               &st->tapset_decision, pf_on&&!shortBlocks, effEnd, C, M);
+      }
+      ec_enc_icdf(enc, st->spread_decision, spread_icdf, 5);
+   }
+
+   ALLOC(cap, st->mode->nbEBands, int);
+   ALLOC(offsets, st->mode->nbEBands, int);
+
+   init_caps(st->mode,cap,LM,C);
+   for (i=0;i<st->mode->nbEBands;i++)
+      offsets[i] = 0;
+   /* Dynamic allocation code */
+   /* Make sure that dynamic allocation can't make us bust the budget */
+   if (effectiveBytes > 50 && LM>=1)
+   {
+      int t1, t2;
+      if (LM <= 1)
+      {
+         t1 = 3;
+         t2 = 5;
+      } else {
+         t1 = 2;
+         t2 = 4;
+      }
+      for (i=st->start+1;i<st->end-1;i++)
+      {
+         opus_val32 d2;
+         d2 = 2*bandLogE[i]-bandLogE[i-1]-bandLogE[i+1];
+         if (C==2)
+            d2 = HALF32(d2 + 2*bandLogE[i+st->mode->nbEBands]-
+                  bandLogE[i-1+st->mode->nbEBands]-bandLogE[i+1+st->mode->nbEBands]);
+#ifdef FUZZING
+         if((rand()&0xF)==0)
+         {
+            offsets[i] += 1;
+            if((rand()&0x3)==0)
+               offsets[i] += 1+(rand()&0x3);
+         }
+#else
+         if (d2 > SHL16(t1,DB_SHIFT))
+            offsets[i] += 1;
+         if (d2 > SHL16(t2,DB_SHIFT))
+            offsets[i] += 1;
+#endif
+      }
+   }
+   dynalloc_logp = 6;
+   total_bits<<=BITRES;
+   total_boost = 0;
+   tell = ec_tell_frac(enc);
+   for (i=st->start;i<st->end;i++)
+   {
+      int width, quanta;
+      int dynalloc_loop_logp;
+      int boost;
+      int j;
+      width = C*(st->mode->eBands[i+1]-st->mode->eBands[i])<<LM;
+      /* quanta is 6 bits, but no more than 1 bit/sample
+         and no less than 1/8 bit/sample */
+      quanta = IMIN(width<<BITRES, IMAX(6<<BITRES, width));
+      dynalloc_loop_logp = dynalloc_logp;
+      boost = 0;
+      for (j = 0; tell+(dynalloc_loop_logp<<BITRES) < total_bits-total_boost
+            && boost < cap[i]; j++)
+      {
+         int flag;
+         flag = j<offsets[i];
+         ec_enc_bit_logp(enc, flag, dynalloc_loop_logp);
+         tell = ec_tell_frac(enc);
+         if (!flag)
+            break;
+         boost += quanta;
+         total_boost += quanta;
+         dynalloc_loop_logp = 1;
+      }
+      /* Making dynalloc more likely */
+      if (j)
+         dynalloc_logp = IMAX(2, dynalloc_logp-1);
+      offsets[i] = boost;
+   }
+   alloc_trim = 5;
+   if (tell+(6<<BITRES) <= total_bits - total_boost)
+   {
+      alloc_trim = alloc_trim_analysis(st->mode, X, bandLogE,
+            st->end, LM, C, N);
+      ec_enc_icdf(enc, alloc_trim, trim_icdf, 7);
+      tell = ec_tell_frac(enc);
+   }
+
+   /* Variable bitrate */
+   if (vbr_rate>0)
+   {
+     opus_val16 alpha;
+     opus_int32 delta;
+     /* The target rate in 8th bits per frame */
+     opus_int32 target;
+     opus_int32 min_allowed;
+     int lm_diff = st->mode->maxLM - LM;
+
+     /* Don't attempt to use more than 510 kb/s, even for frames smaller than 20 ms.
+        The CELT allocator will just not be able to use more than that anyway. */
+     nbCompressedBytes = IMIN(nbCompressedBytes,1275>>(3-LM));
+     target = vbr_rate + (st->vbr_offset>>lm_diff) - ((40*C+20)<<BITRES);
+
+     /* Shortblocks get a large boost in bitrate, but since they
+        are uncommon long blocks are not greatly affected */
+     if (shortBlocks || tf_sum < -2*(st->end-st->start))
+        target = 7*target/4;
+     else if (tf_sum < -(st->end-st->start))
+        target = 3*target/2;
+     else if (M > 1)
+        target-=(target+14)/28;
+
+     /* The current offset is removed from the target and the space used
+        so far is added*/
+     target=target+tell;
+
+     /* In VBR mode the frame size must not be reduced so much that it would
+         result in the encoder running out of bits.
+        The margin of 2 bytes ensures that none of the bust-prevention logic
+         in the decoder will have triggered so far. */
+     min_allowed = ((tell+total_boost+(1<<(BITRES+3))-1)>>(BITRES+3)) + 2 - nbFilledBytes;
+
+     nbAvailableBytes = (target+(1<<(BITRES+2)))>>(BITRES+3);
+     nbAvailableBytes = IMAX(min_allowed,nbAvailableBytes);
+     nbAvailableBytes = IMIN(nbCompressedBytes,nbAvailableBytes+nbFilledBytes) - nbFilledBytes;
+
+     /* By how much did we "miss" the target on that frame */
+     delta = target - vbr_rate;
+
+     target=nbAvailableBytes<<(BITRES+3);
+
+     /*If the frame is silent we don't adjust our drift, otherwise
+       the encoder will shoot to very high rates after hitting a
+       span of silence, but we do allow the bitres to refill.
+       This means that we'll undershoot our target in CVBR/VBR modes
+       on files with lots of silence. */
+     if(silence)
+     {
+       nbAvailableBytes = 2;
+       target = 2*8<<BITRES;
+       delta = 0;
+     }
+
+     if (st->vbr_count < 970)
+     {
+        st->vbr_count++;
+        alpha = celt_rcp(SHL32(EXTEND32(st->vbr_count+20),16));
+     } else
+        alpha = QCONST16(.001f,15);
+     /* How many bits have we used in excess of what we're allowed */
+     if (st->constrained_vbr)
+        st->vbr_reservoir += target - vbr_rate;
+     /*printf ("%d\n", st->vbr_reservoir);*/
+
+     /* Compute the offset we need to apply in order to reach the target */
+     st->vbr_drift += (opus_int32)MULT16_32_Q15(alpha,(delta*(1<<lm_diff))-st->vbr_offset-st->vbr_drift);
+     st->vbr_offset = -st->vbr_drift;
+     /*printf ("%d\n", st->vbr_drift);*/
+
+     if (st->constrained_vbr && st->vbr_reservoir < 0)
+     {
+        /* We're under the min value -- increase rate */
+        int adjust = (-st->vbr_reservoir)/(8<<BITRES);
+        /* Unless we're just coding silence */
+        nbAvailableBytes += silence?0:adjust;
+        st->vbr_reservoir = 0;
+        /*printf ("+%d\n", adjust);*/
+     }
+     nbCompressedBytes = IMIN(nbCompressedBytes,nbAvailableBytes+nbFilledBytes);
+     /* This moves the raw bits to take into account the new compressed size */
+     ec_enc_shrink(enc, nbCompressedBytes);
+   }
+   if (C==2)
+   {
+      int effectiveRate;
+
+      /* Always use MS for 2.5 ms frames until we can do a better analysis */
+      if (LM!=0)
+         dual_stereo = stereo_analysis(st->mode, X, LM, N);
+
+      /* Account for coarse energy */
+      effectiveRate = (8*effectiveBytes - 80)>>LM;
+
+      /* effectiveRate in kb/s */
+      effectiveRate = 2*effectiveRate/5;
+      if (effectiveRate<35)
+         intensity = 8;
+      else if (effectiveRate<50)
+         intensity = 12;
+      else if (effectiveRate<68)
+         intensity = 16;
+      else if (effectiveRate<84)
+         intensity = 18;
+      else if (effectiveRate<102)
+         intensity = 19;
+      else if (effectiveRate<130)
+         intensity = 20;
+      else
+         intensity = 100;
+      intensity = IMIN(st->end,IMAX(st->start, intensity));
+   }
+
+   /* Bit allocation */
+   ALLOC(fine_quant, st->mode->nbEBands, int);
+   ALLOC(pulses, st->mode->nbEBands, int);
+   ALLOC(fine_priority, st->mode->nbEBands, int);
+
+   /* bits =           packet size                    - where we are - safety*/
+   bits = (((opus_int32)nbCompressedBytes*8)<<BITRES) - ec_tell_frac(enc) - 1;
+   anti_collapse_rsv = isTransient&&LM>=2&&bits>=((LM+2)<<BITRES) ? (1<<BITRES) : 0;
+   bits -= anti_collapse_rsv;
+   codedBands = compute_allocation(st->mode, st->start, st->end, offsets, cap,
+         alloc_trim, &intensity, &dual_stereo, bits, &balance, pulses,
+         fine_quant, fine_priority, C, LM, enc, 1, st->lastCodedBands);
+   st->lastCodedBands = codedBands;
+
+   quant_fine_energy(st->mode, st->start, st->end, oldBandE, error, fine_quant, enc, C);
+
+#ifdef MEASURE_NORM_MSE
+   float X0[3000];
+   float bandE0[60];
+   c=0; do
+      for (i=0;i<N;i++)
+         X0[i+c*N] = X[i+c*N];
+   while (++c<C);
+   for (i=0;i<C*st->mode->nbEBands;i++)
+      bandE0[i] = bandE[i];
+#endif
+
+   /* Residual quantisation */
+   ALLOC(collapse_masks, C*st->mode->nbEBands, unsigned char);
+   quant_all_bands(1, st->mode, st->start, st->end, X, C==2 ? X+N : NULL, collapse_masks,
+         bandE, pulses, shortBlocks, st->spread_decision, dual_stereo, intensity, tf_res,
+         nbCompressedBytes*(8<<BITRES)-anti_collapse_rsv, balance, enc, LM, codedBands, &st->rng);
+
+   if (anti_collapse_rsv > 0)
+   {
+      anti_collapse_on = st->consec_transient<2;
+#ifdef FUZZING
+      anti_collapse_on = rand()&0x1;
+#endif
+      ec_enc_bits(enc, anti_collapse_on, 1);
+   }
+   quant_energy_finalise(st->mode, st->start, st->end, oldBandE, error, fine_quant, fine_priority, nbCompressedBytes*8-ec_tell(enc), enc, C);
+
+   if (silence)
+   {
+      for (i=0;i<C*st->mode->nbEBands;i++)
+         oldBandE[i] = -QCONST16(28.f,DB_SHIFT);
+   }
+
+#ifdef RESYNTH
+   /* Re-synthesis of the coded audio if required */
+   {
+      celt_sig *out_mem[2];
+      celt_sig *overlap_mem[2];
+
+      log2Amp(st->mode, st->start, st->end, bandE, oldBandE, C);
+      if (silence)
+      {
+         for (i=0;i<C*st->mode->nbEBands;i++)
+            bandE[i] = 0;
+      }
+
+#ifdef MEASURE_NORM_MSE
+      measure_norm_mse(st->mode, X, X0, bandE, bandE0, M, N, C);
+#endif
+      if (anti_collapse_on)
+      {
+         anti_collapse(st->mode, X, collapse_masks, LM, C, N,
+               st->start, st->end, oldBandE, oldLogE, oldLogE2, pulses, st->rng);
+      }
+
+      /* Synthesis */
+      denormalise_bands(st->mode, X, freq, bandE, effEnd, C, M);
+
+      OPUS_MOVE(st->syn_mem[0], st->syn_mem[0]+N, MAX_PERIOD);
+      if (CC==2)
+         OPUS_MOVE(st->syn_mem[1], st->syn_mem[1]+N, MAX_PERIOD);
+
+      c=0; do
+         for (i=0;i<M*st->mode->eBands[st->start];i++)
+            freq[c*N+i] = 0;
+      while (++c<C);
+      c=0; do
+         for (i=M*st->mode->eBands[st->end];i<N;i++)
+            freq[c*N+i] = 0;
+      while (++c<C);
+
+      if (CC==2&&C==1)
+      {
+         for (i=0;i<N;i++)
+            freq[N+i] = freq[i];
+      }
+
+      out_mem[0] = st->syn_mem[0]+MAX_PERIOD;
+      if (CC==2)
+         out_mem[1] = st->syn_mem[1]+MAX_PERIOD;
+
+      overlap_mem[0] = prefilter_mem+CC*COMBFILTER_MAXPERIOD;
+      if (CC==2)
+         overlap_mem[1] = overlap_mem[0] + st->overlap;
+
+      compute_inv_mdcts(st->mode, shortBlocks, freq, out_mem, overlap_mem, CC, LM);
+
+      c=0; do {
+         st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD);
+         st->prefilter_period_old=IMAX(st->prefilter_period_old, COMBFILTER_MINPERIOD);
+         comb_filter(out_mem[c], out_mem[c], st->prefilter_period_old, st->prefilter_period, st->mode->shortMdctSize,
+               st->prefilter_gain_old, st->prefilter_gain, st->prefilter_tapset_old, st->prefilter_tapset,
+               st->mode->window, st->overlap);
+         if (LM!=0)
+            comb_filter(out_mem[c]+st->mode->shortMdctSize, out_mem[c]+st->mode->shortMdctSize, st->prefilter_period, pitch_index, N-st->mode->shortMdctSize,
+                  st->prefilter_gain, gain1, st->prefilter_tapset, prefilter_tapset,
+                  st->mode->window, st->mode->overlap);
+      } while (++c<CC);
+
+      deemphasis(out_mem, (opus_val16*)pcm, N, CC, st->upsample, st->mode->preemph, st->preemph_memD);
+      st->prefilter_period_old = st->prefilter_period;
+      st->prefilter_gain_old = st->prefilter_gain;
+      st->prefilter_tapset_old = st->prefilter_tapset;
+   }
+#endif
+
+   st->prefilter_period = pitch_index;
+   st->prefilter_gain = gain1;
+   st->prefilter_tapset = prefilter_tapset;
+#ifdef RESYNTH
+   if (LM!=0)
+   {
+      st->prefilter_period_old = st->prefilter_period;
+      st->prefilter_gain_old = st->prefilter_gain;
+      st->prefilter_tapset_old = st->prefilter_tapset;
+   }
+#endif
+
+   if (CC==2&&C==1) {
+      for (i=0;i<st->mode->nbEBands;i++)
+         oldBandE[st->mode->nbEBands+i]=oldBandE[i];
+   }
+
+   if (!isTransient)
+   {
+      for (i=0;i<CC*st->mode->nbEBands;i++)
+         oldLogE2[i] = oldLogE[i];
+      for (i=0;i<CC*st->mode->nbEBands;i++)
+         oldLogE[i] = oldBandE[i];
+   } else {
+      for (i=0;i<CC*st->mode->nbEBands;i++)
+         oldLogE[i] = MIN16(oldLogE[i], oldBandE[i]);
+   }
+   /* In case start or end were to change */
+   c=0; do
+   {
+      for (i=0;i<st->start;i++)
+      {
+         oldBandE[c*st->mode->nbEBands+i]=0;
+         oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT);
+      }
+      for (i=st->end;i<st->mode->nbEBands;i++)
+      {
+         oldBandE[c*st->mode->nbEBands+i]=0;
+         oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT);
+      }
+   } while (++c<CC);
+
+   if (isTransient)
+      st->consec_transient++;
+   else
+      st->consec_transient=0;
+   st->rng = enc->rng;
+
+   /* If there's any room left (can only happen for very high rates),
+      it's already filled with zeros */
+   ec_enc_done(enc);
+
+#ifdef CUSTOM_MODES
+   if (st->signalling)
+      nbCompressedBytes++;
+#endif
+
+   RESTORE_STACK;
+   if (ec_get_error(enc))
+      return OPUS_INTERNAL_ERROR;
+   else
+      return nbCompressedBytes;
+}
+
+
+#ifdef CUSTOM_MODES
+
+#ifdef FIXED_POINT
+int opus_custom_encode(CELTEncoder * OPUS_RESTRICT st, const opus_int16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes)
+{
+   return celt_encode_with_ec(st, pcm, frame_size, compressed, nbCompressedBytes, NULL);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_custom_encode_float(CELTEncoder * OPUS_RESTRICT st, const float * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes)
+{
+   int j, ret, C, N;
+   VARDECL(opus_int16, in);
+   ALLOC_STACK;
+
+   if (pcm==NULL)
+      return OPUS_BAD_ARG;
+
+   C = st->channels;
+   N = frame_size;
+   ALLOC(in, C*N, opus_int16);
+
+   for (j=0;j<C*N;j++)
+     in[j] = FLOAT2INT16(pcm[j]);
+
+   ret=celt_encode_with_ec(st,in,frame_size,compressed,nbCompressedBytes, NULL);
+#ifdef RESYNTH
+   for (j=0;j<C*N;j++)
+      ((float*)pcm)[j]=in[j]*(1.f/32768.f);
+#endif
+   RESTORE_STACK;
+   return ret;
+}
+#endif /* DISABLE_FLOAT_API */
+#else
+
+int opus_custom_encode(CELTEncoder * OPUS_RESTRICT st, const opus_int16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes)
+{
+   int j, ret, C, N;
+   VARDECL(celt_sig, in);
+   ALLOC_STACK;
+
+   if (pcm==NULL)
+      return OPUS_BAD_ARG;
+
+   C=st->channels;
+   N=frame_size;
+   ALLOC(in, C*N, celt_sig);
+   for (j=0;j<C*N;j++) {
+     in[j] = SCALEOUT(pcm[j]);
+   }
+
+   ret = celt_encode_with_ec(st,in,frame_size,compressed,nbCompressedBytes, NULL);
+#ifdef RESYNTH
+   for (j=0;j<C*N;j++)
+      ((opus_int16*)pcm)[j] = FLOAT2INT16(in[j]);
+#endif
+   RESTORE_STACK;
+   return ret;
+}
+
+int opus_custom_encode_float(CELTEncoder * OPUS_RESTRICT st, const float * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes)
+{
+   return celt_encode_with_ec(st, pcm, frame_size, compressed, nbCompressedBytes, NULL);
+}
+
+#endif
+
+#endif /* CUSTOM_MODES */
+
+int opus_custom_encoder_ctl(CELTEncoder * OPUS_RESTRICT st, int request, ...)
+{
+   va_list ap;
+
+   va_start(ap, request);
+   switch (request)
+   {
+      case OPUS_SET_COMPLEXITY_REQUEST:
+      {
+         int value = va_arg(ap, opus_int32);
+         if (value<0 || value>10)
+            goto bad_arg;
+         st->complexity = value;
+      }
+      break;
+      case CELT_SET_START_BAND_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<0 || value>=st->mode->nbEBands)
+            goto bad_arg;
+         st->start = value;
+      }
+      break;
+      case CELT_SET_END_BAND_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<1 || value>st->mode->nbEBands)
+            goto bad_arg;
+         st->end = value;
+      }
+      break;
+      case CELT_SET_PREDICTION_REQUEST:
+      {
+         int value = va_arg(ap, opus_int32);
+         if (value<0 || value>2)
+            goto bad_arg;
+         st->disable_pf = value<=1;
+         st->force_intra = value==0;
+      }
+      break;
+      case OPUS_SET_PACKET_LOSS_PERC_REQUEST:
+      {
+         int value = va_arg(ap, opus_int32);
+         if (value<0 || value>100)
+            goto bad_arg;
+         st->loss_rate = value;
+      }
+      break;
+      case OPUS_SET_VBR_CONSTRAINT_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         st->constrained_vbr = value;
+      }
+      break;
+      case OPUS_SET_VBR_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         st->vbr = value;
+      }
+      break;
+      case OPUS_SET_BITRATE_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<=500 && value!=OPUS_BITRATE_MAX)
+            goto bad_arg;
+         value = IMIN(value, 260000*st->channels);
+         st->bitrate = value;
+      }
+      break;
+      case CELT_SET_CHANNELS_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<1 || value>2)
+            goto bad_arg;
+         st->stream_channels = value;
+      }
+      break;
+      case OPUS_SET_LSB_DEPTH_REQUEST:
+      {
+          opus_int32 value = va_arg(ap, opus_int32);
+          if (value<8 || value>24)
+             goto bad_arg;
+          st->lsb_depth=value;
+      }
+      break;
+      case OPUS_GET_LSB_DEPTH_REQUEST:
+      {
+          opus_int32 *value = va_arg(ap, opus_int32*);
+          *value=st->lsb_depth;
+      }
+      break;
+      case OPUS_RESET_STATE:
+      {
+         int i;
+         opus_val16 *oldBandE, *oldLogE, *oldLogE2;
+         oldBandE = (opus_val16*)(st->in_mem+st->channels*(2*st->overlap+COMBFILTER_MAXPERIOD));
+         oldLogE = oldBandE + st->channels*st->mode->nbEBands;
+         oldLogE2 = oldLogE + st->channels*st->mode->nbEBands;
+         OPUS_CLEAR((char*)&st->ENCODER_RESET_START,
+               opus_custom_encoder_get_size(st->mode, st->channels)-
+               ((char*)&st->ENCODER_RESET_START - (char*)st));
+         for (i=0;i<st->channels*st->mode->nbEBands;i++)
+            oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT);
+         st->vbr_offset = 0;
+         st->delayedIntra = 1;
+         st->spread_decision = SPREAD_NORMAL;
+         st->tonal_average = 256;
+         st->hf_average = 0;
+         st->tapset_decision = 0;
+      }
+      break;
+#ifdef CUSTOM_MODES
+      case CELT_SET_INPUT_CLIPPING_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         st->clip = value;
+      }
+      break;
+#endif
+      case CELT_SET_SIGNALLING_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         st->signalling = value;
+      }
+      break;
+      case CELT_GET_MODE_REQUEST:
+      {
+         const CELTMode ** value = va_arg(ap, const CELTMode**);
+         if (value==0)
+            goto bad_arg;
+         *value=st->mode;
+      }
+      break;
+      case OPUS_GET_FINAL_RANGE_REQUEST:
+      {
+         opus_uint32 * value = va_arg(ap, opus_uint32 *);
+         if (value==0)
+            goto bad_arg;
+         *value=st->rng;
+      }
+      break;
+      default:
+         goto bad_request;
+   }
+   va_end(ap);
+   return OPUS_OK;
+bad_arg:
+   va_end(ap);
+   return OPUS_BAD_ARG;
+bad_request:
+   va_end(ap);
+   return OPUS_UNIMPLEMENTED;
+}
+
+/**********************************************************************/
+/*                                                                    */
+/*                             DECODER                                */
+/*                                                                    */
+/**********************************************************************/
+#define DECODE_BUFFER_SIZE 2048
+
+/** Decoder state
+ @brief Decoder state
+ */
+struct OpusCustomDecoder {
+   const OpusCustomMode *mode;
+   int overlap;
+   int channels;
+   int stream_channels;
+
+   int downsample;
+   int start, end;
+   int signalling;
+
+   /* Everything beyond this point gets cleared on a reset */
+#define DECODER_RESET_START rng
+
+   opus_uint32 rng;
+   int error;
+   int last_pitch_index;
+   int loss_count;
+   int postfilter_period;
+   int postfilter_period_old;
+   opus_val16 postfilter_gain;
+   opus_val16 postfilter_gain_old;
+   int postfilter_tapset;
+   int postfilter_tapset_old;
+
+   celt_sig preemph_memD[2];
+
+   celt_sig _decode_mem[1]; /* Size = channels*(DECODE_BUFFER_SIZE+mode->overlap) */
+   /* opus_val16 lpc[],  Size = channels*LPC_ORDER */
+   /* opus_val16 oldEBands[], Size = 2*mode->nbEBands */
+   /* opus_val16 oldLogE[], Size = 2*mode->nbEBands */
+   /* opus_val16 oldLogE2[], Size = 2*mode->nbEBands */
+   /* opus_val16 backgroundLogE[], Size = 2*mode->nbEBands */
+};
+
+int celt_decoder_get_size(int channels)
+{
+   const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
+   return opus_custom_decoder_get_size(mode, channels);
+}
+
+OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int channels)
+{
+   int size = sizeof(struct CELTDecoder)
+            + (channels*(DECODE_BUFFER_SIZE+mode->overlap)-1)*sizeof(celt_sig)
+            + channels*LPC_ORDER*sizeof(opus_val16)
+            + 4*2*mode->nbEBands*sizeof(opus_val16);
+   return size;
+}
+
+#ifdef CUSTOM_MODES
+CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int *error)
+{
+   int ret;
+   CELTDecoder *st = (CELTDecoder *)opus_alloc(opus_custom_decoder_get_size(mode, channels));
+   ret = opus_custom_decoder_init(st, mode, channels);
+   if (ret != OPUS_OK)
+   {
+      opus_custom_decoder_destroy(st);
+      st = NULL;
+   }
+   if (error)
+      *error = ret;
+   return st;
+}
+#endif /* CUSTOM_MODES */
+
+int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels)
+{
+   int ret;
+   ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels);
+   if (ret != OPUS_OK)
+      return ret;
+   st->downsample = resampling_factor(sampling_rate);
+   if (st->downsample==0)
+      return OPUS_BAD_ARG;
+   else
+      return OPUS_OK;
+}
+
+OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMode *mode, int channels)
+{
+   if (channels < 0 || channels > 2)
+      return OPUS_BAD_ARG;
+
+   if (st==NULL)
+      return OPUS_ALLOC_FAIL;
+
+   OPUS_CLEAR((char*)st, opus_custom_decoder_get_size(mode, channels));
+
+   st->mode = mode;
+   st->overlap = mode->overlap;
+   st->stream_channels = st->channels = channels;
+
+   st->downsample = 1;
+   st->start = 0;
+   st->end = st->mode->effEBands;
+   st->signalling = 1;
+
+   st->loss_count = 0;
+
+   opus_custom_decoder_ctl(st, OPUS_RESET_STATE);
+
+   return OPUS_OK;
+}
+
+#ifdef CUSTOM_MODES
+void opus_custom_decoder_destroy(CELTDecoder *st)
+{
+   opus_free(st);
+}
+#endif /* CUSTOM_MODES */
+
+static void celt_decode_lost(CELTDecoder * OPUS_RESTRICT st, opus_val16 * OPUS_RESTRICT pcm, int N, int LM)
+{
+   int c;
+   int pitch_index;
+   int overlap = st->mode->overlap;
+   opus_val16 fade = Q15ONE;
+   int i, len;
+   const int C = st->channels;
+   int offset;
+   celt_sig *out_mem[2];
+   celt_sig *decode_mem[2];
+   celt_sig *overlap_mem[2];
+   opus_val16 *lpc;
+   opus_val32 *out_syn[2];
+   opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE;
+   SAVE_STACK;
+
+   c=0; do {
+      decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+st->overlap);
+      out_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE-MAX_PERIOD;
+      overlap_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE;
+   } while (++c<C);
+   lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*C);
+   oldBandE = lpc+C*LPC_ORDER;
+   oldLogE = oldBandE + 2*st->mode->nbEBands;
+   oldLogE2 = oldLogE + 2*st->mode->nbEBands;
+   backgroundLogE = oldLogE2  + 2*st->mode->nbEBands;
+
+   out_syn[0] = out_mem[0]+MAX_PERIOD-N;
+   if (C==2)
+      out_syn[1] = out_mem[1]+MAX_PERIOD-N;
+
+   len = N+st->mode->overlap;
+
+   if (st->loss_count >= 5 || st->start!=0)
+   {
+      /* Noise-based PLC/CNG */
+      VARDECL(celt_sig, freq);
+      VARDECL(celt_norm, X);
+      VARDECL(celt_ener, bandE);
+      opus_uint32 seed;
+      int effEnd;
+
+      effEnd = st->end;
+      if (effEnd > st->mode->effEBands)
+         effEnd = st->mode->effEBands;
+
+      ALLOC(freq, C*N, celt_sig); /**< Interleaved signal MDCTs */
+      ALLOC(X, C*N, celt_norm);   /**< Interleaved normalised MDCTs */
+      ALLOC(bandE, st->mode->nbEBands*C, celt_ener);
+
+      if (st->loss_count >= 5)
+         log2Amp(st->mode, st->start, st->end, bandE, backgroundLogE, C);
+      else {
+         /* Energy decay */
+         opus_val16 decay = st->loss_count==0 ? QCONST16(1.5f, DB_SHIFT) : QCONST16(.5f, DB_SHIFT);
+         c=0; do
+         {
+            for (i=st->start;i<st->end;i++)
+               oldBandE[c*st->mode->nbEBands+i] -= decay;
+         } while (++c<C);
+         log2Amp(st->mode, st->start, st->end, bandE, oldBandE, C);
+      }
+      seed = st->rng;
+      for (c=0;c<C;c++)
+      {
+         for (i=0;i<(st->mode->eBands[st->start]<<LM);i++)
+            X[c*N+i] = 0;
+         for (i=st->start;i<st->mode->effEBands;i++)
+         {
+            int j;
+            int boffs;
+            int blen;
+            boffs = N*c+(st->mode->eBands[i]<<LM);
+            blen = (st->mode->eBands[i+1]-st->mode->eBands[i])<<LM;
+            for (j=0;j<blen;j++)
+            {
+               seed = celt_lcg_rand(seed);
+               X[boffs+j] = (celt_norm)((opus_int32)seed>>20);
+            }
+            renormalise_vector(X+boffs, blen, Q15ONE);
+         }
+         for (i=(st->mode->eBands[st->end]<<LM);i<N;i++)
+            X[c*N+i] = 0;
+      }
+      st->rng = seed;
+
+      denormalise_bands(st->mode, X, freq, bandE, st->mode->effEBands, C, 1<<LM);
+
+      c=0; do
+         for (i=0;i<st->mode->eBands[st->start]<<LM;i++)
+            freq[c*N+i] = 0;
+      while (++c<C);
+      c=0; do {
+         int bound = st->mode->eBands[effEnd]<<LM;
+         if (st->downsample!=1)
+            bound = IMIN(bound, N/st->downsample);
+         for (i=bound;i<N;i++)
+            freq[c*N+i] = 0;
+      } while (++c<C);
+      compute_inv_mdcts(st->mode, 0, freq, out_syn, overlap_mem, C, LM);
+   } else {
+      /* Pitch-based PLC */
+      if (st->loss_count == 0)
+      {
+         opus_val16 pitch_buf[DECODE_BUFFER_SIZE>>1];
+         /* Corresponds to a min pitch of 67 Hz. It's possible to save CPU in this
+         search by using only part of the decode buffer */
+         int poffset = 720;
+         pitch_downsample(decode_mem, pitch_buf, DECODE_BUFFER_SIZE, C);
+         /* Max pitch is 100 samples (480 Hz) */
+         pitch_search(pitch_buf+((poffset)>>1), pitch_buf, DECODE_BUFFER_SIZE-poffset,
+               poffset-100, &pitch_index);
+         pitch_index = poffset-pitch_index;
+         st->last_pitch_index = pitch_index;
+      } else {
+         pitch_index = st->last_pitch_index;
+         fade = QCONST16(.8f,15);
+      }
+
+      c=0; do {
+         VARDECL(opus_val32, e);
+         opus_val16 exc[MAX_PERIOD];
+         opus_val32 ac[LPC_ORDER+1];
+         opus_val16 decay = 1;
+         opus_val32 S1=0;
+         opus_val16 mem[LPC_ORDER]={0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
+
+         ALLOC(e, MAX_PERIOD+2*st->mode->overlap, opus_val32);
+
+         offset = MAX_PERIOD-pitch_index;
+         for (i=0;i<MAX_PERIOD;i++)
+            exc[i] = ROUND16(out_mem[c][i], SIG_SHIFT);
+
+         if (st->loss_count == 0)
+         {
+            _celt_autocorr(exc, ac, st->mode->window, st->mode->overlap,
+                  LPC_ORDER, MAX_PERIOD);
+
+            /* Noise floor -40 dB */
+#ifdef FIXED_POINT
+            ac[0] += SHR32(ac[0],13);
+#else
+            ac[0] *= 1.0001f;
+#endif
+            /* Lag windowing */
+            for (i=1;i<=LPC_ORDER;i++)
+            {
+               /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/
+#ifdef FIXED_POINT
+               ac[i] -= MULT16_32_Q15(2*i*i, ac[i]);
+#else
+               ac[i] -= ac[i]*(.008f*i)*(.008f*i);
+#endif
+            }
+
+            _celt_lpc(lpc+c*LPC_ORDER, ac, LPC_ORDER);
+         }
+         for (i=0;i<LPC_ORDER;i++)
+            mem[i] = ROUND16(out_mem[c][MAX_PERIOD-1-i], SIG_SHIFT);
+         celt_fir(exc, lpc+c*LPC_ORDER, exc, MAX_PERIOD, LPC_ORDER, mem);
+         /*for (i=0;i<MAX_PERIOD;i++)printf("%d ", exc[i]); printf("\n");*/
+         /* Check if the waveform is decaying (and if so how fast) */
+         {
+            opus_val32 E1=1, E2=1;
+            int period;
+            if (pitch_index <= MAX_PERIOD/2)
+               period = pitch_index;
+            else
+               period = MAX_PERIOD/2;
+            for (i=0;i<period;i++)
+            {
+               E1 += SHR32(MULT16_16(exc[MAX_PERIOD-period+i],exc[MAX_PERIOD-period+i]),8);
+               E2 += SHR32(MULT16_16(exc[MAX_PERIOD-2*period+i],exc[MAX_PERIOD-2*period+i]),8);
+            }
+            if (E1 > E2)
+               E1 = E2;
+            decay = celt_sqrt(frac_div32(SHR32(E1,1),E2));
+         }
+
+         /* Copy excitation, taking decay into account */
+         for (i=0;i<len+st->mode->overlap;i++)
+         {
+            opus_val16 tmp;
+            if (offset+i >= MAX_PERIOD)
+            {
+               offset -= pitch_index;
+               decay = MULT16_16_Q15(decay, decay);
+            }
+            e[i] = SHL32(EXTEND32(MULT16_16_Q15(decay, exc[offset+i])), SIG_SHIFT);
+            tmp = ROUND16(out_mem[c][offset+i],SIG_SHIFT);
+            S1 += SHR32(MULT16_16(tmp,tmp),8);
+         }
+         for (i=0;i<LPC_ORDER;i++)
+            mem[i] = ROUND16(out_mem[c][MAX_PERIOD-1-i], SIG_SHIFT);
+         for (i=0;i<len+st->mode->overlap;i++)
+            e[i] = MULT16_32_Q15(fade, e[i]);
+         celt_iir(e, lpc+c*LPC_ORDER, e, len+st->mode->overlap, LPC_ORDER, mem);
+
+         {
+            opus_val32 S2=0;
+            for (i=0;i<len+overlap;i++)
+            {
+               opus_val16 tmp = ROUND16(e[i],SIG_SHIFT);
+               S2 += SHR32(MULT16_16(tmp,tmp),8);
+            }
+            /* This checks for an "explosion" in the synthesis */
+#ifdef FIXED_POINT
+            if (!(S1 > SHR32(S2,2)))
+#else
+               /* Float test is written this way to catch NaNs at the same time */
+               if (!(S1 > 0.2f*S2))
+#endif
+               {
+                  for (i=0;i<len+overlap;i++)
+                     e[i] = 0;
+               } else if (S1 < S2)
+               {
+                  opus_val16 ratio = celt_sqrt(frac_div32(SHR32(S1,1)+1,S2+1));
+                  for (i=0;i<len+overlap;i++)
+                     e[i] = MULT16_32_Q15(ratio, e[i]);
+               }
+         }
+
+         /* Apply post-filter to the MDCT overlap of the previous frame */
+         comb_filter(out_mem[c]+MAX_PERIOD, out_mem[c]+MAX_PERIOD, st->postfilter_period, st->postfilter_period, st->overlap,
+               st->postfilter_gain, st->postfilter_gain, st->postfilter_tapset, st->postfilter_tapset,
+               NULL, 0);
+
+         for (i=0;i<MAX_PERIOD+st->mode->overlap-N;i++)
+            out_mem[c][i] = out_mem[c][N+i];
+
+         /* Apply TDAC to the concealed audio so that it blends with the
+         previous and next frames */
+         for (i=0;i<overlap/2;i++)
+         {
+            opus_val32 tmp;
+            tmp = MULT16_32_Q15(st->mode->window[i],           e[N+overlap-1-i]) +
+                  MULT16_32_Q15(st->mode->window[overlap-i-1], e[N+i          ]);
+            out_mem[c][MAX_PERIOD+i] = MULT16_32_Q15(st->mode->window[overlap-i-1], tmp);
+            out_mem[c][MAX_PERIOD+overlap-i-1] = MULT16_32_Q15(st->mode->window[i], tmp);
+         }
+         for (i=0;i<N;i++)
+            out_mem[c][MAX_PERIOD-N+i] = e[i];
+
+         /* Apply pre-filter to the MDCT overlap for the next frame (post-filter will be applied then) */
+         comb_filter(e, out_mem[c]+MAX_PERIOD, st->postfilter_period, st->postfilter_period, st->overlap,
+               -st->postfilter_gain, -st->postfilter_gain, st->postfilter_tapset, st->postfilter_tapset,
+               NULL, 0);
+         for (i=0;i<overlap;i++)
+            out_mem[c][MAX_PERIOD+i] = e[i];
+      } while (++c<C);
+   }
+
+   deemphasis(out_syn, pcm, N, C, st->downsample, st->mode->preemph, st->preemph_memD);
+
+   st->loss_count++;
+
+   RESTORE_STACK;
+}
+
+int celt_decode_with_ec(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec)
+{
+   int c, i, N;
+   int spread_decision;
+   opus_int32 bits;
+   ec_dec _dec;
+   VARDECL(celt_sig, freq);
+   VARDECL(celt_norm, X);
+   VARDECL(celt_ener, bandE);
+   VARDECL(int, fine_quant);
+   VARDECL(int, pulses);
+   VARDECL(int, cap);
+   VARDECL(int, offsets);
+   VARDECL(int, fine_priority);
+   VARDECL(int, tf_res);
+   VARDECL(unsigned char, collapse_masks);
+   celt_sig *out_mem[2];
+   celt_sig *decode_mem[2];
+   celt_sig *overlap_mem[2];
+   celt_sig *out_syn[2];
+   opus_val16 *lpc;
+   opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE;
+
+   int shortBlocks;
+   int isTransient;
+   int intra_ener;
+   const int CC = st->channels;
+   int LM, M;
+   int effEnd;
+   int codedBands;
+   int alloc_trim;
+   int postfilter_pitch;
+   opus_val16 postfilter_gain;
+   int intensity=0;
+   int dual_stereo=0;
+   opus_int32 total_bits;
+   opus_int32 balance;
+   opus_int32 tell;
+   int dynalloc_logp;
+   int postfilter_tapset;
+   int anti_collapse_rsv;
+   int anti_collapse_on=0;
+   int silence;
+   int C = st->stream_channels;
+   ALLOC_STACK;
+
+   frame_size *= st->downsample;
+
+   c=0; do {
+      decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+st->overlap);
+      out_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE-MAX_PERIOD;
+      overlap_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE;
+   } while (++c<CC);
+   lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*CC);
+   oldBandE = lpc+CC*LPC_ORDER;
+   oldLogE = oldBandE + 2*st->mode->nbEBands;
+   oldLogE2 = oldLogE + 2*st->mode->nbEBands;
+   backgroundLogE = oldLogE2  + 2*st->mode->nbEBands;
+
+#ifdef CUSTOM_MODES
+   if (st->signalling && data!=NULL)
+   {
+      int data0=data[0];
+      /* Convert "standard mode" to Opus header */
+      if (st->mode->Fs==48000 && st->mode->shortMdctSize==120)
+      {
+         data0 = fromOpus(data0);
+         if (data0<0)
+            return OPUS_INVALID_PACKET;
+      }
+      st->end = IMAX(1, st->mode->effEBands-2*(data0>>5));
+      LM = (data0>>3)&0x3;
+      C = 1 + ((data0>>2)&0x1);
+      data++;
+      len--;
+      if (LM>st->mode->maxLM)
+         return OPUS_INVALID_PACKET;
+      if (frame_size < st->mode->shortMdctSize<<LM)
+         return OPUS_BUFFER_TOO_SMALL;
+      else
+         frame_size = st->mode->shortMdctSize<<LM;
+   } else {
+#else
+   {
+#endif
+      for (LM=0;LM<=st->mode->maxLM;LM++)
+         if (st->mode->shortMdctSize<<LM==frame_size)
+            break;
+      if (LM>st->mode->maxLM)
+         return OPUS_BAD_ARG;
+   }
+   M=1<<LM;
+
+   if (len<0 || len>1275 || pcm==NULL)
+      return OPUS_BAD_ARG;
+
+   N = M*st->mode->shortMdctSize;
+
+   effEnd = st->end;
+   if (effEnd > st->mode->effEBands)
+      effEnd = st->mode->effEBands;
+
+   ALLOC(freq, IMAX(CC,C)*N, celt_sig); /**< Interleaved signal MDCTs */
+   ALLOC(X, C*N, celt_norm);   /**< Interleaved normalised MDCTs */
+   ALLOC(bandE, st->mode->nbEBands*C, celt_ener);
+   c=0; do
+      for (i=0;i<M*st->mode->eBands[st->start];i++)
+         X[c*N+i] = 0;
+   while (++c<C);
+   c=0; do
+      for (i=M*st->mode->eBands[effEnd];i<N;i++)
+         X[c*N+i] = 0;
+   while (++c<C);
+
+   if (data == NULL || len<=1)
+   {
+      celt_decode_lost(st, pcm, N, LM);
+      RESTORE_STACK;
+      return frame_size/st->downsample;
+   }
+
+   if (dec == NULL)
+   {
+      ec_dec_init(&_dec,(unsigned char*)data,len);
+      dec = &_dec;
+   }
+
+   if (C==1)
+   {
+      for (i=0;i<st->mode->nbEBands;i++)
+         oldBandE[i]=MAX16(oldBandE[i],oldBandE[st->mode->nbEBands+i]);
+   }
+
+   total_bits = len*8;
+   tell = ec_tell(dec);
+
+   if (tell >= total_bits)
+      silence = 1;
+   else if (tell==1)
+      silence = ec_dec_bit_logp(dec, 15);
+   else
+      silence = 0;
+   if (silence)
+   {
+      /* Pretend we've read all the remaining bits */
+      tell = len*8;
+      dec->nbits_total+=tell-ec_tell(dec);
+   }
+
+   postfilter_gain = 0;
+   postfilter_pitch = 0;
+   postfilter_tapset = 0;
+   if (st->start==0 && tell+16 <= total_bits)
+   {
+      if(ec_dec_bit_logp(dec, 1))
+      {
+         int qg, octave;
+         octave = ec_dec_uint(dec, 6);
+         postfilter_pitch = (16<<octave)+ec_dec_bits(dec, 4+octave)-1;
+         qg = ec_dec_bits(dec, 3);
+         if (ec_tell(dec)+2<=total_bits)
+            postfilter_tapset = ec_dec_icdf(dec, tapset_icdf, 2);
+         postfilter_gain = QCONST16(.09375f,15)*(qg+1);
+      }
+      tell = ec_tell(dec);
+   }
+
+   if (LM > 0 && tell+3 <= total_bits)
+   {
+      isTransient = ec_dec_bit_logp(dec, 3);
+      tell = ec_tell(dec);
+   }
+   else
+      isTransient = 0;
+
+   if (isTransient)
+      shortBlocks = M;
+   else
+      shortBlocks = 0;
+
+   /* Decode the global flags (first symbols in the stream) */
+   intra_ener = tell+3<=total_bits ? ec_dec_bit_logp(dec, 3) : 0;
+   /* Get band energies */
+   unquant_coarse_energy(st->mode, st->start, st->end, oldBandE,
+         intra_ener, dec, C, LM);
+
+   ALLOC(tf_res, st->mode->nbEBands, int);
+   tf_decode(st->start, st->end, isTransient, tf_res, LM, dec);
+
+   tell = ec_tell(dec);
+   spread_decision = SPREAD_NORMAL;
+   if (tell+4 <= total_bits)
+      spread_decision = ec_dec_icdf(dec, spread_icdf, 5);
+
+   ALLOC(pulses, st->mode->nbEBands, int);
+   ALLOC(cap, st->mode->nbEBands, int);
+   ALLOC(offsets, st->mode->nbEBands, int);
+   ALLOC(fine_priority, st->mode->nbEBands, int);
+
+   init_caps(st->mode,cap,LM,C);
+
+   dynalloc_logp = 6;
+   total_bits<<=BITRES;
+   tell = ec_tell_frac(dec);
+   for (i=st->start;i<st->end;i++)
+   {
+      int width, quanta;
+      int dynalloc_loop_logp;
+      int boost;
+      width = C*(st->mode->eBands[i+1]-st->mode->eBands[i])<<LM;
+      /* quanta is 6 bits, but no more than 1 bit/sample
+         and no less than 1/8 bit/sample */
+      quanta = IMIN(width<<BITRES, IMAX(6<<BITRES, width));
+      dynalloc_loop_logp = dynalloc_logp;
+      boost = 0;
+      while (tell+(dynalloc_loop_logp<<BITRES) < total_bits && boost < cap[i])
+      {
+         int flag;
+         flag = ec_dec_bit_logp(dec, dynalloc_loop_logp);
+         tell = ec_tell_frac(dec);
+         if (!flag)
+            break;
+         boost += quanta;
+         total_bits -= quanta;
+         dynalloc_loop_logp = 1;
+      }
+      offsets[i] = boost;
+      /* Making dynalloc more likely */
+      if (boost>0)
+         dynalloc_logp = IMAX(2, dynalloc_logp-1);
+   }
+
+   ALLOC(fine_quant, st->mode->nbEBands, int);
+   alloc_trim = tell+(6<<BITRES) <= total_bits ?
+         ec_dec_icdf(dec, trim_icdf, 7) : 5;
+
+   bits = (((opus_int32)len*8)<<BITRES) - ec_tell_frac(dec) - 1;
+   anti_collapse_rsv = isTransient&&LM>=2&&bits>=((LM+2)<<BITRES) ? (1<<BITRES) : 0;
+   bits -= anti_collapse_rsv;
+   codedBands = compute_allocation(st->mode, st->start, st->end, offsets, cap,
+         alloc_trim, &intensity, &dual_stereo, bits, &balance, pulses,
+         fine_quant, fine_priority, C, LM, dec, 0, 0);
+
+   unquant_fine_energy(st->mode, st->start, st->end, oldBandE, fine_quant, dec, C);
+
+   /* Decode fixed codebook */
+   ALLOC(collapse_masks, C*st->mode->nbEBands, unsigned char);
+   quant_all_bands(0, st->mode, st->start, st->end, X, C==2 ? X+N : NULL, collapse_masks,
+         NULL, pulses, shortBlocks, spread_decision, dual_stereo, intensity, tf_res,
+         len*(8<<BITRES)-anti_collapse_rsv, balance, dec, LM, codedBands, &st->rng);
+
+   if (anti_collapse_rsv > 0)
+   {
+      anti_collapse_on = ec_dec_bits(dec, 1);
+   }
+
+   unquant_energy_finalise(st->mode, st->start, st->end, oldBandE,
+         fine_quant, fine_priority, len*8-ec_tell(dec), dec, C);
+
+   if (anti_collapse_on)
+      anti_collapse(st->mode, X, collapse_masks, LM, C, N,
+            st->start, st->end, oldBandE, oldLogE, oldLogE2, pulses, st->rng);
+
+   log2Amp(st->mode, st->start, st->end, bandE, oldBandE, C);
+
+   if (silence)
+   {
+      for (i=0;i<C*st->mode->nbEBands;i++)
+      {
+         bandE[i] = 0;
+         oldBandE[i] = -QCONST16(28.f,DB_SHIFT);
+      }
+   }
+   /* Synthesis */
+   denormalise_bands(st->mode, X, freq, bandE, effEnd, C, M);
+
+   OPUS_MOVE(decode_mem[0], decode_mem[0]+N, DECODE_BUFFER_SIZE-N);
+   if (CC==2)
+      OPUS_MOVE(decode_mem[1], decode_mem[1]+N, DECODE_BUFFER_SIZE-N);
+
+   c=0; do
+      for (i=0;i<M*st->mode->eBands[st->start];i++)
+         freq[c*N+i] = 0;
+   while (++c<C);
+   c=0; do {
+      int bound = M*st->mode->eBands[effEnd];
+      if (st->downsample!=1)
+         bound = IMIN(bound, N/st->downsample);
+      for (i=bound;i<N;i++)
+         freq[c*N+i] = 0;
+   } while (++c<C);
+
+   out_syn[0] = out_mem[0]+MAX_PERIOD-N;
+   if (CC==2)
+      out_syn[1] = out_mem[1]+MAX_PERIOD-N;
+
+   if (CC==2&&C==1)
+   {
+      for (i=0;i<N;i++)
+         freq[N+i] = freq[i];
+   }
+   if (CC==1&&C==2)
+   {
+      for (i=0;i<N;i++)
+         freq[i] = HALF32(ADD32(freq[i],freq[N+i]));
+   }
+
+   /* Compute inverse MDCTs */
+   compute_inv_mdcts(st->mode, shortBlocks, freq, out_syn, overlap_mem, CC, LM);
+
+   c=0; do {
+      st->postfilter_period=IMAX(st->postfilter_period, COMBFILTER_MINPERIOD);
+      st->postfilter_period_old=IMAX(st->postfilter_period_old, COMBFILTER_MINPERIOD);
+      comb_filter(out_syn[c], out_syn[c], st->postfilter_period_old, st->postfilter_period, st->mode->shortMdctSize,
+            st->postfilter_gain_old, st->postfilter_gain, st->postfilter_tapset_old, st->postfilter_tapset,
+            st->mode->window, st->overlap);
+      if (LM!=0)
+         comb_filter(out_syn[c]+st->mode->shortMdctSize, out_syn[c]+st->mode->shortMdctSize, st->postfilter_period, postfilter_pitch, N-st->mode->shortMdctSize,
+               st->postfilter_gain, postfilter_gain, st->postfilter_tapset, postfilter_tapset,
+               st->mode->window, st->mode->overlap);
+
+   } while (++c<CC);
+   st->postfilter_period_old = st->postfilter_period;
+   st->postfilter_gain_old = st->postfilter_gain;
+   st->postfilter_tapset_old = st->postfilter_tapset;
+   st->postfilter_period = postfilter_pitch;
+   st->postfilter_gain = postfilter_gain;
+   st->postfilter_tapset = postfilter_tapset;
+   if (LM!=0)
+   {
+      st->postfilter_period_old = st->postfilter_period;
+      st->postfilter_gain_old = st->postfilter_gain;
+      st->postfilter_tapset_old = st->postfilter_tapset;
+   }
+
+   if (C==1) {
+      for (i=0;i<st->mode->nbEBands;i++)
+         oldBandE[st->mode->nbEBands+i]=oldBandE[i];
+   }
+
+   /* In case start or end were to change */
+   if (!isTransient)
+   {
+      for (i=0;i<2*st->mode->nbEBands;i++)
+         oldLogE2[i] = oldLogE[i];
+      for (i=0;i<2*st->mode->nbEBands;i++)
+         oldLogE[i] = oldBandE[i];
+      for (i=0;i<2*st->mode->nbEBands;i++)
+         backgroundLogE[i] = MIN16(backgroundLogE[i] + M*QCONST16(0.001f,DB_SHIFT), oldBandE[i]);
+   } else {
+      for (i=0;i<2*st->mode->nbEBands;i++)
+         oldLogE[i] = MIN16(oldLogE[i], oldBandE[i]);
+   }
+   c=0; do
+   {
+      for (i=0;i<st->start;i++)
+      {
+         oldBandE[c*st->mode->nbEBands+i]=0;
+         oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT);
+      }
+      for (i=st->end;i<st->mode->nbEBands;i++)
+      {
+         oldBandE[c*st->mode->nbEBands+i]=0;
+         oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT);
+      }
+   } while (++c<2);
+   st->rng = dec->rng;
+
+   deemphasis(out_syn, pcm, N, CC, st->downsample, st->mode->preemph, st->preemph_memD);
+   st->loss_count = 0;
+   RESTORE_STACK;
+   if (ec_tell(dec) > 8*len)
+      return OPUS_INTERNAL_ERROR;
+   if(ec_get_error(dec))
+      st->error = 1;
+   return frame_size/st->downsample;
+}
+
+
+#ifdef CUSTOM_MODES
+
+#ifdef FIXED_POINT
+int opus_custom_decode(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_int16 * OPUS_RESTRICT pcm, int frame_size)
+{
+   return celt_decode_with_ec(st, data, len, pcm, frame_size, NULL);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_custom_decode_float(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, float * OPUS_RESTRICT pcm, int frame_size)
+{
+   int j, ret, C, N;
+   VARDECL(opus_int16, out);
+   ALLOC_STACK;
+
+   if (pcm==NULL)
+      return OPUS_BAD_ARG;
+
+   C = st->channels;
+   N = frame_size;
+
+   ALLOC(out, C*N, opus_int16);
+   ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL);
+   if (ret>0)
+      for (j=0;j<C*ret;j++)
+         pcm[j]=out[j]*(1.f/32768.f);
+
+   RESTORE_STACK;
+   return ret;
+}
+#endif /* DISABLE_FLOAT_API */
+
+#else
+
+int opus_custom_decode_float(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, float * OPUS_RESTRICT pcm, int frame_size)
+{
+   return celt_decode_with_ec(st, data, len, pcm, frame_size, NULL);
+}
+
+int opus_custom_decode(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_int16 * OPUS_RESTRICT pcm, int frame_size)
+{
+   int j, ret, C, N;
+   VARDECL(celt_sig, out);
+   ALLOC_STACK;
+
+   if (pcm==NULL)
+      return OPUS_BAD_ARG;
+
+   C = st->channels;
+   N = frame_size;
+   ALLOC(out, C*N, celt_sig);
+
+   ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL);
+
+   if (ret>0)
+      for (j=0;j<C*ret;j++)
+         pcm[j] = FLOAT2INT16 (out[j]);
+
+   RESTORE_STACK;
+   return ret;
+}
+
+#endif
+#endif /* CUSTOM_MODES */
+
+int opus_custom_decoder_ctl(CELTDecoder * OPUS_RESTRICT st, int request, ...)
+{
+   va_list ap;
+
+   va_start(ap, request);
+   switch (request)
+   {
+      case CELT_SET_START_BAND_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<0 || value>=st->mode->nbEBands)
+            goto bad_arg;
+         st->start = value;
+      }
+      break;
+      case CELT_SET_END_BAND_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<1 || value>st->mode->nbEBands)
+            goto bad_arg;
+         st->end = value;
+      }
+      break;
+      case CELT_SET_CHANNELS_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         if (value<1 || value>2)
+            goto bad_arg;
+         st->stream_channels = value;
+      }
+      break;
+      case CELT_GET_AND_CLEAR_ERROR_REQUEST:
+      {
+         opus_int32 *value = va_arg(ap, opus_int32*);
+         if (value==NULL)
+            goto bad_arg;
+         *value=st->error;
+         st->error = 0;
+      }
+      break;
+      case OPUS_GET_LOOKAHEAD_REQUEST:
+      {
+         opus_int32 *value = va_arg(ap, opus_int32*);
+         if (value==NULL)
+            goto bad_arg;
+         *value = st->overlap/st->downsample;
+      }
+      break;
+      case OPUS_RESET_STATE:
+      {
+         int i;
+         opus_val16 *lpc, *oldBandE, *oldLogE, *oldLogE2;
+         lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*st->channels);
+         oldBandE = lpc+st->channels*LPC_ORDER;
+         oldLogE = oldBandE + 2*st->mode->nbEBands;
+         oldLogE2 = oldLogE + 2*st->mode->nbEBands;
+         OPUS_CLEAR((char*)&st->DECODER_RESET_START,
+               opus_custom_decoder_get_size(st->mode, st->channels)-
+               ((char*)&st->DECODER_RESET_START - (char*)st));
+         for (i=0;i<2*st->mode->nbEBands;i++)
+            oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT);
+      }
+      break;
+      case OPUS_GET_PITCH_REQUEST:
+      {
+         opus_int32 *value = va_arg(ap, opus_int32*);
+         if (value==NULL)
+            goto bad_arg;
+         *value = st->postfilter_period;
+      }
+      break;
+#ifdef OPUS_BUILD
+      case CELT_GET_MODE_REQUEST:
+      {
+         const CELTMode ** value = va_arg(ap, const CELTMode**);
+         if (value==0)
+            goto bad_arg;
+         *value=st->mode;
+      }
+      break;
+      case CELT_SET_SIGNALLING_REQUEST:
+      {
+         opus_int32 value = va_arg(ap, opus_int32);
+         st->signalling = value;
+      }
+      break;
+      case OPUS_GET_FINAL_RANGE_REQUEST:
+      {
+         opus_uint32 * value = va_arg(ap, opus_uint32 *);
+         if (value==0)
+            goto bad_arg;
+         *value=st->rng;
+      }
+      break;
+#endif
+      default:
+         goto bad_request;
+   }
+   va_end(ap);
+   return OPUS_OK;
+bad_arg:
+   va_end(ap);
+   return OPUS_BAD_ARG;
+bad_request:
+      va_end(ap);
+  return OPUS_UNIMPLEMENTED;
+}
+
+
+
+const char *opus_strerror(int error)
+{
+   static const char * const error_strings[8] = {
+      "success",
+      "invalid argument",
+      "buffer too small",
+      "internal error",
+      "corrupted stream",
+      "request not implemented",
+      "invalid state",
+      "memory allocation failed"
+   };
+   if (error > 0 || error < -7)
+      return "unknown error";
+   else
+      return error_strings[-error];
+}
+
+const char *opus_get_version_string(void)
+{
+    return "libopus " OPUS_VERSION
+#ifdef FIXED_POINT
+          "-fixed"
+#endif
+#ifdef FUZZING
+          "-fuzzing"
+#endif
+          ;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/celt.h b/lib/rbcodec/codecs/libopus/celt/celt.h
new file mode 100644
index 0000000000..218cd883df
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/celt.h
@@ -0,0 +1,117 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/**
+  @file celt.h
+  @brief Contains all the functions for encoding and decoding audio
+ */
+
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef CELT_H
+#define CELT_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+#include "opus_custom.h"
+#include "entenc.h"
+#include "entdec.h"
+#include "arch.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define CELTEncoder OpusCustomEncoder
+#define CELTDecoder OpusCustomDecoder
+#define CELTMode OpusCustomMode
+
+#define _celt_check_mode_ptr_ptr(ptr) ((ptr) + ((ptr) - (const CELTMode**)(ptr)))
+
+/* Encoder/decoder Requests */
+
+#define CELT_SET_PREDICTION_REQUEST    10002
+/** Controls the use of interframe prediction.
+    0=Independent frames
+    1=Short term interframe prediction allowed
+    2=Long term prediction allowed
+ */
+#define CELT_SET_PREDICTION(x) CELT_SET_PREDICTION_REQUEST, __opus_check_int(x)
+
+#define CELT_SET_INPUT_CLIPPING_REQUEST    10004
+#define CELT_SET_INPUT_CLIPPING(x) CELT_SET_INPUT_CLIPPING_REQUEST, __opus_check_int(x)
+
+#define CELT_GET_AND_CLEAR_ERROR_REQUEST   10007
+#define CELT_GET_AND_CLEAR_ERROR(x) CELT_GET_AND_CLEAR_ERROR_REQUEST, __opus_check_int_ptr(x)
+
+#define CELT_SET_CHANNELS_REQUEST    10008
+#define CELT_SET_CHANNELS(x) CELT_SET_CHANNELS_REQUEST, __opus_check_int(x)
+
+
+/* Internal */
+#define CELT_SET_START_BAND_REQUEST    10010
+#define CELT_SET_START_BAND(x) CELT_SET_START_BAND_REQUEST, __opus_check_int(x)
+
+#define CELT_SET_END_BAND_REQUEST    10012
+#define CELT_SET_END_BAND(x) CELT_SET_END_BAND_REQUEST, __opus_check_int(x)
+
+#define CELT_GET_MODE_REQUEST    10015
+/** Get the CELTMode used by an encoder or decoder */
+#define CELT_GET_MODE(x) CELT_GET_MODE_REQUEST, _celt_check_mode_ptr_ptr(x)
+
+#define CELT_SET_SIGNALLING_REQUEST    10016
+#define CELT_SET_SIGNALLING(x) CELT_SET_SIGNALLING_REQUEST, __opus_check_int(x)
+
+
+
+/* Encoder stuff */
+
+int celt_encoder_get_size(int channels);
+
+int celt_encode_with_ec(OpusCustomEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc);
+
+int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels);
+
+
+
+/* Decoder stuff */
+
+int celt_decoder_get_size(int channels);
+
+
+int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels);
+
+int celt_decode_with_ec(OpusCustomDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec);
+
+#define celt_encoder_ctl opus_custom_encoder_ctl
+#define celt_decoder_ctl opus_custom_decoder_ctl
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* CELT_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/celt_lpc.c b/lib/rbcodec/codecs/libopus/celt/celt_lpc.c
new file mode 100644
index 0000000000..66aed1de09
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/celt_lpc.c
@@ -0,0 +1,188 @@
+/* Copyright (c) 2009-2010 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "celt_lpc.h"
+#include "stack_alloc.h"
+#include "mathops.h"
+
+void _celt_lpc(
+      opus_val16       *_lpc, /* out: [0...p-1] LPC coefficients      */
+const opus_val32 *ac,  /* in:  [0...p] autocorrelation values  */
+int          p
+)
+{
+   int i, j;
+   opus_val32 r;
+   opus_val32 error = ac[0];
+#ifdef FIXED_POINT
+   opus_val32 lpc[LPC_ORDER];
+#else
+   float *lpc = _lpc;
+#endif
+
+   for (i = 0; i < p; i++)
+      lpc[i] = 0;
+   if (ac[0] != 0)
+   {
+      for (i = 0; i < p; i++) {
+         /* Sum up this iteration's reflection coefficient */
+         opus_val32 rr = 0;
+         for (j = 0; j < i; j++)
+            rr += MULT32_32_Q31(lpc[j],ac[i - j]);
+         rr += SHR32(ac[i + 1],3);
+         r = -frac_div32(SHL32(rr,3), error);
+         /*  Update LPC coefficients and total error */
+         lpc[i] = SHR32(r,3);
+         for (j = 0; j < (i+1)>>1; j++)
+         {
+            opus_val32 tmp1, tmp2;
+            tmp1 = lpc[j];
+            tmp2 = lpc[i-1-j];
+            lpc[j]     = tmp1 + MULT32_32_Q31(r,tmp2);
+            lpc[i-1-j] = tmp2 + MULT32_32_Q31(r,tmp1);
+         }
+
+         error = error - MULT32_32_Q31(MULT32_32_Q31(r,r),error);
+         /* Bail out once we get 30 dB gain */
+#ifdef FIXED_POINT
+         if (error<SHR32(ac[0],10))
+            break;
+#else
+         if (error<.001f*ac[0])
+            break;
+#endif
+      }
+   }
+#ifdef FIXED_POINT
+   for (i=0;i<p;i++)
+      _lpc[i] = ROUND16(lpc[i],16);
+#endif
+}
+
+void celt_fir(const opus_val16 *x,
+         const opus_val16 *num,
+         opus_val16 *y,
+         int N,
+         int ord,
+         opus_val16 *mem)
+{
+   int i,j;
+
+   for (i=0;i<N;i++)
+   {
+      opus_val32 sum = SHL32(EXTEND32(x[i]), SIG_SHIFT);
+      for (j=0;j<ord;j++)
+      {
+         sum += MULT16_16(num[j],mem[j]);
+      }
+      for (j=ord-1;j>=1;j--)
+      {
+         mem[j]=mem[j-1];
+      }
+      mem[0] = x[i];
+      y[i] = ROUND16(sum, SIG_SHIFT);
+   }
+}
+
+void celt_iir(const opus_val32 *x,
+         const opus_val16 *den,
+         opus_val32 *y,
+         int N,
+         int ord,
+         opus_val16 *mem)
+{
+   int i,j;
+   for (i=0;i<N;i++)
+   {
+      opus_val32 sum = x[i];
+      for (j=0;j<ord;j++)
+      {
+         sum -= MULT16_16(den[j],mem[j]);
+      }
+      for (j=ord-1;j>=1;j--)
+      {
+         mem[j]=mem[j-1];
+      }
+      mem[0] = ROUND16(sum,SIG_SHIFT);
+      y[i] = sum;
+   }
+}
+
+void _celt_autocorr(
+                   const opus_val16 *x,   /*  in: [0...n-1] samples x   */
+                   opus_val32       *ac,  /* out: [0...lag-1] ac values */
+                   const opus_val16       *window,
+                   int          overlap,
+                   int          lag,
+                   int          n
+                  )
+{
+   opus_val32 d;
+   int i;
+   VARDECL(opus_val16, xx);
+   SAVE_STACK;
+   ALLOC(xx, n, opus_val16);
+   celt_assert(n>0);
+   celt_assert(overlap>=0);
+   for (i=0;i<n;i++)
+      xx[i] = x[i];
+   for (i=0;i<overlap;i++)
+   {
+      xx[i] = MULT16_16_Q15(x[i],window[i]);
+      xx[n-i-1] = MULT16_16_Q15(x[n-i-1],window[i]);
+   }
+#ifdef FIXED_POINT
+   {
+      opus_val32 ac0=0;
+      int shift;
+      for(i=0;i<n;i++)
+         ac0 += SHR32(MULT16_16(xx[i],xx[i]),9);
+      ac0 += 1+n;
+
+      shift = celt_ilog2(ac0)-30+10;
+      shift = (shift+1)/2;
+      for(i=0;i<n;i++)
+         xx[i] = VSHR32(xx[i], shift);
+   }
+#endif
+   while (lag>=0)
+   {
+      for (i = lag, d = 0; i < n; i++)
+         d += xx[i] * xx[i-lag];
+      ac[lag] = d;
+      /*printf ("%f ", ac[lag]);*/
+      lag--;
+   }
+   /*printf ("\n");*/
+   ac[0] += 10;
+
+   RESTORE_STACK;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/celt_lpc.h b/lib/rbcodec/codecs/libopus/celt/celt_lpc.h
new file mode 100644
index 0000000000..2baa77edf8
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/celt_lpc.h
@@ -0,0 +1,53 @@
+/* Copyright (c) 2009-2010 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef PLC_H
+#define PLC_H
+
+#include "arch.h"
+
+#define LPC_ORDER 24
+
+void _celt_lpc(opus_val16 *_lpc, const opus_val32 *ac, int p);
+
+void celt_fir(const opus_val16 *x,
+         const opus_val16 *num,
+         opus_val16 *y,
+         int N,
+         int ord,
+         opus_val16 *mem);
+
+void celt_iir(const opus_val32 *x,
+         const opus_val16 *den,
+         opus_val32 *y,
+         int N,
+         int ord,
+         opus_val16 *mem);
+
+void _celt_autocorr(const opus_val16 *x, opus_val32 *ac, const opus_val16 *window, int overlap, int lag, int n);
+
+#endif /* PLC_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/cwrs.c b/lib/rbcodec/codecs/libopus/celt/cwrs.c
new file mode 100644
index 0000000000..3d5dd790d9
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/cwrs.c
@@ -0,0 +1,645 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2007-2009 Timothy B. Terriberry
+   Written by Timothy B. Terriberry and Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "os_support.h"
+#include "cwrs.h"
+#include "mathops.h"
+#include "arch.h"
+
+#ifdef CUSTOM_MODES
+
+/*Guaranteed to return a conservatively large estimate of the binary logarithm
+   with frac bits of fractional precision.
+  Tested for all possible 32-bit inputs with frac=4, where the maximum
+   overestimation is 0.06254243 bits.*/
+int log2_frac(opus_uint32 val, int frac)
+{
+  int l;
+  l=EC_ILOG(val);
+  if(val&(val-1)){
+    /*This is (val>>l-16), but guaranteed to round up, even if adding a bias
+       before the shift would cause overflow (e.g., for 0xFFFFxxxx).
+       Doesn't work for val=0, but that case fails the test above.*/
+    if(l>16)val=((val-1)>>(l-16))+1;
+    else val<<=16-l;
+    l=(l-1)<<frac;
+    /*Note that we always need one iteration, since the rounding up above means
+       that we might need to adjust the integer part of the logarithm.*/
+    do{
+      int b;
+      b=(int)(val>>16);
+      l+=b<<frac;
+      val=(val+b)>>b;
+      val=(val*val+0x7FFF)>>15;
+    }
+    while(frac-->0);
+    /*If val is not exactly 0x8000, then we have to round up the remainder.*/
+    return l+(val>0x8000);
+  }
+  /*Exact powers of two require no rounding.*/
+  else return (l-1)<<frac;
+}
+#endif
+
+#ifndef SMALL_FOOTPRINT
+
+#define MASK32 (0xFFFFFFFF)
+
+/*INV_TABLE[i] holds the multiplicative inverse of (2*i+1) mod 2**32.*/
+static const opus_uint32 INV_TABLE[53]={
+  0x00000001,0xAAAAAAAB,0xCCCCCCCD,0xB6DB6DB7,
+  0x38E38E39,0xBA2E8BA3,0xC4EC4EC5,0xEEEEEEEF,
+  0xF0F0F0F1,0x286BCA1B,0x3CF3CF3D,0xE9BD37A7,
+  0xC28F5C29,0x684BDA13,0x4F72C235,0xBDEF7BDF,
+  0x3E0F83E1,0x8AF8AF8B,0x914C1BAD,0x96F96F97,
+  0xC18F9C19,0x2FA0BE83,0xA4FA4FA5,0x677D46CF,
+  0x1A1F58D1,0xFAFAFAFB,0x8C13521D,0x586FB587,
+  0xB823EE09,0xA08AD8F3,0xC10C9715,0xBEFBEFBF,
+  0xC0FC0FC1,0x07A44C6B,0xA33F128D,0xE327A977,
+  0xC7E3F1F9,0x962FC963,0x3F2B3885,0x613716AF,
+  0x781948B1,0x2B2E43DB,0xFCFCFCFD,0x6FD0EB67,
+  0xFA3F47E9,0xD2FD2FD3,0x3F4FD3F5,0xD4E25B9F,
+  0x5F02A3A1,0xBF5A814B,0x7C32B16D,0xD3431B57,
+  0xD8FD8FD9,
+};
+
+/*Computes (_a*_b-_c)/(2*_d+1) when the quotient is known to be exact.
+  _a, _b, _c, and _d may be arbitrary so long as the arbitrary precision result
+   fits in 32 bits, but currently the table for multiplicative inverses is only
+   valid for _d<=52.*/
+static inline opus_uint32 imusdiv32odd(opus_uint32 _a,opus_uint32 _b,
+ opus_uint32 _c,int _d){
+  celt_assert(_d<=52);
+  return (_a*_b-_c)*INV_TABLE[_d]&MASK32;
+}
+
+/*Computes (_a*_b-_c)/_d when the quotient is known to be exact.
+  _d does not actually have to be even, but imusdiv32odd will be faster when
+   it's odd, so you should use that instead.
+  _a and _d are assumed to be small (e.g., _a*_d fits in 32 bits; currently the
+   table for multiplicative inverses is only valid for _d<=54).
+  _b and _c may be arbitrary so long as the arbitrary precision reuslt fits in
+   32 bits.*/
+static inline opus_uint32 imusdiv32even(opus_uint32 _a,opus_uint32 _b,
+ opus_uint32 _c,int _d){
+  opus_uint32 inv;
+  int           mask;
+  int           shift;
+  int           one;
+  celt_assert(_d>0);
+  celt_assert(_d<=54);
+  shift=EC_ILOG(_d^(_d-1));
+  inv=INV_TABLE[(_d-1)>>shift];
+  shift--;
+  one=1<<shift;
+  mask=one-1;
+  return (_a*(_b>>shift)-(_c>>shift)+
+   ((_a*(_b&mask)+one-(_c&mask))>>shift)-1)*inv&MASK32;
+}
+
+#endif /* SMALL_FOOTPRINT */
+
+/*Although derived separately, the pulse vector coding scheme is equivalent to
+   a Pyramid Vector Quantizer \cite{Fis86}.
+  Some additional notes about an early version appear at
+   http://people.xiph.org/~tterribe/notes/cwrs.html, but the codebook ordering
+   and the definitions of some terms have evolved since that was written.
+
+  The conversion from a pulse vector to an integer index (encoding) and back
+   (decoding) is governed by two related functions, V(N,K) and U(N,K).
+
+  V(N,K) = the number of combinations, with replacement, of N items, taken K
+   at a time, when a sign bit is added to each item taken at least once (i.e.,
+   the number of N-dimensional unit pulse vectors with K pulses).
+  One way to compute this is via
+    V(N,K) = K>0 ? sum(k=1...K,2**k*choose(N,k)*choose(K-1,k-1)) : 1,
+   where choose() is the binomial function.
+  A table of values for N<10 and K<10 looks like:
+  V[10][10] = {
+    {1,  0,   0,    0,    0,     0,     0,      0,      0,       0},
+    {1,  2,   2,    2,    2,     2,     2,      2,      2,       2},
+    {1,  4,   8,   12,   16,    20,    24,     28,     32,      36},
+    {1,  6,  18,   38,   66,   102,   146,    198,    258,     326},
+    {1,  8,  32,   88,  192,   360,   608,    952,   1408,    1992},
+    {1, 10,  50,  170,  450,  1002,  1970,   3530,   5890,    9290},
+    {1, 12,  72,  292,  912,  2364,  5336,  10836,  20256,   35436},
+    {1, 14,  98,  462, 1666,  4942, 12642,  28814,  59906,  115598},
+    {1, 16, 128,  688, 2816,  9424, 27008,  68464, 157184,  332688},
+    {1, 18, 162,  978, 4482, 16722, 53154, 148626, 374274,  864146}
+  };
+
+  U(N,K) = the number of such combinations wherein N-1 objects are taken at
+   most K-1 at a time.
+  This is given by
+    U(N,K) = sum(k=0...K-1,V(N-1,k))
+           = K>0 ? (V(N-1,K-1) + V(N,K-1))/2 : 0.
+  The latter expression also makes clear that U(N,K) is half the number of such
+   combinations wherein the first object is taken at least once.
+  Although it may not be clear from either of these definitions, U(N,K) is the
+   natural function to work with when enumerating the pulse vector codebooks,
+   not V(N,K).
+  U(N,K) is not well-defined for N=0, but with the extension
+    U(0,K) = K>0 ? 0 : 1,
+   the function becomes symmetric: U(N,K) = U(K,N), with a similar table:
+  U[10][10] = {
+    {1, 0,  0,   0,    0,    0,     0,     0,      0,      0},
+    {0, 1,  1,   1,    1,    1,     1,     1,      1,      1},
+    {0, 1,  3,   5,    7,    9,    11,    13,     15,     17},
+    {0, 1,  5,  13,   25,   41,    61,    85,    113,    145},
+    {0, 1,  7,  25,   63,  129,   231,   377,    575,    833},
+    {0, 1,  9,  41,  129,  321,   681,  1289,   2241,   3649},
+    {0, 1, 11,  61,  231,  681,  1683,  3653,   7183,  13073},
+    {0, 1, 13,  85,  377, 1289,  3653,  8989,  19825,  40081},
+    {0, 1, 15, 113,  575, 2241,  7183, 19825,  48639, 108545},
+    {0, 1, 17, 145,  833, 3649, 13073, 40081, 108545, 265729}
+  };
+
+  With this extension, V(N,K) may be written in terms of U(N,K):
+    V(N,K) = U(N,K) + U(N,K+1)
+   for all N>=0, K>=0.
+  Thus U(N,K+1) represents the number of combinations where the first element
+   is positive or zero, and U(N,K) represents the number of combinations where
+   it is negative.
+  With a large enough table of U(N,K) values, we could write O(N) encoding
+   and O(min(N*log(K),N+K)) decoding routines, but such a table would be
+   prohibitively large for small embedded devices (K may be as large as 32767
+   for small N, and N may be as large as 200).
+
+  Both functions obey the same recurrence relation:
+    V(N,K) = V(N-1,K) + V(N,K-1) + V(N-1,K-1),
+    U(N,K) = U(N-1,K) + U(N,K-1) + U(N-1,K-1),
+   for all N>0, K>0, with different initial conditions at N=0 or K=0.
+  This allows us to construct a row of one of the tables above given the
+   previous row or the next row.
+  Thus we can derive O(NK) encoding and decoding routines with O(K) memory
+   using only addition and subtraction.
+
+  When encoding, we build up from the U(2,K) row and work our way forwards.
+  When decoding, we need to start at the U(N,K) row and work our way backwards,
+   which requires a means of computing U(N,K).
+  U(N,K) may be computed from two previous values with the same N:
+    U(N,K) = ((2*N-1)*U(N,K-1) - U(N,K-2))/(K-1) + U(N,K-2)
+   for all N>1, and since U(N,K) is symmetric, a similar relation holds for two
+   previous values with the same K:
+    U(N,K>1) = ((2*K-1)*U(N-1,K) - U(N-2,K))/(N-1) + U(N-2,K)
+   for all K>1.
+  This allows us to construct an arbitrary row of the U(N,K) table by starting
+   with the first two values, which are constants.
+  This saves roughly 2/3 the work in our O(NK) decoding routine, but costs O(K)
+   multiplications.
+  Similar relations can be derived for V(N,K), but are not used here.
+
+  For N>0 and K>0, U(N,K) and V(N,K) take on the form of an (N-1)-degree
+   polynomial for fixed N.
+  The first few are
+    U(1,K) = 1,
+    U(2,K) = 2*K-1,
+    U(3,K) = (2*K-2)*K+1,
+    U(4,K) = (((4*K-6)*K+8)*K-3)/3,
+    U(5,K) = ((((2*K-4)*K+10)*K-8)*K+3)/3,
+   and
+    V(1,K) = 2,
+    V(2,K) = 4*K,
+    V(3,K) = 4*K*K+2,
+    V(4,K) = 8*(K*K+2)*K/3,
+    V(5,K) = ((4*K*K+20)*K*K+6)/3,
+   for all K>0.
+  This allows us to derive O(N) encoding and O(N*log(K)) decoding routines for
+   small N (and indeed decoding is also O(N) for N<3).
+
+  @ARTICLE{Fis86,
+    author="Thomas R. Fischer",
+    title="A Pyramid Vector Quantizer",
+    journal="IEEE Transactions on Information Theory",
+    volume="IT-32",
+    number=4,
+    pages="568--583",
+    month=Jul,
+    year=1986
+  }*/
+
+#ifndef SMALL_FOOTPRINT
+/*Compute U(2,_k).
+  Note that this may be called with _k=32768 (maxK[2]+1).*/
+static inline unsigned ucwrs2(unsigned _k){
+  celt_assert(_k>0);
+  return _k+(_k-1);
+}
+
+/*Compute V(2,_k).*/
+static inline opus_uint32 ncwrs2(int _k){
+  celt_assert(_k>0);
+  return 4*(opus_uint32)_k;
+}
+
+/*Compute U(3,_k).
+  Note that this may be called with _k=32768 (maxK[3]+1).*/
+static inline opus_uint32 ucwrs3(unsigned _k){
+  celt_assert(_k>0);
+  return (2*(opus_uint32)_k-2)*_k+1;
+}
+
+/*Compute V(3,_k).*/
+static inline opus_uint32 ncwrs3(int _k){
+  celt_assert(_k>0);
+  return 2*(2*(unsigned)_k*(opus_uint32)_k+1);
+}
+
+/*Compute U(4,_k).*/
+static inline opus_uint32 ucwrs4(int _k){
+  celt_assert(_k>0);
+  return imusdiv32odd(2*_k,(2*_k-3)*(opus_uint32)_k+4,3,1);
+}
+
+/*Compute V(4,_k).*/
+static inline opus_uint32 ncwrs4(int _k){
+  celt_assert(_k>0);
+  return ((_k*(opus_uint32)_k+2)*_k)/3<<3;
+}
+
+#endif /* SMALL_FOOTPRINT */
+
+/*Computes the next row/column of any recurrence that obeys the relation
+   u[i][j]=u[i-1][j]+u[i][j-1]+u[i-1][j-1].
+  _ui0 is the base case for the new row/column.*/
+static inline void unext(opus_uint32 *_ui,unsigned _len,opus_uint32 _ui0){
+  opus_uint32 ui1;
+  unsigned      j;
+  /*This do-while will overrun the array if we don't have storage for at least
+     2 values.*/
+  j=1; do {
+    ui1=UADD32(UADD32(_ui[j],_ui[j-1]),_ui0);
+    _ui[j-1]=_ui0;
+    _ui0=ui1;
+  } while (++j<_len);
+  _ui[j-1]=_ui0;
+}
+
+/*Computes the previous row/column of any recurrence that obeys the relation
+   u[i-1][j]=u[i][j]-u[i][j-1]-u[i-1][j-1].
+  _ui0 is the base case for the new row/column.*/
+static inline void uprev(opus_uint32 *_ui,unsigned _n,opus_uint32 _ui0){
+  opus_uint32 ui1;
+  unsigned      j;
+  /*This do-while will overrun the array if we don't have storage for at least
+     2 values.*/
+  j=1; do {
+    ui1=USUB32(USUB32(_ui[j],_ui[j-1]),_ui0);
+    _ui[j-1]=_ui0;
+    _ui0=ui1;
+  } while (++j<_n);
+  _ui[j-1]=_ui0;
+}
+
+/*Compute V(_n,_k), as well as U(_n,0..._k+1).
+  _u: On exit, _u[i] contains U(_n,i) for i in [0..._k+1].*/
+static opus_uint32 ncwrs_urow(unsigned _n,unsigned _k,opus_uint32 *_u){
+  opus_uint32 um2;
+  unsigned      len;
+  unsigned      k;
+  len=_k+2;
+  /*We require storage at least 3 values (e.g., _k>0).*/
+  celt_assert(len>=3);
+  _u[0]=0;
+  _u[1]=um2=1;
+#ifndef SMALL_FOOTPRINT
+  /*_k>52 doesn't work in the false branch due to the limits of INV_TABLE,
+    but _k isn't tested here because k<=52 for n=7*/
+  if(_n<=6)
+#endif
+ {
+    /*If _n==0, _u[0] should be 1 and the rest should be 0.*/
+    /*If _n==1, _u[i] should be 1 for i>1.*/
+    celt_assert(_n>=2);
+    /*If _k==0, the following do-while loop will overflow the buffer.*/
+    celt_assert(_k>0);
+    k=2;
+    do _u[k]=(k<<1)-1;
+    while(++k<len);
+    for(k=2;k<_n;k++)unext(_u+1,_k+1,1);
+  }
+#ifndef SMALL_FOOTPRINT
+  else{
+    opus_uint32 um1;
+    opus_uint32 n2m1;
+    _u[2]=n2m1=um1=(_n<<1)-1;
+    for(k=3;k<len;k++){
+      /*U(N,K) = ((2*N-1)*U(N,K-1)-U(N,K-2))/(K-1) + U(N,K-2)*/
+      _u[k]=um2=imusdiv32even(n2m1,um1,um2,k-1)+um2;
+      if(++k>=len)break;
+      _u[k]=um1=imusdiv32odd(n2m1,um2,um1,(k-1)>>1)+um1;
+    }
+  }
+#endif /* SMALL_FOOTPRINT */
+  return _u[_k]+_u[_k+1];
+}
+
+#ifndef SMALL_FOOTPRINT
+
+/*Returns the _i'th combination of _k elements (at most 32767) chosen from a
+   set of size 1 with associated sign bits.
+  _y: Returns the vector of pulses.*/
+static inline void cwrsi1(int _k,opus_uint32 _i,int *_y){
+  int s;
+  s=-(int)_i;
+  _y[0]=(_k+s)^s;
+}
+
+/*Returns the _i'th combination of _k elements (at most 32767) chosen from a
+   set of size 2 with associated sign bits.
+  _y: Returns the vector of pulses.*/
+static inline void cwrsi2(int _k,opus_uint32 _i,int *_y){
+  opus_uint32 p;
+  int           s;
+  int           yj;
+  p=ucwrs2(_k+1U);
+  s=-(_i>=p);
+  _i-=p&s;
+  yj=_k;
+  _k=(_i+1)>>1;
+  p=_k?ucwrs2(_k):0;
+  _i-=p;
+  yj-=_k;
+  _y[0]=(yj+s)^s;
+  cwrsi1(_k,_i,_y+1);
+}
+
+/*Returns the _i'th combination of _k elements (at most 32767) chosen from a
+   set of size 3 with associated sign bits.
+  _y: Returns the vector of pulses.*/
+static void cwrsi3(int _k,opus_uint32 _i,int *_y){
+  opus_uint32 p;
+  int           s;
+  int           yj;
+  p=ucwrs3(_k+1U);
+  s=-(_i>=p);
+  _i-=p&s;
+  yj=_k;
+  /*Finds the maximum _k such that ucwrs3(_k)<=_i (tested for all
+     _i<2147418113=U(3,32768)).*/
+  _k=_i>0?(isqrt32(2*_i-1)+1)>>1:0;
+  p=_k?ucwrs3(_k):0;
+  _i-=p;
+  yj-=_k;
+  _y[0]=(yj+s)^s;
+  cwrsi2(_k,_i,_y+1);
+}
+
+/*Returns the _i'th combination of _k elements (at most 1172) chosen from a set
+   of size 4 with associated sign bits.
+  _y: Returns the vector of pulses.*/
+static void cwrsi4(int _k,opus_uint32 _i,int *_y){
+  opus_uint32 p;
+  int           s;
+  int           yj;
+  int           kl;
+  int           kr;
+  p=ucwrs4(_k+1);
+  s=-(_i>=p);
+  _i-=p&s;
+  yj=_k;
+  /*We could solve a cubic for k here, but the form of the direct solution does
+     not lend itself well to exact integer arithmetic.
+    Instead we do a binary search on U(4,K).*/
+  kl=0;
+  kr=_k;
+  for(;;){
+    _k=(kl+kr)>>1;
+    p=_k?ucwrs4(_k):0;
+    if(p<_i){
+      if(_k>=kr)break;
+      kl=_k+1;
+    }
+    else if(p>_i)kr=_k-1;
+    else break;
+  }
+  _i-=p;
+  yj-=_k;
+  _y[0]=(yj+s)^s;
+  cwrsi3(_k,_i,_y+1);
+}
+
+#endif /* SMALL_FOOTPRINT */
+
+/*Returns the _i'th combination of _k elements chosen from a set of size _n
+   with associated sign bits.
+  _y: Returns the vector of pulses.
+  _u: Must contain entries [0..._k+1] of row _n of U() on input.
+      Its contents will be destructively modified.*/
+static void cwrsi(int _n,int _k,opus_uint32 _i,int *_y,opus_uint32 *_u){
+  int j;
+  celt_assert(_n>0);
+  j=0;
+  do{
+    opus_uint32 p;
+    int           s;
+    int           yj;
+    p=_u[_k+1];
+    s=-(_i>=p);
+    _i-=p&s;
+    yj=_k;
+    p=_u[_k];
+    while(p>_i)p=_u[--_k];
+    _i-=p;
+    yj-=_k;
+    _y[j]=(yj+s)^s;
+    uprev(_u,_k+2,0);
+  }
+  while(++j<_n);
+}
+
+/*Returns the index of the given combination of K elements chosen from a set
+   of size 1 with associated sign bits.
+  _y: The vector of pulses, whose sum of absolute values is K.
+  _k: Returns K.*/
+static inline opus_uint32 icwrs1(const int *_y,int *_k){
+  *_k=abs(_y[0]);
+  return _y[0]<0;
+}
+
+#ifndef SMALL_FOOTPRINT
+
+/*Returns the index of the given combination of K elements chosen from a set
+   of size 2 with associated sign bits.
+  _y: The vector of pulses, whose sum of absolute values is K.
+  _k: Returns K.*/
+static inline opus_uint32 icwrs2(const int *_y,int *_k){
+  opus_uint32 i;
+  int           k;
+  i=icwrs1(_y+1,&k);
+  i+=k?ucwrs2(k):0;
+  k+=abs(_y[0]);
+  if(_y[0]<0)i+=ucwrs2(k+1U);
+  *_k=k;
+  return i;
+}
+
+/*Returns the index of the given combination of K elements chosen from a set
+   of size 3 with associated sign bits.
+  _y: The vector of pulses, whose sum of absolute values is K.
+  _k: Returns K.*/
+static inline opus_uint32 icwrs3(const int *_y,int *_k){
+  opus_uint32 i;
+  int           k;
+  i=icwrs2(_y+1,&k);
+  i+=k?ucwrs3(k):0;
+  k+=abs(_y[0]);
+  if(_y[0]<0)i+=ucwrs3(k+1U);
+  *_k=k;
+  return i;
+}
+
+/*Returns the index of the given combination of K elements chosen from a set
+   of size 4 with associated sign bits.
+  _y: The vector of pulses, whose sum of absolute values is K.
+  _k: Returns K.*/
+static inline opus_uint32 icwrs4(const int *_y,int *_k){
+  opus_uint32 i;
+  int           k;
+  i=icwrs3(_y+1,&k);
+  i+=k?ucwrs4(k):0;
+  k+=abs(_y[0]);
+  if(_y[0]<0)i+=ucwrs4(k+1);
+  *_k=k;
+  return i;
+}
+
+#endif /* SMALL_FOOTPRINT */
+
+/*Returns the index of the given combination of K elements chosen from a set
+   of size _n with associated sign bits.
+  _y:  The vector of pulses, whose sum of absolute values must be _k.
+  _nc: Returns V(_n,_k).*/
+static inline opus_uint32 icwrs(int _n,int _k,opus_uint32 *_nc,const int *_y,
+ opus_uint32 *_u){
+  opus_uint32 i;
+  int           j;
+  int           k;
+  /*We can't unroll the first two iterations of the loop unless _n>=2.*/
+  celt_assert(_n>=2);
+  _u[0]=0;
+  for(k=1;k<=_k+1;k++)_u[k]=(k<<1)-1;
+  i=icwrs1(_y+_n-1,&k);
+  j=_n-2;
+  i+=_u[k];
+  k+=abs(_y[j]);
+  if(_y[j]<0)i+=_u[k+1];
+  while(j-->0){
+    unext(_u,_k+2,0);
+    i+=_u[k];
+    k+=abs(_y[j]);
+    if(_y[j]<0)i+=_u[k+1];
+  }
+  *_nc=_u[k]+_u[k+1];
+  return i;
+}
+
+#ifdef CUSTOM_MODES
+void get_required_bits(opus_int16 *_bits,int _n,int _maxk,int _frac){
+  int k;
+  /*_maxk==0 => there's nothing to do.*/
+  celt_assert(_maxk>0);
+  _bits[0]=0;
+  if (_n==1)
+  {
+    for (k=1;k<=_maxk;k++)
+      _bits[k] = 1<<_frac;
+  }
+  else {
+    VARDECL(opus_uint32,u);
+    SAVE_STACK;
+    ALLOC(u,_maxk+2U,opus_uint32);
+    ncwrs_urow(_n,_maxk,u);
+    for(k=1;k<=_maxk;k++)
+      _bits[k]=log2_frac(u[k]+u[k+1],_frac);
+    RESTORE_STACK;
+  }
+}
+#endif /* CUSTOM_MODES */
+
+void encode_pulses(const int *_y,int _n,int _k,ec_enc *_enc){
+  opus_uint32 i;
+  celt_assert(_k>0);
+#ifndef SMALL_FOOTPRINT
+  switch(_n){
+    case 2:{
+      i=icwrs2(_y,&_k);
+      ec_enc_uint(_enc,i,ncwrs2(_k));
+    }break;
+    case 3:{
+      i=icwrs3(_y,&_k);
+      ec_enc_uint(_enc,i,ncwrs3(_k));
+    }break;
+    case 4:{
+      i=icwrs4(_y,&_k);
+      ec_enc_uint(_enc,i,ncwrs4(_k));
+    }break;
+     default:
+    {
+#endif
+      VARDECL(opus_uint32,u);
+      opus_uint32 nc;
+      SAVE_STACK;
+      ALLOC(u,_k+2U,opus_uint32);
+      i=icwrs(_n,_k,&nc,_y,u);
+      ec_enc_uint(_enc,i,nc);
+      RESTORE_STACK;
+#ifndef SMALL_FOOTPRINT
+    }
+    break;
+  }
+#endif
+}
+
+void decode_pulses(int *_y,int _n,int _k,ec_dec *_dec)
+{
+  celt_assert(_k>0);
+#ifndef SMALL_FOOTPRINT
+   switch(_n){
+    case 2:cwrsi2(_k,ec_dec_uint(_dec,ncwrs2(_k)),_y);break;
+    case 3:cwrsi3(_k,ec_dec_uint(_dec,ncwrs3(_k)),_y);break;
+    case 4:cwrsi4(_k,ec_dec_uint(_dec,ncwrs4(_k)),_y);break;
+    default:
+    {
+#endif
+      VARDECL(opus_uint32,u);
+      SAVE_STACK;
+      ALLOC(u,_k+2U,opus_uint32);
+      cwrsi(_n,_k,ec_dec_uint(_dec,ncwrs_urow(_n,_k,u)),_y,u);
+      RESTORE_STACK;
+#ifndef SMALL_FOOTPRINT
+    }
+    break;
+  }
+#endif
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/cwrs.h b/lib/rbcodec/codecs/libopus/celt/cwrs.h
new file mode 100644
index 0000000000..7dfbd076d1
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/cwrs.h
@@ -0,0 +1,48 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2007-2009 Timothy B. Terriberry
+   Written by Timothy B. Terriberry and Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef CWRS_H
+#define CWRS_H
+
+#include "arch.h"
+#include "stack_alloc.h"
+#include "entenc.h"
+#include "entdec.h"
+
+#ifdef CUSTOM_MODES
+int log2_frac(opus_uint32 val, int frac);
+#endif
+
+void get_required_bits(opus_int16 *bits, int N, int K, int frac);
+
+void encode_pulses(const int *_y, int N, int K, ec_enc *enc);
+
+void decode_pulses(int *_y, int N, int K, ec_dec *dec);
+
+#endif /* CWRS_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/ecintrin.h b/lib/rbcodec/codecs/libopus/celt/ecintrin.h
new file mode 100644
index 0000000000..3dffa5f95c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/ecintrin.h
@@ -0,0 +1,87 @@
+/* Copyright (c) 2003-2008 Timothy B. Terriberry
+   Copyright (c) 2008 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/*Some common macros for potential platform-specific optimization.*/
+#include "opus_types.h"
+#include <math.h>
+#include <limits.h>
+#include "arch.h"
+#if !defined(_ecintrin_H)
+# define _ecintrin_H (1)
+
+/*Some specific platforms may have optimized intrinsic or inline assembly
+   versions of these functions which can substantially improve performance.
+  We define macros for them to allow easy incorporation of these non-ANSI
+   features.*/
+
+/*Modern gcc (4.x) can compile the naive versions of min and max with cmov if
+   given an appropriate architecture, but the branchless bit-twiddling versions
+   are just as fast, and do not require any special target architecture.
+  Earlier gcc versions (3.x) compiled both code to the same assembly
+   instructions, because of the way they represented ((_b)>(_a)) internally.*/
+# define EC_MINI(_a,_b)      ((_a)+(((_b)-(_a))&-((_b)<(_a))))
+
+/*Count leading zeros.
+  This macro should only be used for implementing ec_ilog(), if it is defined.
+  All other code should use EC_ILOG() instead.*/
+#if defined(_MSC_VER)
+# include <intrin.h>
+/*In _DEBUG mode this is not an intrinsic by default.*/
+# pragma intrinsic(_BitScanReverse)
+
+static __inline int ec_bsr(unsigned long _x){
+  unsigned long ret;
+  _BitScanReverse(&ret,_x);
+  return (int)ret;
+}
+# define EC_CLZ0    (1)
+# define EC_CLZ(_x) (-ec_bsr(_x))
+#elif defined(ENABLE_TI_DSPLIB)
+# include "dsplib.h"
+# define EC_CLZ0    (31)
+# define EC_CLZ(_x) (_lnorm(_x))
+#elif __GNUC_PREREQ(3,4)
+# if INT_MAX>=2147483647
+#  define EC_CLZ0    ((int)sizeof(unsigned)*CHAR_BIT)
+#  define EC_CLZ(_x) (__builtin_clz(_x))
+# elif LONG_MAX>=2147483647L
+#  define EC_CLZ0    ((int)sizeof(unsigned long)*CHAR_BIT)
+#  define EC_CLZ(_x) (__builtin_clzl(_x))
+# endif
+#endif
+
+#if defined(EC_CLZ)
+/*Note that __builtin_clz is not defined when _x==0, according to the gcc
+   documentation (and that of the BSR instruction that implements it on x86).
+  The majority of the time we can never pass it zero.
+  When we need to, it can be special cased.*/
+# define EC_ILOG(_x) (EC_CLZ0-EC_CLZ(_x))
+#else
+int ec_ilog(opus_uint32 _v);
+# define EC_ILOG(_x) (ec_ilog(_x))
+#endif
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/entcode.c b/lib/rbcodec/codecs/libopus/celt/entcode.c
new file mode 100644
index 0000000000..80e64fefaa
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entcode.c
@@ -0,0 +1,88 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "entcode.h"
+#include "arch.h"
+
+#if !defined(EC_CLZ)
+int ec_ilog(opus_uint32 _v){
+  /*On a Pentium M, this branchless version tested as the fastest on
+     1,000,000,000 random 32-bit integers, edging out a similar version with
+     branches, and a 256-entry LUT version.*/
+  int ret;
+  int m;
+  ret=!!_v;
+  m=!!(_v&0xFFFF0000)<<4;
+  _v>>=m;
+  ret|=m;
+  m=!!(_v&0xFF00)<<3;
+  _v>>=m;
+  ret|=m;
+  m=!!(_v&0xF0)<<2;
+  _v>>=m;
+  ret|=m;
+  m=!!(_v&0xC)<<1;
+  _v>>=m;
+  ret|=m;
+  ret+=!!(_v&0x2);
+  return ret;
+}
+#endif
+
+opus_uint32 ec_tell_frac(ec_ctx *_this){
+  opus_uint32 nbits;
+  opus_uint32 r;
+  int         l;
+  int         i;
+  /*To handle the non-integral number of bits still left in the encoder/decoder
+     state, we compute the worst-case number of bits of val that must be
+     encoded to ensure that the value is inside the range for any possible
+     subsequent bits.
+    The computation here is independent of val itself (the decoder does not
+     even track that value), even though the real number of bits used after
+     ec_enc_done() may be 1 smaller if rng is a power of two and the
+     corresponding trailing bits of val are all zeros.
+    If we did try to track that special case, then coding a value with a
+     probability of 1/(1<<n) might sometimes appear to use more than n bits.
+    This may help explain the surprising result that a newly initialized
+     encoder or decoder claims to have used 1 bit.*/
+  nbits=_this->nbits_total<<BITRES;
+  l=EC_ILOG(_this->rng);
+  r=_this->rng>>(l-16);
+  for(i=BITRES;i-->0;){
+    int b;
+    r=r*r>>15;
+    b=(int)(r>>16);
+    l=l<<1|b;
+    r>>=b;
+  }
+  return nbits-l;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/entcode.h b/lib/rbcodec/codecs/libopus/celt/entcode.h
new file mode 100644
index 0000000000..aebecc0647
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entcode.h
@@ -0,0 +1,116 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "opus_types.h"
+
+#if !defined(_entcode_H)
+# define _entcode_H (1)
+# include <limits.h>
+# include <stddef.h>
+# include "ecintrin.h"
+
+/*OPT: ec_window must be at least 32 bits, but if you have fast arithmetic on a
+   larger type, you can speed up the decoder by using it here.*/
+typedef opus_uint32           ec_window;
+typedef struct ec_ctx         ec_ctx;
+typedef struct ec_ctx         ec_enc;
+typedef struct ec_ctx         ec_dec;
+
+# define EC_WINDOW_SIZE ((int)sizeof(ec_window)*CHAR_BIT)
+
+/*The number of bits to use for the range-coded part of unsigned integers.*/
+# define EC_UINT_BITS   (8)
+
+/*The resolution of fractional-precision bit usage measurements, i.e.,
+   3 => 1/8th bits.*/
+# define BITRES 3
+
+/*The entropy encoder/decoder context.
+  We use the same structure for both, so that common functions like ec_tell()
+   can be used on either one.*/
+struct ec_ctx{
+   /*Buffered input/output.*/
+   unsigned char *buf;
+   /*The size of the buffer.*/
+   opus_uint32    storage;
+   /*The offset at which the last byte containing raw bits was read/written.*/
+   opus_uint32    end_offs;
+   /*Bits that will be read from/written at the end.*/
+   ec_window      end_window;
+   /*Number of valid bits in end_window.*/
+   int            nend_bits;
+   /*The total number of whole bits read/written.
+     This does not include partial bits currently in the range coder.*/
+   int            nbits_total;
+   /*The offset at which the next range coder byte will be read/written.*/
+   opus_uint32    offs;
+   /*The number of values in the current range.*/
+   opus_uint32    rng;
+   /*In the decoder: the difference between the top of the current range and
+      the input value, minus one.
+     In the encoder: the low end of the current range.*/
+   opus_uint32    val;
+   /*In the decoder: the saved normalization factor from ec_decode().
+     In the encoder: the number of oustanding carry propagating symbols.*/
+   opus_uint32    ext;
+   /*A buffered input/output symbol, awaiting carry propagation.*/
+   int            rem;
+   /*Nonzero if an error occurred.*/
+   int            error;
+};
+
+static inline opus_uint32 ec_range_bytes(ec_ctx *_this){
+  return _this->offs;
+}
+
+static inline unsigned char *ec_get_buffer(ec_ctx *_this){
+  return _this->buf;
+}
+
+static inline int ec_get_error(ec_ctx *_this){
+  return _this->error;
+}
+
+/*Returns the number of bits "used" by the encoded or decoded symbols so far.
+  This same number can be computed in either the encoder or the decoder, and is
+   suitable for making coding decisions.
+  Return: The number of bits.
+          This will always be slightly larger than the exact value (e.g., all
+           rounding error is in the positive direction).*/
+static inline int ec_tell(ec_ctx *_this){
+  return _this->nbits_total-EC_ILOG(_this->rng);
+}
+
+/*Returns the number of bits "used" by the encoded or decoded symbols so far.
+  This same number can be computed in either the encoder or the decoder, and is
+   suitable for making coding decisions.
+  Return: The number of bits scaled by 2**BITRES.
+          This will always be slightly larger than the exact value (e.g., all
+           rounding error is in the positive direction).*/
+opus_uint32 ec_tell_frac(ec_ctx *_this);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/entdec.c b/lib/rbcodec/codecs/libopus/celt/entdec.c
new file mode 100644
index 0000000000..ff8442d534
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entdec.c
@@ -0,0 +1,245 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include <stddef.h>
+#include "os_support.h"
+#include "arch.h"
+#include "entdec.h"
+#include "mfrngcod.h"
+
+/*A range decoder.
+  This is an entropy decoder based upon \cite{Mar79}, which is itself a
+   rediscovery of the FIFO arithmetic code introduced by \cite{Pas76}.
+  It is very similar to arithmetic encoding, except that encoding is done with
+   digits in any base, instead of with bits, and so it is faster when using
+   larger bases (i.e.: a byte).
+  The author claims an average waste of $\frac{1}{2}\log_b(2b)$ bits, where $b$
+   is the base, longer than the theoretical optimum, but to my knowledge there
+   is no published justification for this claim.
+  This only seems true when using near-infinite precision arithmetic so that
+   the process is carried out with no rounding errors.
+
+  An excellent description of implementation details is available at
+   http://www.arturocampos.com/ac_range.html
+  A recent work \cite{MNW98} which proposes several changes to arithmetic
+   encoding for efficiency actually re-discovers many of the principles
+   behind range encoding, and presents a good theoretical analysis of them.
+
+  End of stream is handled by writing out the smallest number of bits that
+   ensures that the stream will be correctly decoded regardless of the value of
+   any subsequent bits.
+  ec_tell() can be used to determine how many bits were needed to decode
+   all the symbols thus far; other data can be packed in the remaining bits of
+   the input buffer.
+  @PHDTHESIS{Pas76,
+    author="Richard Clark Pasco",
+    title="Source coding algorithms for fast data compression",
+    school="Dept. of Electrical Engineering, Stanford University",
+    address="Stanford, CA",
+    month=May,
+    year=1976
+  }
+  @INPROCEEDINGS{Mar79,
+   author="Martin, G.N.N.",
+   title="Range encoding: an algorithm for removing redundancy from a digitised
+    message",
+   booktitle="Video & Data Recording Conference",
+   year=1979,
+   address="Southampton",
+   month=Jul
+  }
+  @ARTICLE{MNW98,
+   author="Alistair Moffat and Radford Neal and Ian H. Witten",
+   title="Arithmetic Coding Revisited",
+   journal="{ACM} Transactions on Information Systems",
+   year=1998,
+   volume=16,
+   number=3,
+   pages="256--294",
+   month=Jul,
+   URL="http://www.stanford.edu/class/ee398/handouts/papers/Moffat98ArithmCoding.pdf"
+  }*/
+
+static int ec_read_byte(ec_dec *_this){
+  return _this->offs<_this->storage?_this->buf[_this->offs++]:0;
+}
+
+static int ec_read_byte_from_end(ec_dec *_this){
+  return _this->end_offs<_this->storage?
+   _this->buf[_this->storage-++(_this->end_offs)]:0;
+}
+
+/*Normalizes the contents of val and rng so that rng lies entirely in the
+   high-order symbol.*/
+static void ec_dec_normalize(ec_dec *_this){
+  /*If the range is too small, rescale it and input some bits.*/
+  while(_this->rng<=EC_CODE_BOT){
+    int sym;
+    _this->nbits_total+=EC_SYM_BITS;
+    _this->rng<<=EC_SYM_BITS;
+    /*Use up the remaining bits from our last symbol.*/
+    sym=_this->rem;
+    /*Read the next value from the input.*/
+    _this->rem=ec_read_byte(_this);
+    /*Take the rest of the bits we need from this new symbol.*/
+    sym=(sym<<EC_SYM_BITS|_this->rem)>>(EC_SYM_BITS-EC_CODE_EXTRA);
+    /*And subtract them from val, capped to be less than EC_CODE_TOP.*/
+    _this->val=((_this->val<<EC_SYM_BITS)+(EC_SYM_MAX&~sym))&(EC_CODE_TOP-1);
+  }
+}
+
+void ec_dec_init(ec_dec *_this,unsigned char *_buf,opus_uint32 _storage){
+  _this->buf=_buf;
+  _this->storage=_storage;
+  _this->end_offs=0;
+  _this->end_window=0;
+  _this->nend_bits=0;
+  /*This is the offset from which ec_tell() will subtract partial bits.
+    The final value after the ec_dec_normalize() call will be the same as in
+     the encoder, but we have to compensate for the bits that are added there.*/
+  _this->nbits_total=EC_CODE_BITS+1
+   -((EC_CODE_BITS-EC_CODE_EXTRA)/EC_SYM_BITS)*EC_SYM_BITS;
+  _this->offs=0;
+  _this->rng=1U<<EC_CODE_EXTRA;
+  _this->rem=ec_read_byte(_this);
+  _this->val=_this->rng-1-(_this->rem>>(EC_SYM_BITS-EC_CODE_EXTRA));
+  _this->error=0;
+  /*Normalize the interval.*/
+  ec_dec_normalize(_this);
+}
+
+unsigned ec_decode(ec_dec *_this,unsigned _ft){
+  unsigned s;
+  _this->ext=_this->rng/_ft;
+  s=(unsigned)(_this->val/_this->ext);
+  return _ft-EC_MINI(s+1,_ft);
+}
+
+unsigned ec_decode_bin(ec_dec *_this,unsigned _bits){
+   unsigned s;
+   _this->ext=_this->rng>>_bits;
+   s=(unsigned)(_this->val/_this->ext);
+   return (1U<<_bits)-EC_MINI(s+1U,1U<<_bits);
+}
+
+void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft){
+  opus_uint32 s;
+  s=IMUL32(_this->ext,_ft-_fh);
+  _this->val-=s;
+  _this->rng=_fl>0?IMUL32(_this->ext,_fh-_fl):_this->rng-s;
+  ec_dec_normalize(_this);
+}
+
+/*The probability of having a "one" is 1/(1<<_logp).*/
+int ec_dec_bit_logp(ec_dec *_this,unsigned _logp){
+  opus_uint32 r;
+  opus_uint32 d;
+  opus_uint32 s;
+  int         ret;
+  r=_this->rng;
+  d=_this->val;
+  s=r>>_logp;
+  ret=d<s;
+  if(!ret)_this->val=d-s;
+  _this->rng=ret?s:r-s;
+  ec_dec_normalize(_this);
+  return ret;
+}
+
+int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb){
+  opus_uint32 r;
+  opus_uint32 d;
+  opus_uint32 s;
+  opus_uint32 t;
+  int         ret;
+  s=_this->rng;
+  d=_this->val;
+  r=s>>_ftb;
+  ret=-1;
+  do{
+    t=s;
+    s=IMUL32(r,_icdf[++ret]);
+  }
+  while(d<s);
+  _this->val=d-s;
+  _this->rng=t-s;
+  ec_dec_normalize(_this);
+  return ret;
+}
+
+opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft){
+  unsigned ft;
+  unsigned s;
+  int      ftb;
+  /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/
+  celt_assert(_ft>1);
+  _ft--;
+  ftb=EC_ILOG(_ft);
+  if(ftb>EC_UINT_BITS){
+    opus_uint32 t;
+    ftb-=EC_UINT_BITS;
+    ft=(unsigned)(_ft>>ftb)+1;
+    s=ec_decode(_this,ft);
+    ec_dec_update(_this,s,s+1,ft);
+    t=(opus_uint32)s<<ftb|ec_dec_bits(_this,ftb);
+    if(t<=_ft)return t;
+    _this->error=1;
+    return _ft;
+  }
+  else{
+    _ft++;
+    s=ec_decode(_this,(unsigned)_ft);
+    ec_dec_update(_this,s,s+1,(unsigned)_ft);
+    return s;
+  }
+}
+
+opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _bits){
+  ec_window   window;
+  int         available;
+  opus_uint32 ret;
+  window=_this->end_window;
+  available=_this->nend_bits;
+  if((unsigned)available<_bits){
+    do{
+      window|=(ec_window)ec_read_byte_from_end(_this)<<available;
+      available+=EC_SYM_BITS;
+    }
+    while(available<=EC_WINDOW_SIZE-EC_SYM_BITS);
+  }
+  ret=(opus_uint32)window&(((opus_uint32)1<<_bits)-1U);
+  window>>=_bits;
+  available-=_bits;
+  _this->end_window=window;
+  _this->nend_bits=available;
+  _this->nbits_total+=_bits;
+  return ret;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/entdec.h b/lib/rbcodec/codecs/libopus/celt/entdec.h
new file mode 100644
index 0000000000..d8ab318730
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entdec.h
@@ -0,0 +1,100 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#if !defined(_entdec_H)
+# define _entdec_H (1)
+# include <limits.h>
+# include "entcode.h"
+
+/*Initializes the decoder.
+  _buf: The input buffer to use.
+  Return: 0 on success, or a negative value on error.*/
+void ec_dec_init(ec_dec *_this,unsigned char *_buf,opus_uint32 _storage);
+
+/*Calculates the cumulative frequency for the next symbol.
+  This can then be fed into the probability model to determine what that
+   symbol is, and the additional frequency information required to advance to
+   the next symbol.
+  This function cannot be called more than once without a corresponding call to
+   ec_dec_update(), or decoding will not proceed correctly.
+  _ft: The total frequency of the symbols in the alphabet the next symbol was
+        encoded with.
+  Return: A cumulative frequency representing the encoded symbol.
+          If the cumulative frequency of all the symbols before the one that
+           was encoded was fl, and the cumulative frequency of all the symbols
+           up to and including the one encoded is fh, then the returned value
+           will fall in the range [fl,fh).*/
+unsigned ec_decode(ec_dec *_this,unsigned _ft);
+
+/*Equivalent to ec_decode() with _ft==1<<_bits.*/
+unsigned ec_decode_bin(ec_dec *_this,unsigned _bits);
+
+/*Advance the decoder past the next symbol using the frequency information the
+   symbol was encoded with.
+  Exactly one call to ec_decode() must have been made so that all necessary
+   intermediate calculations are performed.
+  _fl:  The cumulative frequency of all symbols that come before the symbol
+         decoded.
+  _fh:  The cumulative frequency of all symbols up to and including the symbol
+         decoded.
+        Together with _fl, this defines the range [_fl,_fh) in which the value
+         returned above must fall.
+  _ft:  The total frequency of the symbols in the alphabet the symbol decoded
+         was encoded in.
+        This must be the same as passed to the preceding call to ec_decode().*/
+void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft);
+
+/* Decode a bit that has a 1/(1<<_logp) probability of being a one */
+int ec_dec_bit_logp(ec_dec *_this,unsigned _logp);
+
+/*Decodes a symbol given an "inverse" CDF table.
+  No call to ec_dec_update() is necessary after this call.
+  _icdf: The "inverse" CDF, such that symbol s falls in the range
+          [s>0?ft-_icdf[s-1]:0,ft-_icdf[s]), where ft=1<<_ftb.
+         The values must be monotonically non-increasing, and the last value
+          must be 0.
+  _ftb: The number of bits of precision in the cumulative distribution.
+  Return: The decoded symbol s.*/
+int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb);
+
+/*Extracts a raw unsigned integer with a non-power-of-2 range from the stream.
+  The bits must have been encoded with ec_enc_uint().
+  No call to ec_dec_update() is necessary after this call.
+  _ft: The number of integers that can be decoded (one more than the max).
+       This must be at least one, and no more than 2**32-1.
+  Return: The decoded bits.*/
+opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft);
+
+/*Extracts a sequence of raw bits from the stream.
+  The bits must have been encoded with ec_enc_bits().
+  No call to ec_dec_update() is necessary after this call.
+  _ftb: The number of bits to extract.
+        This must be between 0 and 25, inclusive.
+  Return: The decoded bits.*/
+opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _ftb);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/entenc.c b/lib/rbcodec/codecs/libopus/celt/entenc.c
new file mode 100644
index 0000000000..0ec6e91fd7
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entenc.c
@@ -0,0 +1,294 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#if defined(HAVE_CONFIG_H)
+# include "opus_config.h"
+#endif
+#include "os_support.h"
+#include "arch.h"
+#include "entenc.h"
+#include "mfrngcod.h"
+
+/*A range encoder.
+  See entdec.c and the references for implementation details \cite{Mar79,MNW98}.
+
+  @INPROCEEDINGS{Mar79,
+   author="Martin, G.N.N.",
+   title="Range encoding: an algorithm for removing redundancy from a digitised
+    message",
+   booktitle="Video \& Data Recording Conference",
+   year=1979,
+   address="Southampton",
+   month=Jul
+  }
+  @ARTICLE{MNW98,
+   author="Alistair Moffat and Radford Neal and Ian H. Witten",
+   title="Arithmetic Coding Revisited",
+   journal="{ACM} Transactions on Information Systems",
+   year=1998,
+   volume=16,
+   number=3,
+   pages="256--294",
+   month=Jul,
+   URL="http://www.stanford.edu/class/ee398/handouts/papers/Moffat98ArithmCoding.pdf"
+  }*/
+
+static int ec_write_byte(ec_enc *_this,unsigned _value){
+  if(_this->offs+_this->end_offs>=_this->storage)return -1;
+  _this->buf[_this->offs++]=(unsigned char)_value;
+  return 0;
+}
+
+static int ec_write_byte_at_end(ec_enc *_this,unsigned _value){
+  if(_this->offs+_this->end_offs>=_this->storage)return -1;
+  _this->buf[_this->storage-++(_this->end_offs)]=(unsigned char)_value;
+  return 0;
+}
+
+/*Outputs a symbol, with a carry bit.
+  If there is a potential to propagate a carry over several symbols, they are
+   buffered until it can be determined whether or not an actual carry will
+   occur.
+  If the counter for the buffered symbols overflows, then the stream becomes
+   undecodable.
+  This gives a theoretical limit of a few billion symbols in a single packet on
+   32-bit systems.
+  The alternative is to truncate the range in order to force a carry, but
+   requires similar carry tracking in the decoder, needlessly slowing it down.*/
+static void ec_enc_carry_out(ec_enc *_this,int _c){
+  if(_c!=EC_SYM_MAX){
+    /*No further carry propagation possible, flush buffer.*/
+    int carry;
+    carry=_c>>EC_SYM_BITS;
+    /*Don't output a byte on the first write.
+      This compare should be taken care of by branch-prediction thereafter.*/
+    if(_this->rem>=0)_this->error|=ec_write_byte(_this,_this->rem+carry);
+    if(_this->ext>0){
+      unsigned sym;
+      sym=(EC_SYM_MAX+carry)&EC_SYM_MAX;
+      do _this->error|=ec_write_byte(_this,sym);
+      while(--(_this->ext)>0);
+    }
+    _this->rem=_c&EC_SYM_MAX;
+  }
+  else _this->ext++;
+}
+
+static void ec_enc_normalize(ec_enc *_this){
+  /*If the range is too small, output some bits and rescale it.*/
+  while(_this->rng<=EC_CODE_BOT){
+    ec_enc_carry_out(_this,(int)(_this->val>>EC_CODE_SHIFT));
+    /*Move the next-to-high-order symbol into the high-order position.*/
+    _this->val=(_this->val<<EC_SYM_BITS)&(EC_CODE_TOP-1);
+    _this->rng<<=EC_SYM_BITS;
+    _this->nbits_total+=EC_SYM_BITS;
+  }
+}
+
+void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size){
+  _this->buf=_buf;
+  _this->end_offs=0;
+  _this->end_window=0;
+  _this->nend_bits=0;
+  /*This is the offset from which ec_tell() will subtract partial bits.*/
+  _this->nbits_total=EC_CODE_BITS+1;
+  _this->offs=0;
+  _this->rng=EC_CODE_TOP;
+  _this->rem=-1;
+  _this->val=0;
+  _this->ext=0;
+  _this->storage=_size;
+  _this->error=0;
+}
+
+void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft){
+  opus_uint32 r;
+  r=_this->rng/_ft;
+  if(_fl>0){
+    _this->val+=_this->rng-IMUL32(r,(_ft-_fl));
+    _this->rng=IMUL32(r,(_fh-_fl));
+  }
+  else _this->rng-=IMUL32(r,(_ft-_fh));
+  ec_enc_normalize(_this);
+}
+
+void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits){
+  opus_uint32 r;
+  r=_this->rng>>_bits;
+  if(_fl>0){
+    _this->val+=_this->rng-IMUL32(r,((1U<<_bits)-_fl));
+    _this->rng=IMUL32(r,(_fh-_fl));
+  }
+  else _this->rng-=IMUL32(r,((1U<<_bits)-_fh));
+  ec_enc_normalize(_this);
+}
+
+/*The probability of having a "one" is 1/(1<<_logp).*/
+void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp){
+  opus_uint32 r;
+  opus_uint32 s;
+  opus_uint32 l;
+  r=_this->rng;
+  l=_this->val;
+  s=r>>_logp;
+  r-=s;
+  if(_val)_this->val=l+r;
+  _this->rng=_val?s:r;
+  ec_enc_normalize(_this);
+}
+
+void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb){
+  opus_uint32 r;
+  r=_this->rng>>_ftb;
+  if(_s>0){
+    _this->val+=_this->rng-IMUL32(r,_icdf[_s-1]);
+    _this->rng=IMUL32(r,_icdf[_s-1]-_icdf[_s]);
+  }
+  else _this->rng-=IMUL32(r,_icdf[_s]);
+  ec_enc_normalize(_this);
+}
+
+void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft){
+  unsigned  ft;
+  unsigned  fl;
+  int       ftb;
+  /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/
+  celt_assert(_ft>1);
+  _ft--;
+  ftb=EC_ILOG(_ft);
+  if(ftb>EC_UINT_BITS){
+    ftb-=EC_UINT_BITS;
+    ft=(_ft>>ftb)+1;
+    fl=(unsigned)(_fl>>ftb);
+    ec_encode(_this,fl,fl+1,ft);
+    ec_enc_bits(_this,_fl&(((opus_uint32)1<<ftb)-1U),ftb);
+  }
+  else ec_encode(_this,_fl,_fl+1,_ft+1);
+}
+
+void ec_enc_bits(ec_enc *_this,opus_uint32 _fl,unsigned _bits){
+  ec_window window;
+  int       used;
+  window=_this->end_window;
+  used=_this->nend_bits;
+  celt_assert(_bits>0);
+  if(used+_bits>EC_WINDOW_SIZE){
+    do{
+      _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX);
+      window>>=EC_SYM_BITS;
+      used-=EC_SYM_BITS;
+    }
+    while(used>=EC_SYM_BITS);
+  }
+  window|=(ec_window)_fl<<used;
+  used+=_bits;
+  _this->end_window=window;
+  _this->nend_bits=used;
+  _this->nbits_total+=_bits;
+}
+
+void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits){
+  int      shift;
+  unsigned mask;
+  celt_assert(_nbits<=EC_SYM_BITS);
+  shift=EC_SYM_BITS-_nbits;
+  mask=((1<<_nbits)-1)<<shift;
+  if(_this->offs>0){
+    /*The first byte has been finalized.*/
+    _this->buf[0]=(unsigned char)((_this->buf[0]&~mask)|_val<<shift);
+  }
+  else if(_this->rem>=0){
+    /*The first byte is still awaiting carry propagation.*/
+    _this->rem=(_this->rem&~mask)|_val<<shift;
+  }
+  else if(_this->rng<=(EC_CODE_TOP>>_nbits)){
+    /*The renormalization loop has never been run.*/
+    _this->val=(_this->val&~((opus_uint32)mask<<EC_CODE_SHIFT))|
+     (opus_uint32)_val<<(EC_CODE_SHIFT+shift);
+  }
+  /*The encoder hasn't even encoded _nbits of data yet.*/
+  else _this->error=-1;
+}
+
+void ec_enc_shrink(ec_enc *_this,opus_uint32 _size){
+  celt_assert(_this->offs+_this->end_offs<=_size);
+  OPUS_MOVE(_this->buf+_size-_this->end_offs,
+   _this->buf+_this->storage-_this->end_offs,_this->end_offs);
+  _this->storage=_size;
+}
+
+void ec_enc_done(ec_enc *_this){
+  ec_window   window;
+  int         used;
+  opus_uint32 msk;
+  opus_uint32 end;
+  int         l;
+  /*We output the minimum number of bits that ensures that the symbols encoded
+     thus far will be decoded correctly regardless of the bits that follow.*/
+  l=EC_CODE_BITS-EC_ILOG(_this->rng);
+  msk=(EC_CODE_TOP-1)>>l;
+  end=(_this->val+msk)&~msk;
+  if((end|msk)>=_this->val+_this->rng){
+    l++;
+    msk>>=1;
+    end=(_this->val+msk)&~msk;
+  }
+  while(l>0){
+    ec_enc_carry_out(_this,(int)(end>>EC_CODE_SHIFT));
+    end=(end<<EC_SYM_BITS)&(EC_CODE_TOP-1);
+    l-=EC_SYM_BITS;
+  }
+  /*If we have a buffered byte flush it into the output buffer.*/
+  if(_this->rem>=0||_this->ext>0)ec_enc_carry_out(_this,0);
+  /*If we have buffered extra bits, flush them as well.*/
+  window=_this->end_window;
+  used=_this->nend_bits;
+  while(used>=EC_SYM_BITS){
+    _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX);
+    window>>=EC_SYM_BITS;
+    used-=EC_SYM_BITS;
+  }
+  /*Clear any excess space and add any remaining extra bits to the last byte.*/
+  if(!_this->error){
+    OPUS_CLEAR(_this->buf+_this->offs,
+     _this->storage-_this->offs-_this->end_offs);
+    if(used>0){
+      /*If there's no range coder data at all, give up.*/
+      if(_this->end_offs>=_this->storage)_this->error=-1;
+      else{
+        l=-l;
+        /*If we've busted, don't add too many extra bits to the last byte; it
+           would corrupt the range coder data, and that's more important.*/
+        if(_this->offs+_this->end_offs>=_this->storage&&l<used){
+          window&=(1<<l)-1;
+          _this->error=-1;
+        }
+        _this->buf[_this->storage-_this->end_offs-1]|=(unsigned char)window;
+      }
+    }
+  }
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/entenc.h b/lib/rbcodec/codecs/libopus/celt/entenc.h
new file mode 100644
index 0000000000..796bc4d572
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/entenc.h
@@ -0,0 +1,110 @@
+/* Copyright (c) 2001-2011 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#if !defined(_entenc_H)
+# define _entenc_H (1)
+# include <stddef.h>
+# include "entcode.h"
+
+/*Initializes the encoder.
+  _buf:  The buffer to store output bytes in.
+  _size: The size of the buffer, in chars.*/
+void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size);
+/*Encodes a symbol given its frequency information.
+  The frequency information must be discernable by the decoder, assuming it
+   has read only the previous symbols from the stream.
+  It is allowable to change the frequency information, or even the entire
+   source alphabet, so long as the decoder can tell from the context of the
+   previously encoded information that it is supposed to do so as well.
+  _fl: The cumulative frequency of all symbols that come before the one to be
+        encoded.
+  _fh: The cumulative frequency of all symbols up to and including the one to
+        be encoded.
+       Together with _fl, this defines the range [_fl,_fh) in which the
+        decoded value will fall.
+  _ft: The sum of the frequencies of all the symbols*/
+void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft);
+
+/*Equivalent to ec_encode() with _ft==1<<_bits.*/
+void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits);
+
+/* Encode a bit that has a 1/(1<<_logp) probability of being a one */
+void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp);
+
+/*Encodes a symbol given an "inverse" CDF table.
+  _s:    The index of the symbol to encode.
+  _icdf: The "inverse" CDF, such that symbol _s falls in the range
+          [_s>0?ft-_icdf[_s-1]:0,ft-_icdf[_s]), where ft=1<<_ftb.
+         The values must be monotonically non-increasing, and the last value
+          must be 0.
+  _ftb: The number of bits of precision in the cumulative distribution.*/
+void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb);
+
+/*Encodes a raw unsigned integer in the stream.
+  _fl: The integer to encode.
+  _ft: The number of integers that can be encoded (one more than the max).
+       This must be at least one, and no more than 2**32-1.*/
+void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft);
+
+/*Encodes a sequence of raw bits in the stream.
+  _fl:  The bits to encode.
+  _ftb: The number of bits to encode.
+        This must be between 1 and 25, inclusive.*/
+void ec_enc_bits(ec_enc *_this,opus_uint32 _fl,unsigned _ftb);
+
+/*Overwrites a few bits at the very start of an existing stream, after they
+   have already been encoded.
+  This makes it possible to have a few flags up front, where it is easy for
+   decoders to access them without parsing the whole stream, even if their
+   values are not determined until late in the encoding process, without having
+   to buffer all the intermediate symbols in the encoder.
+  In order for this to work, at least _nbits bits must have already been
+   encoded using probabilities that are an exact power of two.
+  The encoder can verify the number of encoded bits is sufficient, but cannot
+   check this latter condition.
+  _val:   The bits to encode (in the least _nbits significant bits).
+          They will be decoded in order from most-significant to least.
+  _nbits: The number of bits to overwrite.
+          This must be no more than 8.*/
+void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits);
+
+/*Compacts the data to fit in the target size.
+  This moves up the raw bits at the end of the current buffer so they are at
+   the end of the new buffer size.
+  The caller must ensure that the amount of data that's already been written
+   will fit in the new size.
+  _size: The number of bytes in the new buffer.
+         This must be large enough to contain the bits already written, and
+          must be no larger than the existing size.*/
+void ec_enc_shrink(ec_enc *_this,opus_uint32 _size);
+
+/*Indicates that there are no more symbols to encode.
+  All reamining output bytes are flushed to the output buffer.
+  ec_enc_init() must be called before the encoder can be used again.*/
+void ec_enc_done(ec_enc *_this);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/fixed_generic.h b/lib/rbcodec/codecs/libopus/celt/fixed_generic.h
new file mode 100644
index 0000000000..71e28d62a8
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/fixed_generic.h
@@ -0,0 +1,129 @@
+/* Copyright (C) 2007-2009 Xiph.Org Foundation
+   Copyright (C) 2003-2008 Jean-Marc Valin
+   Copyright (C) 2007-2008 CSIRO */
+/**
+   @file fixed_generic.h
+   @brief Generic fixed-point operations
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef FIXED_GENERIC_H
+#define FIXED_GENERIC_H
+
+/** Multiply a 16-bit signed value by a 16-bit unsigned value. The result is a 32-bit signed value */
+#define MULT16_16SU(a,b) ((opus_val32)(opus_val16)(a)*(opus_val32)(opus_uint16)(b))
+
+/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */
+#define MULT16_32_Q16(a,b) ADD32(MULT16_16((a),SHR((b),16)), SHR(MULT16_16SU((a),((b)&0x0000ffff)),16))
+
+/** 16x32 multiplication, followed by a 16-bit shift right (round-to-nearest). Results fits in 32 bits */
+#define MULT16_32_P16(a,b) ADD32(MULT16_16((a),SHR((b),16)), PSHR(MULT16_16((a),((b)&0x0000ffff)),16))
+
+/** 16x32 multiplication, followed by a 15-bit shift right. Results fits in 32 bits */
+#define MULT16_32_Q15(a,b) ADD32(SHL(MULT16_16((a),SHR((b),16)),1), SHR(MULT16_16SU((a),((b)&0x0000ffff)),15))
+
+/** 32x32 multiplication, followed by a 31-bit shift right. Results fits in 32 bits */
+#define MULT32_32_Q31(a,b) ADD32(ADD32(SHL(MULT16_16(SHR((a),16),SHR((b),16)),1), SHR(MULT16_16SU(SHR((a),16),((b)&0x0000ffff)),15)), SHR(MULT16_16SU(SHR((b),16),((a)&0x0000ffff)),15))
+
+/** Compile-time conversion of float constant to 16-bit value */
+#define QCONST16(x,bits) ((opus_val16)(.5+(x)*(((opus_val32)1)<<(bits))))
+
+/** Compile-time conversion of float constant to 32-bit value */
+#define QCONST32(x,bits) ((opus_val32)(.5+(x)*(((opus_val32)1)<<(bits))))
+
+/** Negate a 16-bit value */
+#define NEG16(x) (-(x))
+/** Negate a 32-bit value */
+#define NEG32(x) (-(x))
+
+/** Change a 32-bit value into a 16-bit value. The value is assumed to fit in 16-bit, otherwise the result is undefined */
+#define EXTRACT16(x) ((opus_val16)(x))
+/** Change a 16-bit value into a 32-bit value */
+#define EXTEND32(x) ((opus_val32)(x))
+
+/** Arithmetic shift-right of a 16-bit value */
+#define SHR16(a,shift) ((a) >> (shift))
+/** Arithmetic shift-left of a 16-bit value */
+#define SHL16(a,shift) ((opus_int16)((opus_uint16)(a)<<(shift)))
+/** Arithmetic shift-right of a 32-bit value */
+#define SHR32(a,shift) ((a) >> (shift))
+/** Arithmetic shift-left of a 32-bit value */
+#define SHL32(a,shift) ((opus_int32)((opus_uint32)(a)<<(shift)))
+
+/** 32-bit arithmetic shift right with rounding-to-nearest instead of rounding down */
+#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift))
+/** 32-bit arithmetic shift right where the argument can be negative */
+#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift)))
+
+/** "RAW" macros, should not be used outside of this header file */
+#define SHR(a,shift) ((a) >> (shift))
+#define SHL(a,shift) SHL32(a,shift)
+#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift))
+#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
+
+/** Shift by a and round-to-neareast 32-bit value. Result is a 16-bit value */
+#define ROUND16(x,a) (EXTRACT16(PSHR32((x),(a))))
+/** Divide by two */
+#define HALF16(x)  (SHR16(x,1))
+#define HALF32(x)  (SHR32(x,1))
+
+/** Add two 16-bit values */
+#define ADD16(a,b) ((opus_val16)((opus_val16)(a)+(opus_val16)(b)))
+/** Subtract two 16-bit values */
+#define SUB16(a,b) ((opus_val16)(a)-(opus_val16)(b))
+/** Add two 32-bit values */
+#define ADD32(a,b) ((opus_val32)(a)+(opus_val32)(b))
+/** Subtract two 32-bit values */
+#define SUB32(a,b) ((opus_val32)(a)-(opus_val32)(b))
+
+/** 16x16 multiplication where the result fits in 16 bits */
+#define MULT16_16_16(a,b)     ((((opus_val16)(a))*((opus_val16)(b))))
+
+/* (opus_val32)(opus_val16) gives TI compiler a hint that it's 16x16->32 multiply */
+/** 16x16 multiplication where the result fits in 32 bits */
+#define MULT16_16(a,b)     (((opus_val32)(opus_val16)(a))*((opus_val32)(opus_val16)(b)))
+
+/** 16x16 multiply-add where the result fits in 32 bits */
+#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b))))
+/** 16x32 multiply-add, followed by a 15-bit shift right. Results fits in 32 bits */
+#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)))
+
+#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11))
+#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13))
+#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14))
+#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15))
+
+#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13))
+#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14))
+#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15))
+
+/** Divide a 32-bit value by a 16-bit value. Result fits in 16 bits */
+#define DIV32_16(a,b) ((opus_val16)(((opus_val32)(a))/((opus_val16)(b))))
+
+/** Divide a 32-bit value by a 32-bit value. Result fits in 32 bits */
+#define DIV32(a,b) (((opus_val32)(a))/((opus_val32)(b)))
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/float_cast.h b/lib/rbcodec/codecs/libopus/celt/float_cast.h
new file mode 100644
index 0000000000..5ded291599
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/float_cast.h
@@ -0,0 +1,140 @@
+/* Copyright (C) 2001 Erik de Castro Lopo <erikd AT mega-nerd DOT com> */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/* Version 1.1 */
+
+#ifndef FLOAT_CAST_H
+#define FLOAT_CAST_H
+
+
+#include "arch.h"
+
+/*============================================================================
+**      On Intel Pentium processors (especially PIII and probably P4), converting
+**      from float to int is very slow. To meet the C specs, the code produced by
+**      most C compilers targeting Pentium needs to change the FPU rounding mode
+**      before the float to int conversion is performed.
+**
+**      Changing the FPU rounding mode causes the FPU pipeline to be flushed. It
+**      is this flushing of the pipeline which is so slow.
+**
+**      Fortunately the ISO C99 specifications define the functions lrint, lrintf,
+**      llrint and llrintf which fix this problem as a side effect.
+**
+**      On Unix-like systems, the configure process should have detected the
+**      presence of these functions. If they weren't found we have to replace them
+**      here with a standard C cast.
+*/
+
+/*
+**      The C99 prototypes for lrint and lrintf are as follows:
+**
+**              long int lrintf (float x) ;
+**              long int lrint  (double x) ;
+*/
+
+/*      The presence of the required functions are detected during the configure
+**      process and the values HAVE_LRINT and HAVE_LRINTF are set accordingly in
+**      the config.h file.
+*/
+
+#if (HAVE_LRINTF)
+
+/*      These defines enable functionality introduced with the 1999 ISO C
+**      standard. They must be defined before the inclusion of math.h to
+**      engage them. If optimisation is enabled, these functions will be
+**      inlined. With optimisation switched off, you have to link in the
+**      maths library using -lm.
+*/
+
+#define _ISOC9X_SOURCE 1
+#define _ISOC99_SOURCE 1
+
+#define __USE_ISOC9X 1
+#define __USE_ISOC99 1
+
+#include <math.h>
+#define float2int(x) lrintf(x)
+
+#elif (defined(HAVE_LRINT))
+
+#define _ISOC9X_SOURCE 1
+#define _ISOC99_SOURCE 1
+
+#define __USE_ISOC9X 1
+#define __USE_ISOC99 1
+
+#include <math.h>
+#define float2int(x) lrint(x)
+
+#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && (defined (WIN64) || defined (_WIN64))
+        #include <xmmintrin.h>
+
+        __inline long int float2int(float value)
+        {
+                return _mm_cvtss_si32(_mm_load_ss(&value));
+        }
+#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && (defined (WIN32) || defined (_WIN32))
+        #include <math.h>
+
+        /*      Win32 doesn't seem to have these functions.
+        **      Therefore implement inline versions of these functions here.
+        */
+
+        __inline long int
+        float2int (float flt)
+        {       int intgr;
+
+                _asm
+                {       fld flt
+                        fistp intgr
+                } ;
+
+                return intgr ;
+        }
+
+#else
+
+#if (defined(__GNUC__) && defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L)
+        /* supported by gcc in C99 mode, but not by all other compilers */
+        #warning "Don't have the functions lrint() and lrintf ()."
+        #warning "Replacing these functions with a standard C cast."
+#endif /* __STDC_VERSION__ >= 199901L */
+        #include <math.h>
+        #define float2int(flt) ((int)(floor(.5+flt)))
+#endif
+
+#ifndef DISABLE_FLOAT_API
+static inline opus_int16 FLOAT2INT16(float x)
+{
+   x = x*CELT_SIG_SCALE;
+   x = MAX32(x, -32768);
+   x = MIN32(x, 32767);
+   return (opus_int16)float2int(x);
+}
+#endif /* DISABLE_FLOAT_API */
+
+#endif /* FLOAT_CAST_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/kiss_fft.c b/lib/rbcodec/codecs/libopus/celt/kiss_fft.c
new file mode 100644
index 0000000000..3ba075ab0c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/kiss_fft.c
@@ -0,0 +1,722 @@
+/*Copyright (c) 2003-2004, Mark Borgerding
+  Lots of modifications by Jean-Marc Valin
+  Copyright (c) 2005-2007, Xiph.Org Foundation
+  Copyright (c) 2008,      Xiph.Org Foundation, CSIRO
+
+  All rights reserved.
+
+  Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are met:
+
+    * Redistributions of source code must retain the above copyright notice,
+       this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright notice,
+       this list of conditions and the following disclaimer in the
+       documentation and/or other materials provided with the distribution.
+
+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+  AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+  IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+  ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+  LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+  CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+  SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+  INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+  CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+  ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+  POSSIBILITY OF SUCH DAMAGE.*/
+
+/* This code is originally from Mark Borgerding's KISS-FFT but has been
+   heavily modified to better suit Opus */
+
+#ifndef SKIP_CONFIG_H
+#  ifdef HAVE_CONFIG_H
+#    include "opus_config.h"
+#  endif
+#endif
+
+#include "_kiss_fft_guts.h"
+#include "arch.h"
+#include "os_support.h"
+#include "mathops.h"
+#include "stack_alloc.h"
+#include "os_support.h"
+
+/* The guts header contains all the multiplication and addition macros that are defined for
+   complex numbers.  It also delares the kf_ internal functions.
+*/
+
+static void kf_bfly2(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   kiss_fft_cpx * Fout2;
+   const kiss_twiddle_cpx * tw1;
+   int i,j;
+   kiss_fft_cpx * Fout_beg = Fout;
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      Fout2 = Fout + m;
+      tw1 = st->twiddles;
+      for(j=0;j<m;j++)
+      {
+         kiss_fft_cpx t;
+         Fout->r = SHR32(Fout->r, 1);Fout->i = SHR32(Fout->i, 1);
+         Fout2->r = SHR32(Fout2->r, 1);Fout2->i = SHR32(Fout2->i, 1);
+         C_MUL (t,  *Fout2 , *tw1);
+         tw1 += fstride;
+         C_SUB( *Fout2 ,  *Fout , t );
+         C_ADDTO( *Fout ,  t );
+         ++Fout2;
+         ++Fout;
+      }
+   }
+}
+
+static void ki_bfly2(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   kiss_fft_cpx * Fout2;
+   const kiss_twiddle_cpx * tw1;
+   kiss_fft_cpx t;
+   int i,j;
+   kiss_fft_cpx * Fout_beg = Fout;
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      Fout2 = Fout + m;
+      tw1 = st->twiddles;
+      for(j=0;j<m;j++)
+      {
+         C_MULC (t,  *Fout2 , *tw1);
+         tw1 += fstride;
+         C_SUB( *Fout2 ,  *Fout , t );
+         C_ADDTO( *Fout ,  t );
+         ++Fout2;
+         ++Fout;
+      }
+   }
+}
+
+static void kf_bfly4(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   const kiss_twiddle_cpx *tw1,*tw2,*tw3;
+   kiss_fft_cpx scratch[6];
+   const size_t m2=2*m;
+   const size_t m3=3*m;
+   int i, j;
+
+   kiss_fft_cpx * Fout_beg = Fout;
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      tw3 = tw2 = tw1 = st->twiddles;
+      for (j=0;j<m;j++)
+      {
+         C_MUL4(scratch[0],Fout[m] , *tw1 );
+         C_MUL4(scratch[1],Fout[m2] , *tw2 );
+         C_MUL4(scratch[2],Fout[m3] , *tw3 );
+
+         Fout->r = PSHR32(Fout->r, 2);
+         Fout->i = PSHR32(Fout->i, 2);
+         C_SUB( scratch[5] , *Fout, scratch[1] );
+         C_ADDTO(*Fout, scratch[1]);
+         C_ADD( scratch[3] , scratch[0] , scratch[2] );
+         C_SUB( scratch[4] , scratch[0] , scratch[2] );
+         Fout[m2].r = PSHR32(Fout[m2].r, 2);
+         Fout[m2].i = PSHR32(Fout[m2].i, 2);
+         C_SUB( Fout[m2], *Fout, scratch[3] );
+         tw1 += fstride;
+         tw2 += fstride*2;
+         tw3 += fstride*3;
+         C_ADDTO( *Fout , scratch[3] );
+
+         Fout[m].r = scratch[5].r + scratch[4].i;
+         Fout[m].i = scratch[5].i - scratch[4].r;
+         Fout[m3].r = scratch[5].r - scratch[4].i;
+         Fout[m3].i = scratch[5].i + scratch[4].r;
+         ++Fout;
+      }
+   }
+}
+
+static void ki_bfly4(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   const kiss_twiddle_cpx *tw1,*tw2,*tw3;
+   kiss_fft_cpx scratch[6];
+   const size_t m2=2*m;
+   const size_t m3=3*m;
+   int i, j;
+
+   kiss_fft_cpx * Fout_beg = Fout;
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      tw3 = tw2 = tw1 = st->twiddles;
+      for (j=0;j<m;j++)
+      {
+         C_MULC(scratch[0],Fout[m] , *tw1 );
+         C_MULC(scratch[1],Fout[m2] , *tw2 );
+         C_MULC(scratch[2],Fout[m3] , *tw3 );
+
+         C_SUB( scratch[5] , *Fout, scratch[1] );
+         C_ADDTO(*Fout, scratch[1]);
+         C_ADD( scratch[3] , scratch[0] , scratch[2] );
+         C_SUB( scratch[4] , scratch[0] , scratch[2] );
+         C_SUB( Fout[m2], *Fout, scratch[3] );
+         tw1 += fstride;
+         tw2 += fstride*2;
+         tw3 += fstride*3;
+         C_ADDTO( *Fout , scratch[3] );
+
+         Fout[m].r = scratch[5].r - scratch[4].i;
+         Fout[m].i = scratch[5].i + scratch[4].r;
+         Fout[m3].r = scratch[5].r + scratch[4].i;
+         Fout[m3].i = scratch[5].i - scratch[4].r;
+         ++Fout;
+      }
+   }
+}
+
+#ifndef RADIX_TWO_ONLY
+
+static void kf_bfly3(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   int i;
+   size_t k;
+   const size_t m2 = 2*m;
+   const kiss_twiddle_cpx *tw1,*tw2;
+   kiss_fft_cpx scratch[5];
+   kiss_twiddle_cpx epi3;
+
+   kiss_fft_cpx * Fout_beg = Fout;
+   epi3 = st->twiddles[fstride*m];
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      tw1=tw2=st->twiddles;
+      k=m;
+      do {
+         C_FIXDIV(*Fout,3); C_FIXDIV(Fout[m],3); C_FIXDIV(Fout[m2],3);
+
+         C_MUL(scratch[1],Fout[m] , *tw1);
+         C_MUL(scratch[2],Fout[m2] , *tw2);
+
+         C_ADD(scratch[3],scratch[1],scratch[2]);
+         C_SUB(scratch[0],scratch[1],scratch[2]);
+         tw1 += fstride;
+         tw2 += fstride*2;
+
+         Fout[m].r = Fout->r - HALF_OF(scratch[3].r);
+         Fout[m].i = Fout->i - HALF_OF(scratch[3].i);
+
+         C_MULBYSCALAR( scratch[0] , epi3.i );
+
+         C_ADDTO(*Fout,scratch[3]);
+
+         Fout[m2].r = Fout[m].r + scratch[0].i;
+         Fout[m2].i = Fout[m].i - scratch[0].r;
+
+         Fout[m].r -= scratch[0].i;
+         Fout[m].i += scratch[0].r;
+
+         ++Fout;
+      } while(--k);
+   }
+}
+
+static void ki_bfly3(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   int i, k;
+   const size_t m2 = 2*m;
+   const kiss_twiddle_cpx *tw1,*tw2;
+   kiss_fft_cpx scratch[5];
+   kiss_twiddle_cpx epi3;
+
+   kiss_fft_cpx * Fout_beg = Fout;
+   epi3 = st->twiddles[fstride*m];
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      tw1=tw2=st->twiddles;
+      k=m;
+      do{
+
+         C_MULC(scratch[1],Fout[m] , *tw1);
+         C_MULC(scratch[2],Fout[m2] , *tw2);
+
+         C_ADD(scratch[3],scratch[1],scratch[2]);
+         C_SUB(scratch[0],scratch[1],scratch[2]);
+         tw1 += fstride;
+         tw2 += fstride*2;
+
+         Fout[m].r = Fout->r - HALF_OF(scratch[3].r);
+         Fout[m].i = Fout->i - HALF_OF(scratch[3].i);
+
+         C_MULBYSCALAR( scratch[0] , -epi3.i );
+
+         C_ADDTO(*Fout,scratch[3]);
+
+         Fout[m2].r = Fout[m].r + scratch[0].i;
+         Fout[m2].i = Fout[m].i - scratch[0].r;
+
+         Fout[m].r -= scratch[0].i;
+         Fout[m].i += scratch[0].r;
+
+         ++Fout;
+      }while(--k);
+   }
+}
+
+static void kf_bfly5(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   kiss_fft_cpx *Fout0,*Fout1,*Fout2,*Fout3,*Fout4;
+   int i, u;
+   kiss_fft_cpx scratch[13];
+   const kiss_twiddle_cpx * twiddles = st->twiddles;
+   const kiss_twiddle_cpx *tw;
+   kiss_twiddle_cpx ya,yb;
+   kiss_fft_cpx * Fout_beg = Fout;
+
+   ya = twiddles[fstride*m];
+   yb = twiddles[fstride*2*m];
+   tw=st->twiddles;
+
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      Fout0=Fout;
+      Fout1=Fout0+m;
+      Fout2=Fout0+2*m;
+      Fout3=Fout0+3*m;
+      Fout4=Fout0+4*m;
+
+      for ( u=0; u<m; ++u ) {
+         C_FIXDIV( *Fout0,5); C_FIXDIV( *Fout1,5); C_FIXDIV( *Fout2,5); C_FIXDIV( *Fout3,5); C_FIXDIV( *Fout4,5);
+         scratch[0] = *Fout0;
+
+         C_MUL(scratch[1] ,*Fout1, tw[u*fstride]);
+         C_MUL(scratch[2] ,*Fout2, tw[2*u*fstride]);
+         C_MUL(scratch[3] ,*Fout3, tw[3*u*fstride]);
+         C_MUL(scratch[4] ,*Fout4, tw[4*u*fstride]);
+
+         C_ADD( scratch[7],scratch[1],scratch[4]);
+         C_SUB( scratch[10],scratch[1],scratch[4]);
+         C_ADD( scratch[8],scratch[2],scratch[3]);
+         C_SUB( scratch[9],scratch[2],scratch[3]);
+
+         Fout0->r += scratch[7].r + scratch[8].r;
+         Fout0->i += scratch[7].i + scratch[8].i;
+
+         scratch[5].r = scratch[0].r + S_MUL(scratch[7].r,ya.r) + S_MUL(scratch[8].r,yb.r);
+         scratch[5].i = scratch[0].i + S_MUL(scratch[7].i,ya.r) + S_MUL(scratch[8].i,yb.r);
+
+         scratch[6].r =  S_MUL(scratch[10].i,ya.i) + S_MUL(scratch[9].i,yb.i);
+         scratch[6].i = -S_MUL(scratch[10].r,ya.i) - S_MUL(scratch[9].r,yb.i);
+
+         C_SUB(*Fout1,scratch[5],scratch[6]);
+         C_ADD(*Fout4,scratch[5],scratch[6]);
+
+         scratch[11].r = scratch[0].r + S_MUL(scratch[7].r,yb.r) + S_MUL(scratch[8].r,ya.r);
+         scratch[11].i = scratch[0].i + S_MUL(scratch[7].i,yb.r) + S_MUL(scratch[8].i,ya.r);
+         scratch[12].r = - S_MUL(scratch[10].i,yb.i) + S_MUL(scratch[9].i,ya.i);
+         scratch[12].i = S_MUL(scratch[10].r,yb.i) - S_MUL(scratch[9].r,ya.i);
+
+         C_ADD(*Fout2,scratch[11],scratch[12]);
+         C_SUB(*Fout3,scratch[11],scratch[12]);
+
+         ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4;
+      }
+   }
+}
+
+static void ki_bfly5(
+                     kiss_fft_cpx * Fout,
+                     const size_t fstride,
+                     const kiss_fft_state *st,
+                     int m,
+                     int N,
+                     int mm
+                    )
+{
+   kiss_fft_cpx *Fout0,*Fout1,*Fout2,*Fout3,*Fout4;
+   int i, u;
+   kiss_fft_cpx scratch[13];
+   const kiss_twiddle_cpx * twiddles = st->twiddles;
+   const kiss_twiddle_cpx *tw;
+   kiss_twiddle_cpx ya,yb;
+   kiss_fft_cpx * Fout_beg = Fout;
+
+   ya = twiddles[fstride*m];
+   yb = twiddles[fstride*2*m];
+   tw=st->twiddles;
+
+   for (i=0;i<N;i++)
+   {
+      Fout = Fout_beg + i*mm;
+      Fout0=Fout;
+      Fout1=Fout0+m;
+      Fout2=Fout0+2*m;
+      Fout3=Fout0+3*m;
+      Fout4=Fout0+4*m;
+
+      for ( u=0; u<m; ++u ) {
+         scratch[0] = *Fout0;
+
+         C_MULC(scratch[1] ,*Fout1, tw[u*fstride]);
+         C_MULC(scratch[2] ,*Fout2, tw[2*u*fstride]);
+         C_MULC(scratch[3] ,*Fout3, tw[3*u*fstride]);
+         C_MULC(scratch[4] ,*Fout4, tw[4*u*fstride]);
+
+         C_ADD( scratch[7],scratch[1],scratch[4]);
+         C_SUB( scratch[10],scratch[1],scratch[4]);
+         C_ADD( scratch[8],scratch[2],scratch[3]);
+         C_SUB( scratch[9],scratch[2],scratch[3]);
+
+         Fout0->r += scratch[7].r + scratch[8].r;
+         Fout0->i += scratch[7].i + scratch[8].i;
+
+         scratch[5].r = scratch[0].r + S_MUL(scratch[7].r,ya.r) + S_MUL(scratch[8].r,yb.r);
+         scratch[5].i = scratch[0].i + S_MUL(scratch[7].i,ya.r) + S_MUL(scratch[8].i,yb.r);
+
+         scratch[6].r = -S_MUL(scratch[10].i,ya.i) - S_MUL(scratch[9].i,yb.i);
+         scratch[6].i =  S_MUL(scratch[10].r,ya.i) + S_MUL(scratch[9].r,yb.i);
+
+         C_SUB(*Fout1,scratch[5],scratch[6]);
+         C_ADD(*Fout4,scratch[5],scratch[6]);
+
+         scratch[11].r = scratch[0].r + S_MUL(scratch[7].r,yb.r) + S_MUL(scratch[8].r,ya.r);
+         scratch[11].i = scratch[0].i + S_MUL(scratch[7].i,yb.r) + S_MUL(scratch[8].i,ya.r);
+         scratch[12].r =  S_MUL(scratch[10].i,yb.i) - S_MUL(scratch[9].i,ya.i);
+         scratch[12].i = -S_MUL(scratch[10].r,yb.i) + S_MUL(scratch[9].r,ya.i);
+
+         C_ADD(*Fout2,scratch[11],scratch[12]);
+         C_SUB(*Fout3,scratch[11],scratch[12]);
+
+         ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4;
+      }
+   }
+}
+
+#endif
+
+
+#ifdef CUSTOM_MODES
+
+static
+void compute_bitrev_table(
+         int Fout,
+         opus_int16 *f,
+         const size_t fstride,
+         int in_stride,
+         opus_int16 * factors,
+         const kiss_fft_state *st
+            )
+{
+   const int p=*factors++; /* the radix  */
+   const int m=*factors++; /* stage's fft length/p */
+
+    /*printf ("fft %d %d %d %d %d %d\n", p*m, m, p, s2, fstride*in_stride, N);*/
+   if (m==1)
+   {
+      int j;
+      for (j=0;j<p;j++)
+      {
+         *f = Fout+j;
+         f += fstride*in_stride;
+      }
+   } else {
+      int j;
+      for (j=0;j<p;j++)
+      {
+         compute_bitrev_table( Fout , f, fstride*p, in_stride, factors,st);
+         f += fstride*in_stride;
+         Fout += m;
+      }
+   }
+}
+
+/*  facbuf is populated by p1,m1,p2,m2, ...
+    where
+    p[i] * m[i] = m[i-1]
+    m0 = n                  */
+static
+int kf_factor(int n,opus_int16 * facbuf)
+{
+    int p=4;
+
+    /*factor out powers of 4, powers of 2, then any remaining primes */
+    do {
+        while (n % p) {
+            switch (p) {
+                case 4: p = 2; break;
+                case 2: p = 3; break;
+                default: p += 2; break;
+            }
+            if (p>32000 || (opus_int32)p*(opus_int32)p > n)
+                p = n;          /* no more factors, skip to end */
+        }
+        n /= p;
+#ifdef RADIX_TWO_ONLY
+        if (p!=2 && p != 4)
+#else
+        if (p>5)
+#endif
+        {
+           return 0;
+        }
+        *facbuf++ = p;
+        *facbuf++ = n;
+    } while (n > 1);
+    return 1;
+}
+
+static void compute_twiddles(kiss_twiddle_cpx *twiddles, int nfft)
+{
+   int i;
+#ifdef FIXED_POINT
+   for (i=0;i<nfft;++i) {
+      opus_val32 phase = -i;
+      kf_cexp2(twiddles+i, DIV32(SHL32(phase,17),nfft));
+   }
+#else
+   for (i=0;i<nfft;++i) {
+      const double pi=3.14159265358979323846264338327;
+      double phase = ( -2*pi /nfft ) * i;
+      kf_cexp(twiddles+i, phase );
+   }
+#endif
+}
+
+/*
+ *
+ * Allocates all necessary storage space for the fft and ifft.
+ * The return value is a contiguous block of memory.  As such,
+ * It can be freed with free().
+ * */
+kiss_fft_state *opus_fft_alloc_twiddles(int nfft,void * mem,size_t * lenmem,  const kiss_fft_state *base)
+{
+    kiss_fft_state *st=NULL;
+    size_t memneeded = sizeof(struct kiss_fft_state); /* twiddle factors*/
+
+    if ( lenmem==NULL ) {
+        st = ( kiss_fft_state*)KISS_FFT_MALLOC( memneeded );
+    }else{
+        if (mem != NULL && *lenmem >= memneeded)
+            st = (kiss_fft_state*)mem;
+        *lenmem = memneeded;
+    }
+    if (st) {
+        opus_int16 *bitrev;
+        kiss_twiddle_cpx *twiddles;
+
+        st->nfft=nfft;
+#ifndef FIXED_POINT
+        st->scale = 1.f/nfft;
+#endif
+        if (base != NULL)
+        {
+           st->twiddles = base->twiddles;
+           st->shift = 0;
+           while (nfft<<st->shift != base->nfft && st->shift < 32)
+              st->shift++;
+           if (st->shift>=32)
+              goto fail;
+        } else {
+           st->twiddles = twiddles = (kiss_twiddle_cpx*)KISS_FFT_MALLOC(sizeof(kiss_twiddle_cpx)*nfft);
+           compute_twiddles(twiddles, nfft);
+           st->shift = -1;
+        }
+        if (!kf_factor(nfft,st->factors))
+        {
+           goto fail;
+        }
+
+        /* bitrev */
+        st->bitrev = bitrev = (opus_int16*)KISS_FFT_MALLOC(sizeof(opus_int16)*nfft);
+        if (st->bitrev==NULL)
+            goto fail;
+        compute_bitrev_table(0, bitrev, 1,1, st->factors,st);
+    }
+    return st;
+fail:
+    opus_fft_free(st);
+    return NULL;
+}
+
+kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem )
+{
+   return opus_fft_alloc_twiddles(nfft, mem, lenmem, NULL);
+}
+
+void opus_fft_free(const kiss_fft_state *cfg)
+{
+   if (cfg)
+   {
+      opus_free((opus_int16*)cfg->bitrev);
+      if (cfg->shift < 0)
+         opus_free((kiss_twiddle_cpx*)cfg->twiddles);
+      opus_free((kiss_fft_state*)cfg);
+   }
+}
+
+#endif /* CUSTOM_MODES */
+
+void opus_fft(const kiss_fft_state *st,const kiss_fft_cpx *fin,kiss_fft_cpx *fout)
+{
+    int m2, m;
+    int p;
+    int L;
+    int fstride[MAXFACTORS];
+    int i;
+    int shift;
+
+    /* st->shift can be -1 */
+    shift = st->shift>0 ? st->shift : 0;
+
+    celt_assert2 (fin != fout, "In-place FFT not supported");
+    /* Bit-reverse the input */
+    for (i=0;i<st->nfft;i++)
+    {
+       fout[st->bitrev[i]] = fin[i];
+#ifndef FIXED_POINT
+       fout[st->bitrev[i]].r *= st->scale;
+       fout[st->bitrev[i]].i *= st->scale;
+#endif
+    }
+
+    fstride[0] = 1;
+    L=0;
+    do {
+       p = st->factors[2*L];
+       m = st->factors[2*L+1];
+       fstride[L+1] = fstride[L]*p;
+       L++;
+    } while(m!=1);
+    m = st->factors[2*L-1];
+    for (i=L-1;i>=0;i--)
+    {
+       if (i!=0)
+          m2 = st->factors[2*i-1];
+       else
+          m2 = 1;
+       switch (st->factors[2*i])
+       {
+       case 2:
+          kf_bfly2(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+          break;
+       case 4:
+          kf_bfly4(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+          break;
+ #ifndef RADIX_TWO_ONLY
+       case 3:
+          kf_bfly3(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+          break;
+       case 5:
+          kf_bfly5(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+          break;
+ #endif
+       }
+       m = m2;
+    }
+}
+
+void opus_ifft(const kiss_fft_state *st,const kiss_fft_cpx *fin,kiss_fft_cpx *fout)
+{
+   int m2, m;
+   int p;
+   int L;
+   int fstride[MAXFACTORS];
+   int i;
+   int shift;
+
+   /* st->shift can be -1 */
+   shift = st->shift>0 ? st->shift : 0;
+   celt_assert2 (fin != fout, "In-place FFT not supported");
+   /* Bit-reverse the input */
+   for (i=0;i<st->nfft;i++)
+      fout[st->bitrev[i]] = fin[i];
+
+   fstride[0] = 1;
+   L=0;
+   do {
+      p = st->factors[2*L];
+      m = st->factors[2*L+1];
+      fstride[L+1] = fstride[L]*p;
+      L++;
+   } while(m!=1);
+   m = st->factors[2*L-1];
+   for (i=L-1;i>=0;i--)
+   {
+      if (i!=0)
+         m2 = st->factors[2*i-1];
+      else
+         m2 = 1;
+      switch (st->factors[2*i])
+      {
+      case 2:
+         ki_bfly2(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+         break;
+      case 4:
+         ki_bfly4(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+         break;
+#ifndef RADIX_TWO_ONLY
+      case 3:
+         ki_bfly3(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+         break;
+      case 5:
+         ki_bfly5(fout,fstride[i]<<shift,st,m, fstride[i], m2);
+         break;
+#endif
+      }
+      m = m2;
+   }
+}
+
diff --git a/lib/rbcodec/codecs/libopus/celt/kiss_fft.h b/lib/rbcodec/codecs/libopus/celt/kiss_fft.h
new file mode 100644
index 0000000000..66332e3bb9
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/kiss_fft.h
@@ -0,0 +1,139 @@
+/*Copyright (c) 2003-2004, Mark Borgerding
+  Lots of modifications by Jean-Marc Valin
+  Copyright (c) 2005-2007, Xiph.Org Foundation
+  Copyright (c) 2008,      Xiph.Org Foundation, CSIRO
+
+  All rights reserved.
+
+  Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are met:
+
+    * Redistributions of source code must retain the above copyright notice,
+       this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright notice,
+       this list of conditions and the following disclaimer in the
+       documentation and/or other materials provided with the distribution.
+
+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+  AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+  IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+  ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+  LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+  CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+  SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+  INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+  CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+  ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+  POSSIBILITY OF SUCH DAMAGE.*/
+
+#ifndef KISS_FFT_H
+#define KISS_FFT_H
+
+#include <stdlib.h>
+#include <math.h>
+#include "arch.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef USE_SIMD
+# include <xmmintrin.h>
+# define kiss_fft_scalar __m128
+#define KISS_FFT_MALLOC(nbytes) memalign(16,nbytes)
+#else
+#define KISS_FFT_MALLOC opus_alloc
+#endif
+
+#ifdef FIXED_POINT
+#include "arch.h"
+
+#  define kiss_fft_scalar opus_int32
+#  define kiss_twiddle_scalar opus_int16
+
+
+#else
+# ifndef kiss_fft_scalar
+/*  default is float */
+#   define kiss_fft_scalar float
+#   define kiss_twiddle_scalar float
+#   define KF_SUFFIX _celt_single
+# endif
+#endif
+
+typedef struct {
+    kiss_fft_scalar r;
+    kiss_fft_scalar i;
+}kiss_fft_cpx;
+
+typedef struct {
+   kiss_twiddle_scalar r;
+   kiss_twiddle_scalar i;
+}kiss_twiddle_cpx;
+
+#define MAXFACTORS 8
+/* e.g. an fft of length 128 has 4 factors
+ as far as kissfft is concerned
+ 4*4*4*2
+ */
+
+typedef struct kiss_fft_state{
+    int nfft;
+#ifndef FIXED_POINT
+    kiss_fft_scalar scale;
+#endif
+    int shift;
+    opus_int16 factors[2*MAXFACTORS];
+    const opus_int16 *bitrev;
+    const kiss_twiddle_cpx *twiddles;
+} kiss_fft_state;
+
+/*typedef struct kiss_fft_state* kiss_fft_cfg;*/
+
+/**
+ *  opus_fft_alloc
+ *
+ *  Initialize a FFT (or IFFT) algorithm's cfg/state buffer.
+ *
+ *  typical usage:      kiss_fft_cfg mycfg=opus_fft_alloc(1024,0,NULL,NULL);
+ *
+ *  The return value from fft_alloc is a cfg buffer used internally
+ *  by the fft routine or NULL.
+ *
+ *  If lenmem is NULL, then opus_fft_alloc will allocate a cfg buffer using malloc.
+ *  The returned value should be free()d when done to avoid memory leaks.
+ *
+ *  The state can be placed in a user supplied buffer 'mem':
+ *  If lenmem is not NULL and mem is not NULL and *lenmem is large enough,
+ *      then the function places the cfg in mem and the size used in *lenmem
+ *      and returns mem.
+ *
+ *  If lenmem is not NULL and ( mem is NULL or *lenmem is not large enough),
+ *      then the function returns NULL and places the minimum cfg
+ *      buffer size in *lenmem.
+ * */
+
+kiss_fft_state *opus_fft_alloc_twiddles(int nfft,void * mem,size_t * lenmem, const kiss_fft_state *base);
+
+kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem);
+
+/**
+ * opus_fft(cfg,in_out_buf)
+ *
+ * Perform an FFT on a complex input buffer.
+ * for a forward FFT,
+ * fin should be  f[0] , f[1] , ... ,f[nfft-1]
+ * fout will be   F[0] , F[1] , ... ,F[nfft-1]
+ * Note that each element is complex and can be accessed like
+    f[k].r and f[k].i
+ * */
+void opus_fft(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout);
+void opus_ifft(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout);
+
+void opus_fft_free(const kiss_fft_state *cfg);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/laplace.c b/lib/rbcodec/codecs/libopus/celt/laplace.c
new file mode 100644
index 0000000000..6fa4009d57
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/laplace.c
@@ -0,0 +1,134 @@
+/* Copyright (c) 2007 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "laplace.h"
+#include "mathops.h"
+
+/* The minimum probability of an energy delta (out of 32768). */
+#define LAPLACE_LOG_MINP (0)
+#define LAPLACE_MINP (1<<LAPLACE_LOG_MINP)
+/* The minimum number of guaranteed representable energy deltas (in one
+    direction). */
+#define LAPLACE_NMIN (16)
+
+/* When called, decay is positive and at most 11456. */
+static unsigned ec_laplace_get_freq1(unsigned fs0, int decay)
+{
+   unsigned ft;
+   ft = 32768 - LAPLACE_MINP*(2*LAPLACE_NMIN) - fs0;
+   return ft*(opus_int32)(16384-decay)>>15;
+}
+
+void ec_laplace_encode(ec_enc *enc, int *value, unsigned fs, int decay)
+{
+   unsigned fl;
+   int val = *value;
+   fl = 0;
+   if (val)
+   {
+      int s;
+      int i;
+      s = -(val<0);
+      val = (val+s)^s;
+      fl = fs;
+      fs = ec_laplace_get_freq1(fs, decay);
+      /* Search the decaying part of the PDF.*/
+      for (i=1; fs > 0 && i < val; i++)
+      {
+         fs *= 2;
+         fl += fs+2*LAPLACE_MINP;
+         fs = (fs*(opus_int32)decay)>>15;
+      }
+      /* Everything beyond that has probability LAPLACE_MINP. */
+      if (!fs)
+      {
+         int di;
+         int ndi_max;
+         ndi_max = (32768-fl+LAPLACE_MINP-1)>>LAPLACE_LOG_MINP;
+         ndi_max = (ndi_max-s)>>1;
+         di = IMIN(val - i, ndi_max - 1);
+         fl += (2*di+1+s)*LAPLACE_MINP;
+         fs = IMIN(LAPLACE_MINP, 32768-fl);
+         *value = (i+di+s)^s;
+      }
+      else
+      {
+         fs += LAPLACE_MINP;
+         fl += fs&~s;
+      }
+      celt_assert(fl+fs<=32768);
+      celt_assert(fs>0);
+   }
+   ec_encode_bin(enc, fl, fl+fs, 15);
+}
+
+int ec_laplace_decode(ec_dec *dec, unsigned fs, int decay)
+{
+   int val=0;
+   unsigned fl;
+   unsigned fm;
+   fm = ec_decode_bin(dec, 15);
+   fl = 0;
+   if (fm >= fs)
+   {
+      val++;
+      fl = fs;
+      fs = ec_laplace_get_freq1(fs, decay)+LAPLACE_MINP;
+      /* Search the decaying part of the PDF.*/
+      while(fs > LAPLACE_MINP && fm >= fl+2*fs)
+      {
+         fs *= 2;
+         fl += fs;
+         fs = ((fs-2*LAPLACE_MINP)*(opus_int32)decay)>>15;
+         fs += LAPLACE_MINP;
+         val++;
+      }
+      /* Everything beyond that has probability LAPLACE_MINP. */
+      if (fs <= LAPLACE_MINP)
+      {
+         int di;
+         di = (fm-fl)>>(LAPLACE_LOG_MINP+1);
+         val += di;
+         fl += 2*di*LAPLACE_MINP;
+      }
+      if (fm < fl+fs)
+         val = -val;
+      else
+         fl += fs;
+   }
+   celt_assert(fl<32768);
+   celt_assert(fs>0);
+   celt_assert(fl<=fm);
+   celt_assert(fm<IMIN(fl+fs,32768));
+   ec_dec_update(dec, fl, IMIN(fl+fs,32768), 32768);
+   return val;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/laplace.h b/lib/rbcodec/codecs/libopus/celt/laplace.h
new file mode 100644
index 0000000000..46c14b5da5
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/laplace.h
@@ -0,0 +1,48 @@
+/* Copyright (c) 2007 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "entenc.h"
+#include "entdec.h"
+
+/** Encode a value that is assumed to be the realisation of a
+    Laplace-distributed random process
+ @param enc Entropy encoder state
+ @param value Value to encode
+ @param fs Probability of 0, multiplied by 32768
+ @param decay Probability of the value +/- 1, multiplied by 16384
+*/
+void ec_laplace_encode(ec_enc *enc, int *value, unsigned fs, int decay);
+
+/** Decode a value that is assumed to be the realisation of a
+    Laplace-distributed random process
+ @param dec Entropy decoder state
+ @param fs Probability of 0, multiplied by 32768
+ @param decay Probability of the value +/- 1, multiplied by 16384
+ @return Value decoded
+ */
+int ec_laplace_decode(ec_dec *dec, unsigned fs, int decay);
diff --git a/lib/rbcodec/codecs/libopus/celt/mathops.c b/lib/rbcodec/codecs/libopus/celt/mathops.c
new file mode 100644
index 0000000000..1af6672592
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/mathops.c
@@ -0,0 +1,206 @@
+/* Copyright (c) 2002-2008 Jean-Marc Valin
+   Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file mathops.h
+   @brief Various math functions
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "mathops.h"
+
+/*Compute floor(sqrt(_val)) with exact arithmetic.
+  This has been tested on all possible 32-bit inputs.*/
+unsigned isqrt32(opus_uint32 _val){
+  unsigned b;
+  unsigned g;
+  int      bshift;
+  /*Uses the second method from
+     http://www.azillionmonkeys.com/qed/sqroot.html
+    The main idea is to search for the largest binary digit b such that
+     (g+b)*(g+b) <= _val, and add it to the solution g.*/
+  g=0;
+  bshift=(EC_ILOG(_val)-1)>>1;
+  b=1U<<bshift;
+  do{
+    opus_uint32 t;
+    t=(((opus_uint32)g<<1)+b)<<bshift;
+    if(t<=_val){
+      g+=b;
+      _val-=t;
+    }
+    b>>=1;
+    bshift--;
+  }
+  while(bshift>=0);
+  return g;
+}
+
+#ifdef FIXED_POINT
+
+opus_val32 frac_div32(opus_val32 a, opus_val32 b)
+{
+   opus_val16 rcp;
+   opus_val32 result, rem;
+   int shift = celt_ilog2(b)-29;
+   a = VSHR32(a,shift);
+   b = VSHR32(b,shift);
+   /* 16-bit reciprocal */
+   rcp = ROUND16(celt_rcp(ROUND16(b,16)),3);
+   result = MULT16_32_Q15(rcp, a);
+   rem = PSHR32(a,2)-MULT32_32_Q31(result, b);
+   result = ADD32(result, SHL32(MULT16_32_Q15(rcp, rem),2));
+   if (result >= 536870912)       /*  2^29 */
+      return 2147483647;          /*  2^31 - 1 */
+   else if (result <= -536870912) /* -2^29 */
+      return -2147483647;         /* -2^31 */
+   else
+      return SHL32(result, 2);
+}
+
+/** Reciprocal sqrt approximation in the range [0.25,1) (Q16 in, Q14 out) */
+opus_val16 celt_rsqrt_norm(opus_val32 x)
+{
+   opus_val16 n;
+   opus_val16 r;
+   opus_val16 r2;
+   opus_val16 y;
+   /* Range of n is [-16384,32767] ([-0.5,1) in Q15). */
+   n = x-32768;
+   /* Get a rough initial guess for the root.
+      The optimal minimax quadratic approximation (using relative error) is
+       r = 1.437799046117536+n*(-0.823394375837328+n*0.4096419668459485).
+      Coefficients here, and the final result r, are Q14.*/
+   r = ADD16(23557, MULT16_16_Q15(n, ADD16(-13490, MULT16_16_Q15(n, 6713))));
+   /* We want y = x*r*r-1 in Q15, but x is 32-bit Q16 and r is Q14.
+      We can compute the result from n and r using Q15 multiplies with some
+       adjustment, carefully done to avoid overflow.
+      Range of y is [-1564,1594]. */
+   r2 = MULT16_16_Q15(r, r);
+   y = SHL16(SUB16(ADD16(MULT16_16_Q15(r2, n), r2), 16384), 1);
+   /* Apply a 2nd-order Householder iteration: r += r*y*(y*0.375-0.5).
+      This yields the Q14 reciprocal square root of the Q16 x, with a maximum
+       relative error of 1.04956E-4, a (relative) RMSE of 2.80979E-5, and a
+       peak absolute error of 2.26591/16384. */
+   return ADD16(r, MULT16_16_Q15(r, MULT16_16_Q15(y,
+              SUB16(MULT16_16_Q15(y, 12288), 16384))));
+}
+
+/** Sqrt approximation (QX input, QX/2 output) */
+opus_val32 celt_sqrt(opus_val32 x)
+{
+   int k;
+   opus_val16 n;
+   opus_val32 rt;
+   static const opus_val16 C[5] = {23175, 11561, -3011, 1699, -664};
+   if (x==0)
+      return 0;
+   k = (celt_ilog2(x)>>1)-7;
+   x = VSHR32(x, 2*k);
+   n = x-32768;
+   rt = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2],
+              MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, (C[4])))))))));
+   rt = VSHR32(rt,7-k);
+   return rt;
+}
+
+#define L1 32767
+#define L2 -7651
+#define L3 8277
+#define L4 -626
+
+static inline opus_val16 _celt_cos_pi_2(opus_val16 x)
+{
+   opus_val16 x2;
+
+   x2 = MULT16_16_P15(x,x);
+   return ADD16(1,MIN16(32766,ADD32(SUB16(L1,x2), MULT16_16_P15(x2, ADD32(L2, MULT16_16_P15(x2, ADD32(L3, MULT16_16_P15(L4, x2
+                                                                                ))))))));
+}
+
+#undef L1
+#undef L2
+#undef L3
+#undef L4
+
+opus_val16 celt_cos_norm(opus_val32 x)
+{
+   x = x&0x0001ffff;
+   if (x>SHL32(EXTEND32(1), 16))
+      x = SUB32(SHL32(EXTEND32(1), 17),x);
+   if (x&0x00007fff)
+   {
+      if (x<SHL32(EXTEND32(1), 15))
+      {
+         return _celt_cos_pi_2(EXTRACT16(x));
+      } else {
+         return NEG32(_celt_cos_pi_2(EXTRACT16(65536-x)));
+      }
+   } else {
+      if (x&0x0000ffff)
+         return 0;
+      else if (x&0x0001ffff)
+         return -32767;
+      else
+         return 32767;
+   }
+}
+
+/** Reciprocal approximation (Q15 input, Q16 output) */
+opus_val32 celt_rcp(opus_val32 x)
+{
+   int i;
+   opus_val16 n;
+   opus_val16 r;
+   celt_assert2(x>0, "celt_rcp() only defined for positive values");
+   i = celt_ilog2(x);
+   /* n is Q15 with range [0,1). */
+   n = VSHR32(x,i-15)-32768;
+   /* Start with a linear approximation:
+      r = 1.8823529411764706-0.9411764705882353*n.
+      The coefficients and the result are Q14 in the range [15420,30840].*/
+   r = ADD16(30840, MULT16_16_Q15(-15420, n));
+   /* Perform two Newton iterations:
+      r -= r*((r*n)-1.Q15)
+         = r*((r*n)+(r-1.Q15)). */
+   r = SUB16(r, MULT16_16_Q15(r,
+             ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768))));
+   /* We subtract an extra 1 in the second iteration to avoid overflow; it also
+       neatly compensates for truncation error in the rest of the process. */
+   r = SUB16(r, ADD16(1, MULT16_16_Q15(r,
+             ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768)))));
+   /* r is now the Q15 solution to 2/(n+1), with a maximum relative error
+       of 7.05346E-5, a (relative) RMSE of 2.14418E-5, and a peak absolute
+       error of 1.24665/32768. */
+   return VSHR32(EXTEND32(r),i-16);
+}
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/mathops.h b/lib/rbcodec/codecs/libopus/celt/mathops.h
new file mode 100644
index 0000000000..4e97795606
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/mathops.h
@@ -0,0 +1,237 @@
+/* Copyright (c) 2002-2008 Jean-Marc Valin
+   Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file mathops.h
+   @brief Various math functions
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef MATHOPS_H
+#define MATHOPS_H
+
+#include "arch.h"
+#include "entcode.h"
+#include "os_support.h"
+
+/* Multiplies two 16-bit fractional values. Bit-exactness of this macro is important */
+#define FRAC_MUL16(a,b) ((16384+((opus_int32)(opus_int16)(a)*(opus_int16)(b)))>>15)
+
+unsigned isqrt32(opus_uint32 _val);
+
+#ifndef FIXED_POINT
+
+#define PI 3.141592653f
+#define celt_sqrt(x) ((float)sqrt(x))
+#define celt_rsqrt(x) (1.f/celt_sqrt(x))
+#define celt_rsqrt_norm(x) (celt_rsqrt(x))
+#define celt_cos_norm(x) ((float)cos((.5f*PI)*(x)))
+#define celt_rcp(x) (1.f/(x))
+#define celt_div(a,b) ((a)/(b))
+#define frac_div32(a,b) ((float)(a)/(b))
+
+#ifdef FLOAT_APPROX
+
+/* Note: This assumes radix-2 floating point with the exponent at bits 23..30 and an offset of 127
+         denorm, +/- inf and NaN are *not* handled */
+
+/** Base-2 log approximation (log2(x)). */
+static inline float celt_log2(float x)
+{
+   int integer;
+   float frac;
+   union {
+      float f;
+      opus_uint32 i;
+   } in;
+   in.f = x;
+   integer = (in.i>>23)-127;
+   in.i -= integer<<23;
+   frac = in.f - 1.5f;
+   frac = -0.41445418f + frac*(0.95909232f
+          + frac*(-0.33951290f + frac*0.16541097f));
+   return 1+integer+frac;
+}
+
+/** Base-2 exponential approximation (2^x). */
+static inline float celt_exp2(float x)
+{
+   int integer;
+   float frac;
+   union {
+      float f;
+      opus_uint32 i;
+   } res;
+   integer = floor(x);
+   if (integer < -50)
+      return 0;
+   frac = x-integer;
+   /* K0 = 1, K1 = log(2), K2 = 3-4*log(2), K3 = 3*log(2) - 2 */
+   res.f = 0.99992522f + frac * (0.69583354f
+           + frac * (0.22606716f + 0.078024523f*frac));
+   res.i = (res.i + (integer<<23)) & 0x7fffffff;
+   return res.f;
+}
+
+#else
+#define celt_log2(x) ((float)(1.442695040888963387*log(x)))
+#define celt_exp2(x) ((float)exp(0.6931471805599453094*(x)))
+#endif
+
+#endif
+
+#ifdef FIXED_POINT
+
+#include "os_support.h"
+
+#ifndef OVERRIDE_CELT_ILOG2
+/** Integer log in base2. Undefined for zero and negative numbers */
+static inline opus_int16 celt_ilog2(opus_int32 x)
+{
+   celt_assert2(x>0, "celt_ilog2() only defined for strictly positive numbers");
+   return EC_ILOG(x)-1;
+}
+#endif
+
+#ifndef OVERRIDE_CELT_MAXABS16
+static inline opus_val16 celt_maxabs16(opus_val16 *x, int len)
+{
+   int i;
+   opus_val16 maxval = 0;
+   for (i=0;i<len;i++)
+      maxval = MAX16(maxval, ABS16(x[i]));
+   return maxval;
+}
+#endif
+
+#ifndef OVERRIDE_CELT_MAXABS32
+static inline opus_val32 celt_maxabs32(opus_val32 *x, int len)
+{
+   int i;
+   opus_val32 maxval = 0;
+   for (i=0;i<len;i++)
+      maxval = MAX32(maxval, ABS32(x[i]));
+   return maxval;
+}
+#endif
+
+/** Integer log in base2. Defined for zero, but not for negative numbers */
+static inline opus_int16 celt_zlog2(opus_val32 x)
+{
+   return x <= 0 ? 0 : celt_ilog2(x);
+}
+
+opus_val16 celt_rsqrt_norm(opus_val32 x);
+
+opus_val32 celt_sqrt(opus_val32 x);
+
+opus_val16 celt_cos_norm(opus_val32 x);
+
+static inline opus_val16 celt_log2(opus_val32 x)
+{
+   int i;
+   opus_val16 n, frac;
+   /* -0.41509302963303146, 0.9609890551383969, -0.31836011537636605,
+       0.15530808010959576, -0.08556153059057618 */
+   static const opus_val16 C[5] = {-6801+(1<<(13-DB_SHIFT)), 15746, -5217, 2545, -1401};
+   if (x==0)
+      return -32767;
+   i = celt_ilog2(x);
+   n = VSHR32(x,i-15)-32768-16384;
+   frac = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2], MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, C[4]))))))));
+   return SHL16(i-13,DB_SHIFT)+SHR16(frac,14-DB_SHIFT);
+}
+
+/*
+ K0 = 1
+ K1 = log(2)
+ K2 = 3-4*log(2)
+ K3 = 3*log(2) - 2
+*/
+#define D0 16383
+#define D1 22804
+#define D2 14819
+#define D3 10204
+/** Base-2 exponential approximation (2^x). (Q10 input, Q16 output) */
+static inline opus_val32 celt_exp2(opus_val16 x)
+{
+   int integer;
+   opus_val16 frac;
+   integer = SHR16(x,10);
+   if (integer>14)
+      return 0x7f000000;
+   else if (integer < -15)
+      return 0;
+   frac = SHL16(x-SHL16(integer,10),4);
+   frac = ADD16(D0, MULT16_16_Q15(frac, ADD16(D1, MULT16_16_Q15(frac, ADD16(D2 , MULT16_16_Q15(D3,frac))))));
+   return VSHR32(EXTEND32(frac), -integer-2);
+}
+
+opus_val32 celt_rcp(opus_val32 x);
+
+#define celt_div(a,b) MULT32_32_Q31((opus_val32)(a),celt_rcp(b))
+
+opus_val32 frac_div32(opus_val32 a, opus_val32 b);
+
+#define M1 32767
+#define M2 -21
+#define M3 -11943
+#define M4 4936
+
+/* Atan approximation using a 4th order polynomial. Input is in Q15 format
+   and normalized by pi/4. Output is in Q15 format */
+static inline opus_val16 celt_atan01(opus_val16 x)
+{
+   return MULT16_16_P15(x, ADD32(M1, MULT16_16_P15(x, ADD32(M2, MULT16_16_P15(x, ADD32(M3, MULT16_16_P15(M4, x)))))));
+}
+
+#undef M1
+#undef M2
+#undef M3
+#undef M4
+
+/* atan2() approximation valid for positive input values */
+static inline opus_val16 celt_atan2p(opus_val16 y, opus_val16 x)
+{
+   if (y < x)
+   {
+      opus_val32 arg;
+      arg = celt_div(SHL32(EXTEND32(y),15),x);
+      if (arg >= 32767)
+         arg = 32767;
+      return SHR16(celt_atan01(EXTRACT16(arg)),1);
+   } else {
+      opus_val32 arg;
+      arg = celt_div(SHL32(EXTEND32(x),15),y);
+      if (arg >= 32767)
+         arg = 32767;
+      return 25736-SHR16(celt_atan01(EXTRACT16(arg)),1);
+   }
+}
+
+#endif /* FIXED_POINT */
+#endif /* MATHOPS_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/mdct.c b/lib/rbcodec/codecs/libopus/celt/mdct.c
new file mode 100644
index 0000000000..abf4e79d8d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/mdct.c
@@ -0,0 +1,332 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2008 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/* This is a simple MDCT implementation that uses a N/4 complex FFT
+   to do most of the work. It should be relatively straightforward to
+   plug in pretty much and FFT here.
+
+   This replaces the Vorbis FFT (and uses the exact same API), which
+   was a bit too messy and that was ending up duplicating code
+   (might as well use the same FFT everywhere).
+
+   The algorithm is similar to (and inspired from) Fabrice Bellard's
+   MDCT implementation in FFMPEG, but has differences in signs, ordering
+   and scaling in many places.
+*/
+
+#ifndef SKIP_CONFIG_H
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+#endif
+
+#include "mdct.h"
+#include "kiss_fft.h"
+#include "_kiss_fft_guts.h"
+#include <math.h>
+#include "os_support.h"
+#include "mathops.h"
+#include "stack_alloc.h"
+
+#ifdef CUSTOM_MODES
+
+int clt_mdct_init(mdct_lookup *l,int N, int maxshift)
+{
+   int i;
+   int N4;
+   kiss_twiddle_scalar *trig;
+#if defined(FIXED_POINT)
+   int N2=N>>1;
+#endif
+   l->n = N;
+   N4 = N>>2;
+   l->maxshift = maxshift;
+   for (i=0;i<=maxshift;i++)
+   {
+      if (i==0)
+         l->kfft[i] = opus_fft_alloc(N>>2>>i, 0, 0);
+      else
+         l->kfft[i] = opus_fft_alloc_twiddles(N>>2>>i, 0, 0, l->kfft[0]);
+#ifndef ENABLE_TI_DSPLIB55
+      if (l->kfft[i]==NULL)
+         return 0;
+#endif
+   }
+   l->trig = trig = (kiss_twiddle_scalar*)opus_alloc((N4+1)*sizeof(kiss_twiddle_scalar));
+   if (l->trig==NULL)
+     return 0;
+   /* We have enough points that sine isn't necessary */
+#if defined(FIXED_POINT)
+   for (i=0;i<=N4;i++)
+      trig[i] = TRIG_UPSCALE*celt_cos_norm(DIV32(ADD32(SHL32(EXTEND32(i),17),N2),N));
+#else
+   for (i=0;i<=N4;i++)
+      trig[i] = (kiss_twiddle_scalar)cos(2*PI*i/N);
+#endif
+   return 1;
+}
+
+void clt_mdct_clear(mdct_lookup *l)
+{
+   int i;
+   for (i=0;i<=l->maxshift;i++)
+      opus_fft_free(l->kfft[i]);
+   opus_free((kiss_twiddle_scalar*)l->trig);
+}
+
+#endif /* CUSTOM_MODES */
+
+/* Forward MDCT trashes the input array */
+void clt_mdct_forward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out,
+      const opus_val16 *window, int overlap, int shift, int stride)
+{
+   int i;
+   int N, N2, N4;
+   kiss_twiddle_scalar sine;
+   VARDECL(kiss_fft_scalar, f);
+   SAVE_STACK;
+   N = l->n;
+   N >>= shift;
+   N2 = N>>1;
+   N4 = N>>2;
+   ALLOC(f, N2, kiss_fft_scalar);
+   /* sin(x) ~= x here */
+#ifdef FIXED_POINT
+   sine = TRIG_UPSCALE*(QCONST16(0.7853981f, 15)+N2)/N;
+#else
+   sine = (kiss_twiddle_scalar)2*PI*(.125f)/N;
+#endif
+
+   /* Consider the input to be composed of four blocks: [a, b, c, d] */
+   /* Window, shuffle, fold */
+   {
+      /* Temp pointers to make it really clear to the compiler what we're doing */
+      const kiss_fft_scalar * OPUS_RESTRICT xp1 = in+(overlap>>1);
+      const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+N2-1+(overlap>>1);
+      kiss_fft_scalar * OPUS_RESTRICT yp = f;
+      const opus_val16 * OPUS_RESTRICT wp1 = window+(overlap>>1);
+      const opus_val16 * OPUS_RESTRICT wp2 = window+(overlap>>1)-1;
+      for(i=0;i<(overlap>>2);i++)
+      {
+         /* Real part arranged as -d-cR, Imag part arranged as -b+aR*/
+         *yp++ = MULT16_32_Q15(*wp2, xp1[N2]) + MULT16_32_Q15(*wp1,*xp2);
+         *yp++ = MULT16_32_Q15(*wp1, *xp1)    - MULT16_32_Q15(*wp2, xp2[-N2]);
+         xp1+=2;
+         xp2-=2;
+         wp1+=2;
+         wp2-=2;
+      }
+      wp1 = window;
+      wp2 = window+overlap-1;
+      for(;i<N4-(overlap>>2);i++)
+      {
+         /* Real part arranged as a-bR, Imag part arranged as -c-dR */
+         *yp++ = *xp2;
+         *yp++ = *xp1;
+         xp1+=2;
+         xp2-=2;
+      }
+      for(;i<N4;i++)
+      {
+         /* Real part arranged as a-bR, Imag part arranged as -c-dR */
+         *yp++ =  -MULT16_32_Q15(*wp1, xp1[-N2]) + MULT16_32_Q15(*wp2, *xp2);
+         *yp++ = MULT16_32_Q15(*wp2, *xp1)     + MULT16_32_Q15(*wp1, xp2[N2]);
+         xp1+=2;
+         xp2-=2;
+         wp1+=2;
+         wp2-=2;
+      }
+   }
+   /* Pre-rotation */
+   {
+      kiss_fft_scalar * OPUS_RESTRICT yp = f;
+      const kiss_twiddle_scalar *t = &l->trig[0];
+      for(i=0;i<N4;i++)
+      {
+         kiss_fft_scalar re, im, yr, yi;
+         re = yp[0];
+         im = yp[1];
+         yr = -S_MUL(re,t[i<<shift])  -  S_MUL(im,t[(N4-i)<<shift]);
+         yi = -S_MUL(im,t[i<<shift])  +  S_MUL(re,t[(N4-i)<<shift]);
+         /* works because the cos is nearly one */
+         *yp++ = yr + S_MUL(yi,sine);
+         *yp++ = yi - S_MUL(yr,sine);
+      }
+   }
+
+   /* N/4 complex FFT, down-scales by 4/N */
+   opus_fft(l->kfft[shift], (kiss_fft_cpx *)f, (kiss_fft_cpx *)in);
+
+   /* Post-rotate */
+   {
+      /* Temp pointers to make it really clear to the compiler what we're doing */
+      const kiss_fft_scalar * OPUS_RESTRICT fp = in;
+      kiss_fft_scalar * OPUS_RESTRICT yp1 = out;
+      kiss_fft_scalar * OPUS_RESTRICT yp2 = out+stride*(N2-1);
+      const kiss_twiddle_scalar *t = &l->trig[0];
+      /* Temp pointers to make it really clear to the compiler what we're doing */
+      for(i=0;i<N4;i++)
+      {
+         kiss_fft_scalar yr, yi;
+         yr = S_MUL(fp[1],t[(N4-i)<<shift]) + S_MUL(fp[0],t[i<<shift]);
+         yi = S_MUL(fp[0],t[(N4-i)<<shift]) - S_MUL(fp[1],t[i<<shift]);
+         /* works because the cos is nearly one */
+         *yp1 = yr - S_MUL(yi,sine);
+         *yp2 = yi + S_MUL(yr,sine);;
+         fp += 2;
+         yp1 += 2*stride;
+         yp2 -= 2*stride;
+      }
+   }
+   RESTORE_STACK;
+}
+
+void clt_mdct_backward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out,
+      const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride)
+{
+   int i;
+   int N, N2, N4;
+   kiss_twiddle_scalar sine;
+   VARDECL(kiss_fft_scalar, f);
+   VARDECL(kiss_fft_scalar, f2);
+   SAVE_STACK;
+   N = l->n;
+   N >>= shift;
+   N2 = N>>1;
+   N4 = N>>2;
+   ALLOC(f, N2, kiss_fft_scalar);
+   ALLOC(f2, N2, kiss_fft_scalar);
+   /* sin(x) ~= x here */
+#ifdef FIXED_POINT
+   sine = TRIG_UPSCALE*(QCONST16(0.7853981f, 15)+N2)/N;
+#else
+   sine = (kiss_twiddle_scalar)2*PI*(.125f)/N;
+#endif
+
+   /* Pre-rotate */
+   {
+      /* Temp pointers to make it really clear to the compiler what we're doing */
+      const kiss_fft_scalar * OPUS_RESTRICT xp1 = in;
+      const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+stride*(N2-1);
+      kiss_fft_scalar * OPUS_RESTRICT yp = f2;
+      const kiss_twiddle_scalar *t = &l->trig[0];
+      for(i=0;i<N4;i++)
+      {
+         kiss_fft_scalar yr, yi;
+         yr = -S_MUL(*xp2, t[i<<shift]) + S_MUL(*xp1,t[(N4-i)<<shift]);
+         yi =  -S_MUL(*xp2, t[(N4-i)<<shift]) - S_MUL(*xp1,t[i<<shift]);
+         /* works because the cos is nearly one */
+         *yp++ = yr - S_MUL(yi,sine);
+         *yp++ = yi + S_MUL(yr,sine);
+         xp1+=2*stride;
+         xp2-=2*stride;
+      }
+   }
+
+   /* Inverse N/4 complex FFT. This one should *not* downscale even in fixed-point */
+   opus_ifft(l->kfft[shift], (kiss_fft_cpx *)f2, (kiss_fft_cpx *)f);
+
+   /* Post-rotate */
+   {
+      kiss_fft_scalar * OPUS_RESTRICT fp = f;
+      const kiss_twiddle_scalar *t = &l->trig[0];
+
+      for(i=0;i<N4;i++)
+      {
+         kiss_fft_scalar re, im, yr, yi;
+         re = fp[0];
+         im = fp[1];
+         /* We'd scale up by 2 here, but instead it's done when mixing the windows */
+         yr = S_MUL(re,t[i<<shift]) - S_MUL(im,t[(N4-i)<<shift]);
+         yi = S_MUL(im,t[i<<shift]) + S_MUL(re,t[(N4-i)<<shift]);
+         /* works because the cos is nearly one */
+         *fp++ = yr - S_MUL(yi,sine);
+         *fp++ = yi + S_MUL(yr,sine);
+      }
+   }
+   /* De-shuffle the components for the middle of the window only */
+   {
+      const kiss_fft_scalar * OPUS_RESTRICT fp1 = f;
+      const kiss_fft_scalar * OPUS_RESTRICT fp2 = f+N2-1;
+      kiss_fft_scalar * OPUS_RESTRICT yp = f2;
+      for(i = 0; i < N4; i++)
+      {
+         *yp++ =-*fp1;
+         *yp++ = *fp2;
+         fp1 += 2;
+         fp2 -= 2;
+      }
+   }
+   out -= (N2-overlap)>>1;
+   /* Mirror on both sides for TDAC */
+   {
+      kiss_fft_scalar * OPUS_RESTRICT fp1 = f2+N4-1;
+      kiss_fft_scalar * OPUS_RESTRICT xp1 = out+N2-1;
+      kiss_fft_scalar * OPUS_RESTRICT yp1 = out+N4-overlap/2;
+      const opus_val16 * OPUS_RESTRICT wp1 = window;
+      const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1;
+      for(i = 0; i< N4-overlap/2; i++)
+      {
+         *xp1 = *fp1;
+         xp1--;
+         fp1--;
+      }
+      for(; i < N4; i++)
+      {
+         kiss_fft_scalar x1;
+         x1 = *fp1--;
+         *yp1++ +=-MULT16_32_Q15(*wp1, x1);
+         *xp1-- += MULT16_32_Q15(*wp2, x1);
+         wp1++;
+         wp2--;
+      }
+   }
+   {
+      kiss_fft_scalar * OPUS_RESTRICT fp2 = f2+N4;
+      kiss_fft_scalar * OPUS_RESTRICT xp2 = out+N2;
+      kiss_fft_scalar * OPUS_RESTRICT yp2 = out+N-1-(N4-overlap/2);
+      const opus_val16 * OPUS_RESTRICT wp1 = window;
+      const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1;
+      for(i = 0; i< N4-overlap/2; i++)
+      {
+         *xp2 = *fp2;
+         xp2++;
+         fp2++;
+      }
+      for(; i < N4; i++)
+      {
+         kiss_fft_scalar x2;
+         x2 = *fp2++;
+         *yp2--  = MULT16_32_Q15(*wp1, x2);
+         *xp2++  = MULT16_32_Q15(*wp2, x2);
+         wp1++;
+         wp2--;
+      }
+   }
+   RESTORE_STACK;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/mdct.h b/lib/rbcodec/codecs/libopus/celt/mdct.h
new file mode 100644
index 0000000000..d72182138a
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/mdct.h
@@ -0,0 +1,70 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2008 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/* This is a simple MDCT implementation that uses a N/4 complex FFT
+   to do most of the work. It should be relatively straightforward to
+   plug in pretty much and FFT here.
+
+   This replaces the Vorbis FFT (and uses the exact same API), which
+   was a bit too messy and that was ending up duplicating code
+   (might as well use the same FFT everywhere).
+
+   The algorithm is similar to (and inspired from) Fabrice Bellard's
+   MDCT implementation in FFMPEG, but has differences in signs, ordering
+   and scaling in many places.
+*/
+
+#ifndef MDCT_H
+#define MDCT_H
+
+#include "opus_defines.h"
+#include "kiss_fft.h"
+#include "arch.h"
+
+typedef struct {
+   int n;
+   int maxshift;
+   const kiss_fft_state *kfft[4];
+   const kiss_twiddle_scalar * OPUS_RESTRICT trig;
+} mdct_lookup;
+
+int clt_mdct_init(mdct_lookup *l,int N, int maxshift);
+void clt_mdct_clear(mdct_lookup *l);
+
+/** Compute a forward MDCT and scale by 4/N, trashes the input array */
+void clt_mdct_forward(const mdct_lookup *l, kiss_fft_scalar *in,
+      kiss_fft_scalar * OPUS_RESTRICT out,
+      const opus_val16 *window, int overlap, int shift, int stride);
+
+/** Compute a backward MDCT (no scaling) and performs weighted overlap-add
+    (scales implicitly by 1/2) */
+void clt_mdct_backward(const mdct_lookup *l, kiss_fft_scalar *in,
+      kiss_fft_scalar * OPUS_RESTRICT out,
+      const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/mfrngcod.h b/lib/rbcodec/codecs/libopus/celt/mfrngcod.h
new file mode 100644
index 0000000000..809152a59a
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/mfrngcod.h
@@ -0,0 +1,48 @@
+/* Copyright (c) 2001-2008 Timothy B. Terriberry
+   Copyright (c) 2008-2009 Xiph.Org Foundation */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#if !defined(_mfrngcode_H)
+# define _mfrngcode_H (1)
+# include "entcode.h"
+
+/*Constants used by the entropy encoder/decoder.*/
+
+/*The number of bits to output at a time.*/
+# define EC_SYM_BITS   (8)
+/*The total number of bits in each of the state registers.*/
+# define EC_CODE_BITS  (32)
+/*The maximum symbol value.*/
+# define EC_SYM_MAX    ((1U<<EC_SYM_BITS)-1)
+/*Bits to shift by to move a symbol into the high-order position.*/
+# define EC_CODE_SHIFT (EC_CODE_BITS-EC_SYM_BITS-1)
+/*Carry bit of the high-order range symbol.*/
+# define EC_CODE_TOP   (((opus_uint32)1U)<<(EC_CODE_BITS-1))
+/*Low-order bit of the high-order range symbol.*/
+# define EC_CODE_BOT   (EC_CODE_TOP>>EC_SYM_BITS)
+/*The number of bits available for the last, partial symbol in the code field.*/
+# define EC_CODE_EXTRA ((EC_CODE_BITS-2)%EC_SYM_BITS+1)
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/modes.c b/lib/rbcodec/codecs/libopus/celt/modes.c
new file mode 100644
index 0000000000..d44cb3b9de
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/modes.c
@@ -0,0 +1,430 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "celt.h"
+#include "modes.h"
+#include "rate.h"
+#include "os_support.h"
+#include "stack_alloc.h"
+#include "quant_bands.h"
+
+static const opus_int16 eband5ms[] = {
+/*0  200 400 600 800  1k 1.2 1.4 1.6  2k 2.4 2.8 3.2  4k 4.8 5.6 6.8  8k 9.6 12k 15.6 */
+  0,  1,  2,  3,  4,  5,  6,  7,  8, 10, 12, 14, 16, 20, 24, 28, 34, 40, 48, 60, 78, 100
+};
+
+/* Alternate tuning (partially derived from Vorbis) */
+#define BITALLOC_SIZE 11
+/* Bit allocation table in units of 1/32 bit/sample (0.1875 dB SNR) */
+static const unsigned char band_allocation[] = {
+/*0  200 400 600 800  1k 1.2 1.4 1.6  2k 2.4 2.8 3.2  4k 4.8 5.6 6.8  8k 9.6 12k 15.6 */
+  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
+ 90, 80, 75, 69, 63, 56, 49, 40, 34, 29, 20, 18, 10,  0,  0,  0,  0,  0,  0,  0,  0,
+110,100, 90, 84, 78, 71, 65, 58, 51, 45, 39, 32, 26, 20, 12,  0,  0,  0,  0,  0,  0,
+118,110,103, 93, 86, 80, 75, 70, 65, 59, 53, 47, 40, 31, 23, 15,  4,  0,  0,  0,  0,
+126,119,112,104, 95, 89, 83, 78, 72, 66, 60, 54, 47, 39, 32, 25, 17, 12,  1,  0,  0,
+134,127,120,114,103, 97, 91, 85, 78, 72, 66, 60, 54, 47, 41, 35, 29, 23, 16, 10,  1,
+144,137,130,124,113,107,101, 95, 88, 82, 76, 70, 64, 57, 51, 45, 39, 33, 26, 15,  1,
+152,145,138,132,123,117,111,105, 98, 92, 86, 80, 74, 67, 61, 55, 49, 43, 36, 20,  1,
+162,155,148,142,133,127,121,115,108,102, 96, 90, 84, 77, 71, 65, 59, 53, 46, 30,  1,
+172,165,158,152,143,137,131,125,118,112,106,100, 94, 87, 81, 75, 69, 63, 56, 45, 20,
+200,200,200,200,200,200,200,200,198,193,188,183,178,173,168,163,158,153,148,129,104,
+};
+
+#ifndef CUSTOM_MODES_ONLY
+ #ifdef FIXED_POINT
+  #include "static_modes_fixed.h"
+ #else
+  #include "static_modes_float.h"
+ #endif
+#endif /* CUSTOM_MODES_ONLY */
+
+#ifndef M_PI
+#define M_PI 3.141592653
+#endif
+
+#ifdef CUSTOM_MODES
+
+/* Defining 25 critical bands for the full 0-20 kHz audio bandwidth
+   Taken from http://ccrma.stanford.edu/~jos/bbt/Bark_Frequency_Scale.html */
+#define BARK_BANDS 25
+static const opus_int16 bark_freq[BARK_BANDS+1] = {
+      0,   100,   200,   300,   400,
+    510,   630,   770,   920,  1080,
+   1270,  1480,  1720,  2000,  2320,
+   2700,  3150,  3700,  4400,  5300,
+   6400,  7700,  9500, 12000, 15500,
+  20000};
+
+static opus_int16 *compute_ebands(opus_int32 Fs, int frame_size, int res, int *nbEBands)
+{
+   opus_int16 *eBands;
+   int i, j, lin, low, high, nBark, offset=0;
+
+   /* All modes that have 2.5 ms short blocks use the same definition */
+   if (Fs == 400*(opus_int32)frame_size)
+   {
+      *nbEBands = sizeof(eband5ms)/sizeof(eband5ms[0])-1;
+      eBands = opus_alloc(sizeof(opus_int16)*(*nbEBands+1));
+      for (i=0;i<*nbEBands+1;i++)
+         eBands[i] = eband5ms[i];
+      return eBands;
+   }
+   /* Find the number of critical bands supported by our sampling rate */
+   for (nBark=1;nBark<BARK_BANDS;nBark++)
+    if (bark_freq[nBark+1]*2 >= Fs)
+       break;
+
+   /* Find where the linear part ends (i.e. where the spacing is more than min_width */
+   for (lin=0;lin<nBark;lin++)
+      if (bark_freq[lin+1]-bark_freq[lin] >= res)
+         break;
+
+   low = (bark_freq[lin]+res/2)/res;
+   high = nBark-lin;
+   *nbEBands = low+high;
+   eBands = opus_alloc(sizeof(opus_int16)*(*nbEBands+2));
+
+   if (eBands==NULL)
+      return NULL;
+
+   /* Linear spacing (min_width) */
+   for (i=0;i<low;i++)
+      eBands[i] = i;
+   if (low>0)
+      offset = eBands[low-1]*res - bark_freq[lin-1];
+   /* Spacing follows critical bands */
+   for (i=0;i<high;i++)
+   {
+      int target = bark_freq[lin+i];
+      /* Round to an even value */
+      eBands[i+low] = (target+offset/2+res)/(2*res)*2;
+      offset = eBands[i+low]*res - target;
+   }
+   /* Enforce the minimum spacing at the boundary */
+   for (i=0;i<*nbEBands;i++)
+      if (eBands[i] < i)
+         eBands[i] = i;
+   /* Round to an even value */
+   eBands[*nbEBands] = (bark_freq[nBark]+res)/(2*res)*2;
+   if (eBands[*nbEBands] > frame_size)
+      eBands[*nbEBands] = frame_size;
+   for (i=1;i<*nbEBands-1;i++)
+   {
+      if (eBands[i+1]-eBands[i] < eBands[i]-eBands[i-1])
+      {
+         eBands[i] -= (2*eBands[i]-eBands[i-1]-eBands[i+1])/2;
+      }
+   }
+   /* Remove any empty bands. */
+   for (i=j=0;i<*nbEBands;i++)
+      if(eBands[i+1]>eBands[j])
+         eBands[++j]=eBands[i+1];
+   *nbEBands=j;
+
+   for (i=1;i<*nbEBands;i++)
+   {
+      /* Every band must be smaller than the last band. */
+      celt_assert(eBands[i]-eBands[i-1]<=eBands[*nbEBands]-eBands[*nbEBands-1]);
+      /* Each band must be no larger than twice the size of the previous one. */
+      celt_assert(eBands[i+1]-eBands[i]<=2*(eBands[i]-eBands[i-1]));
+   }
+
+   return eBands;
+}
+
+static void compute_allocation_table(CELTMode *mode)
+{
+   int i, j;
+   unsigned char *allocVectors;
+   int maxBands = sizeof(eband5ms)/sizeof(eband5ms[0])-1;
+
+   mode->nbAllocVectors = BITALLOC_SIZE;
+   allocVectors = opus_alloc(sizeof(unsigned char)*(BITALLOC_SIZE*mode->nbEBands));
+   if (allocVectors==NULL)
+      return;
+
+   /* Check for standard mode */
+   if (mode->Fs == 400*(opus_int32)mode->shortMdctSize)
+   {
+      for (i=0;i<BITALLOC_SIZE*mode->nbEBands;i++)
+         allocVectors[i] = band_allocation[i];
+      mode->allocVectors = allocVectors;
+      return;
+   }
+   /* If not the standard mode, interpolate */
+   /* Compute per-codec-band allocation from per-critical-band matrix */
+   for (i=0;i<BITALLOC_SIZE;i++)
+   {
+      for (j=0;j<mode->nbEBands;j++)
+      {
+         int k;
+         for (k=0;k<maxBands;k++)
+         {
+            if (400*(opus_int32)eband5ms[k] > mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize)
+               break;
+         }
+         if (k>maxBands-1)
+            allocVectors[i*mode->nbEBands+j] = band_allocation[i*maxBands + maxBands-1];
+         else {
+            opus_int32 a0, a1;
+            a1 = mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize - 400*(opus_int32)eband5ms[k-1];
+            a0 = 400*(opus_int32)eband5ms[k] - mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize;
+            allocVectors[i*mode->nbEBands+j] = (a0*band_allocation[i*maxBands+k-1]
+                                             + a1*band_allocation[i*maxBands+k])/(a0+a1);
+         }
+      }
+   }
+
+   /*printf ("\n");
+   for (i=0;i<BITALLOC_SIZE;i++)
+   {
+      for (j=0;j<mode->nbEBands;j++)
+         printf ("%d ", allocVectors[i*mode->nbEBands+j]);
+      printf ("\n");
+   }
+   exit(0);*/
+
+   mode->allocVectors = allocVectors;
+}
+
+#endif /* CUSTOM_MODES */
+
+CELTMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error)
+{
+   int i;
+#ifdef CUSTOM_MODES
+   CELTMode *mode=NULL;
+   int res;
+   opus_val16 *window;
+   opus_int16 *logN;
+   int LM;
+   ALLOC_STACK;
+#if !defined(VAR_ARRAYS) && !defined(USE_ALLOCA)
+   if (global_stack==NULL)
+      goto failure;
+#endif
+#endif
+
+#ifndef CUSTOM_MODES_ONLY
+   for (i=0;i<TOTAL_MODES;i++)
+   {
+      int j;
+      for (j=0;j<4;j++)
+      {
+         if (Fs == static_mode_list[i]->Fs &&
+               (frame_size<<j) == static_mode_list[i]->shortMdctSize*static_mode_list[i]->nbShortMdcts)
+         {
+            if (error)
+               *error = OPUS_OK;
+            return (CELTMode*)static_mode_list[i];
+         }
+      }
+   }
+#endif /* CUSTOM_MODES_ONLY */
+
+#ifndef CUSTOM_MODES
+   if (error)
+      *error = OPUS_BAD_ARG;
+   return NULL;
+#else
+
+   /* The good thing here is that permutation of the arguments will automatically be invalid */
+
+   if (Fs < 8000 || Fs > 96000)
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+   if (frame_size < 40 || frame_size > 1024 || frame_size%2!=0)
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+   /* Frames of less than 1ms are not supported. */
+   if ((opus_int32)frame_size*1000 < Fs)
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+
+   if ((opus_int32)frame_size*75 >= Fs && (frame_size%16)==0)
+   {
+     LM = 3;
+   } else if ((opus_int32)frame_size*150 >= Fs && (frame_size%8)==0)
+   {
+     LM = 2;
+   } else if ((opus_int32)frame_size*300 >= Fs && (frame_size%4)==0)
+   {
+     LM = 1;
+   } else
+   {
+     LM = 0;
+   }
+
+   /* Shorts longer than 3.3ms are not supported. */
+   if ((opus_int32)(frame_size>>LM)*300 > Fs)
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+
+   mode = opus_alloc(sizeof(CELTMode));
+   if (mode==NULL)
+      goto failure;
+   mode->Fs = Fs;
+
+   /* Pre/de-emphasis depends on sampling rate. The "standard" pre-emphasis
+      is defined as A(z) = 1 - 0.85*z^-1 at 48 kHz. Other rates should
+      approximate that. */
+   if(Fs < 12000) /* 8 kHz */
+   {
+      mode->preemph[0] =  QCONST16(0.3500061035f, 15);
+      mode->preemph[1] = -QCONST16(0.1799926758f, 15);
+      mode->preemph[2] =  QCONST16(0.2719968125f, SIG_SHIFT); /* exact 1/preemph[3] */
+      mode->preemph[3] =  QCONST16(3.6765136719f, 13);
+   } else if(Fs < 24000) /* 16 kHz */
+   {
+      mode->preemph[0] =  QCONST16(0.6000061035f, 15);
+      mode->preemph[1] = -QCONST16(0.1799926758f, 15);
+      mode->preemph[2] =  QCONST16(0.4424998650f, SIG_SHIFT); /* exact 1/preemph[3] */
+      mode->preemph[3] =  QCONST16(2.2598876953f, 13);
+   } else if(Fs < 40000) /* 32 kHz */
+   {
+      mode->preemph[0] =  QCONST16(0.7799987793f, 15);
+      mode->preemph[1] = -QCONST16(0.1000061035f, 15);
+      mode->preemph[2] =  QCONST16(0.7499771125f, SIG_SHIFT); /* exact 1/preemph[3] */
+      mode->preemph[3] =  QCONST16(1.3333740234f, 13);
+   } else /* 48 kHz */
+   {
+      mode->preemph[0] =  QCONST16(0.8500061035f, 15);
+      mode->preemph[1] =  QCONST16(0.0f, 15);
+      mode->preemph[2] =  QCONST16(1.f, SIG_SHIFT);
+      mode->preemph[3] =  QCONST16(1.f, 13);
+   }
+
+   mode->maxLM = LM;
+   mode->nbShortMdcts = 1<<LM;
+   mode->shortMdctSize = frame_size/mode->nbShortMdcts;
+   res = (mode->Fs+mode->shortMdctSize)/(2*mode->shortMdctSize);
+
+   mode->eBands = compute_ebands(Fs, mode->shortMdctSize, res, &mode->nbEBands);
+   if (mode->eBands==NULL)
+      goto failure;
+
+   mode->effEBands = mode->nbEBands;
+   while (mode->eBands[mode->effEBands] > mode->shortMdctSize)
+      mode->effEBands--;
+
+   /* Overlap must be divisible by 4 */
+   mode->overlap = ((mode->shortMdctSize>>2)<<2);
+
+   compute_allocation_table(mode);
+   if (mode->allocVectors==NULL)
+      goto failure;
+
+   window = (opus_val16*)opus_alloc(mode->overlap*sizeof(opus_val16));
+   if (window==NULL)
+      goto failure;
+
+#ifndef FIXED_POINT
+   for (i=0;i<mode->overlap;i++)
+      window[i] = Q15ONE*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap));
+#else
+   for (i=0;i<mode->overlap;i++)
+      window[i] = MIN32(32767,floor(.5+32768.*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap))));
+#endif
+   mode->window = window;
+
+   logN = (opus_int16*)opus_alloc(mode->nbEBands*sizeof(opus_int16));
+   if (logN==NULL)
+      goto failure;
+
+   for (i=0;i<mode->nbEBands;i++)
+      logN[i] = log2_frac(mode->eBands[i+1]-mode->eBands[i], BITRES);
+   mode->logN = logN;
+
+   compute_pulse_cache(mode, mode->maxLM);
+
+   if (clt_mdct_init(&mode->mdct, 2*mode->shortMdctSize*mode->nbShortMdcts,
+           mode->maxLM) == 0)
+      goto failure;
+
+   if (error)
+      *error = OPUS_OK;
+
+   return mode;
+failure:
+   if (error)
+      *error = OPUS_ALLOC_FAIL;
+   if (mode!=NULL)
+      opus_custom_mode_destroy(mode);
+   return NULL;
+#endif /* !CUSTOM_MODES */
+}
+
+#ifdef CUSTOM_MODES
+void opus_custom_mode_destroy(CELTMode *mode)
+{
+   if (mode == NULL)
+      return;
+#ifndef CUSTOM_MODES_ONLY
+   {
+     int i;
+     for (i=0;i<TOTAL_MODES;i++)
+     {
+        if (mode == static_mode_list[i])
+        {
+           return;
+        }
+     }
+   }
+#endif /* CUSTOM_MODES_ONLY */
+   opus_free((opus_int16*)mode->eBands);
+   opus_free((opus_int16*)mode->allocVectors);
+
+   opus_free((opus_val16*)mode->window);
+   opus_free((opus_int16*)mode->logN);
+
+   opus_free((opus_int16*)mode->cache.index);
+   opus_free((unsigned char*)mode->cache.bits);
+   opus_free((unsigned char*)mode->cache.caps);
+   clt_mdct_clear(&mode->mdct);
+
+   opus_free((CELTMode *)mode);
+}
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/modes.h b/lib/rbcodec/codecs/libopus/celt/modes.h
new file mode 100644
index 0000000000..c8340f9875
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/modes.h
@@ -0,0 +1,83 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef MODES_H
+#define MODES_H
+
+#include "opus_types.h"
+#include "celt.h"
+#include "arch.h"
+#include "mdct.h"
+#include "entenc.h"
+#include "entdec.h"
+
+#define MAX_PERIOD 1024
+
+#ifndef OVERLAP
+#define OVERLAP(mode) ((mode)->overlap)
+#endif
+
+#ifndef FRAMESIZE
+#define FRAMESIZE(mode) ((mode)->mdctSize)
+#endif
+
+typedef struct {
+   int size;
+   const opus_int16 *index;
+   const unsigned char *bits;
+   const unsigned char *caps;
+} PulseCache;
+
+/** Mode definition (opaque)
+ @brief Mode definition
+ */
+struct OpusCustomMode {
+   opus_int32 Fs;
+   int          overlap;
+
+   int          nbEBands;
+   int          effEBands;
+   opus_val16    preemph[4];
+   const opus_int16   *eBands;   /**< Definition for each "pseudo-critical band" */
+
+   int         maxLM;
+   int         nbShortMdcts;
+   int         shortMdctSize;
+
+   int          nbAllocVectors; /**< Number of lines in the matrix below */
+   const unsigned char   *allocVectors;   /**< Number of bits in each band for several rates */
+   const opus_int16 *logN;
+
+   const opus_val16 *window;
+   mdct_lookup mdct;
+   PulseCache cache;
+};
+
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/os_support.h b/lib/rbcodec/codecs/libopus/celt/os_support.h
new file mode 100644
index 0000000000..2484f0b2f7
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/os_support.h
@@ -0,0 +1,89 @@
+/* Copyright (C) 2007 Jean-Marc Valin
+
+   File: os_support.h
+   This is the (tiny) OS abstraction layer. Aside from math.h, this is the
+   only place where system headers are allowed.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are
+   met:
+
+   1. Redistributions of source code must retain the above copyright notice,
+   this list of conditions and the following disclaimer.
+
+   2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
+   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
+   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
+   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef OS_SUPPORT_H
+#define OS_SUPPORT_H
+
+#ifdef CUSTOM_SUPPORT
+#  include "custom_support.h"
+#endif
+
+#include <string.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+/** Opus wrapper for malloc(). To do your own dynamic allocation, all you need to do is replace this function and opus_free */
+#ifndef OVERRIDE_OPUS_ALLOC
+static inline void *opus_alloc (size_t size)
+{
+   return malloc(size);
+}
+#endif
+
+/** Same as celt_alloc(), except that the area is only needed inside a CELT call (might cause problem with wideband though) */
+#ifndef OVERRIDE_OPUS_ALLOC_SCRATCH
+static inline void *opus_alloc_scratch (size_t size)
+{
+   /* Scratch space doesn't need to be cleared */
+   return opus_alloc(size);
+}
+#endif
+
+/** Opus wrapper for free(). To do your own dynamic allocation, all you need to do is replace this function and opus_alloc */
+#ifndef OVERRIDE_OPUS_FREE
+static inline void opus_free (void *ptr)
+{
+   free(ptr);
+}
+#endif
+
+/** Copy n bytes of memory from src to dst. The 0* term provides compile-time type checking  */
+#ifndef OVERRIDE_OPUS_COPY
+#define OPUS_COPY(dst, src, n) (memcpy((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) ))
+#endif
+
+/** Copy n bytes of memory from src to dst, allowing overlapping regions. The 0* term
+    provides compile-time type checking */
+#ifndef OVERRIDE_OPUS_MOVE
+#define OPUS_MOVE(dst, src, n) (memmove((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) ))
+#endif
+
+/** Set n elements of dst to zero, starting at address s */
+#ifndef OVERRIDE_OPUS_CLEAR
+#define OPUS_CLEAR(dst, n) (memset((dst), 0, (n)*sizeof(*(dst))))
+#endif
+
+/*#ifdef __GNUC__
+#pragma GCC poison printf sprintf
+#pragma GCC poison malloc free realloc calloc
+#endif*/
+
+#endif /* OS_SUPPORT_H */
+
diff --git a/lib/rbcodec/codecs/libopus/celt/pitch.c b/lib/rbcodec/codecs/libopus/celt/pitch.c
new file mode 100644
index 0000000000..3bad8e46a7
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/pitch.c
@@ -0,0 +1,410 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file pitch.c
+   @brief Pitch analysis
+ */
+
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "pitch.h"
+#include "os_support.h"
+#include "modes.h"
+#include "stack_alloc.h"
+#include "mathops.h"
+#include "celt_lpc.h"
+
+static void find_best_pitch(opus_val32 *xcorr, opus_val16 *y, int len,
+                            int max_pitch, int *best_pitch
+#ifdef FIXED_POINT
+                            , int yshift, opus_val32 maxcorr
+#endif
+                            )
+{
+   int i, j;
+   opus_val32 Syy=1;
+   opus_val16 best_num[2];
+   opus_val32 best_den[2];
+#ifdef FIXED_POINT
+   int xshift;
+
+   xshift = celt_ilog2(maxcorr)-14;
+#endif
+
+   best_num[0] = -1;
+   best_num[1] = -1;
+   best_den[0] = 0;
+   best_den[1] = 0;
+   best_pitch[0] = 0;
+   best_pitch[1] = 1;
+   for (j=0;j<len;j++)
+      Syy = ADD32(Syy, SHR32(MULT16_16(y[j],y[j]), yshift));
+   for (i=0;i<max_pitch;i++)
+   {
+      if (xcorr[i]>0)
+      {
+         opus_val16 num;
+         opus_val32 xcorr16;
+         xcorr16 = EXTRACT16(VSHR32(xcorr[i], xshift));
+#ifndef FIXED_POINT
+         /* Considering the range of xcorr16, this should avoid both underflows
+            and overflows (inf) when squaring xcorr16 */
+         xcorr16 *= 1e-12;
+#endif
+         num = MULT16_16_Q15(xcorr16,xcorr16);
+         if (MULT16_32_Q15(num,best_den[1]) > MULT16_32_Q15(best_num[1],Syy))
+         {
+            if (MULT16_32_Q15(num,best_den[0]) > MULT16_32_Q15(best_num[0],Syy))
+            {
+               best_num[1] = best_num[0];
+               best_den[1] = best_den[0];
+               best_pitch[1] = best_pitch[0];
+               best_num[0] = num;
+               best_den[0] = Syy;
+               best_pitch[0] = i;
+            } else {
+               best_num[1] = num;
+               best_den[1] = Syy;
+               best_pitch[1] = i;
+            }
+         }
+      }
+      Syy += SHR32(MULT16_16(y[i+len],y[i+len]),yshift) - SHR32(MULT16_16(y[i],y[i]),yshift);
+      Syy = MAX32(1, Syy);
+   }
+}
+
+void pitch_downsample(celt_sig * OPUS_RESTRICT x[], opus_val16 * OPUS_RESTRICT x_lp,
+      int len, int C)
+{
+   int i;
+   opus_val32 ac[5];
+   opus_val16 tmp=Q15ONE;
+   opus_val16 lpc[4], mem[4]={0,0,0,0};
+#ifdef FIXED_POINT
+   int shift;
+   opus_val32 maxabs = celt_maxabs32(x[0], len);
+   if (C==2)
+   {
+      opus_val32 maxabs_1 = celt_maxabs32(x[1], len);
+      maxabs = MAX32(maxabs, maxabs_1);
+   }
+   if (maxabs<1)
+      maxabs=1;
+   shift = celt_ilog2(maxabs)-10;
+   if (shift<0)
+      shift=0;
+   if (C==2)
+      shift++;
+#endif
+   for (i=1;i<len>>1;i++)
+      x_lp[i] = SHR32(HALF32(HALF32(x[0][(2*i-1)]+x[0][(2*i+1)])+x[0][2*i]), shift);
+   x_lp[0] = SHR32(HALF32(HALF32(x[0][1])+x[0][0]), shift);
+   if (C==2)
+   {
+      for (i=1;i<len>>1;i++)
+         x_lp[i] += SHR32(HALF32(HALF32(x[1][(2*i-1)]+x[1][(2*i+1)])+x[1][2*i]), shift);
+      x_lp[0] += SHR32(HALF32(HALF32(x[1][1])+x[1][0]), shift);
+   }
+
+   _celt_autocorr(x_lp, ac, NULL, 0,
+                  4, len>>1);
+
+   /* Noise floor -40 dB */
+#ifdef FIXED_POINT
+   ac[0] += SHR32(ac[0],13);
+#else
+   ac[0] *= 1.0001f;
+#endif
+   /* Lag windowing */
+   for (i=1;i<=4;i++)
+   {
+      /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/
+#ifdef FIXED_POINT
+      ac[i] -= MULT16_32_Q15(2*i*i, ac[i]);
+#else
+      ac[i] -= ac[i]*(.008f*i)*(.008f*i);
+#endif
+   }
+
+   _celt_lpc(lpc, ac, 4);
+   for (i=0;i<4;i++)
+   {
+      tmp = MULT16_16_Q15(QCONST16(.9f,15), tmp);
+      lpc[i] = MULT16_16_Q15(lpc[i], tmp);
+   }
+   celt_fir(x_lp, lpc, x_lp, len>>1, 4, mem);
+
+   mem[0]=0;
+   lpc[0]=QCONST16(.8f,12);
+   celt_fir(x_lp, lpc, x_lp, len>>1, 1, mem);
+
+}
+
+void pitch_search(const opus_val16 * OPUS_RESTRICT x_lp, opus_val16 * OPUS_RESTRICT y,
+                  int len, int max_pitch, int *pitch)
+{
+   int i, j;
+   int lag;
+   int best_pitch[2]={0,0};
+   VARDECL(opus_val16, x_lp4);
+   VARDECL(opus_val16, y_lp4);
+   VARDECL(opus_val32, xcorr);
+#ifdef FIXED_POINT
+   opus_val32 maxcorr=1;
+   opus_val16 xmax, ymax;
+   int shift=0;
+#endif
+   int offset;
+
+   SAVE_STACK;
+
+   celt_assert(len>0);
+   celt_assert(max_pitch>0);
+   lag = len+max_pitch;
+
+   ALLOC(x_lp4, len>>2, opus_val16);
+   ALLOC(y_lp4, lag>>2, opus_val16);
+   ALLOC(xcorr, max_pitch>>1, opus_val32);
+
+   /* Downsample by 2 again */
+   for (j=0;j<len>>2;j++)
+      x_lp4[j] = x_lp[2*j];
+   for (j=0;j<lag>>2;j++)
+      y_lp4[j] = y[2*j];
+
+#ifdef FIXED_POINT
+   xmax = celt_maxabs16(x_lp4, len>>2);
+   ymax = celt_maxabs16(y_lp4, lag>>2);
+   shift = celt_ilog2(MAX16(1, MAX16(xmax, ymax)))-11;
+   if (shift>0)
+   {
+      for (j=0;j<len>>2;j++)
+         x_lp4[j] = SHR16(x_lp4[j], shift);
+      for (j=0;j<lag>>2;j++)
+         y_lp4[j] = SHR16(y_lp4[j], shift);
+      /* Use double the shift for a MAC */
+      shift *= 2;
+   } else {
+      shift = 0;
+   }
+#endif
+
+   /* Coarse search with 4x decimation */
+
+   for (i=0;i<max_pitch>>2;i++)
+   {
+      opus_val32 sum = 0;
+      for (j=0;j<len>>2;j++)
+         sum = MAC16_16(sum, x_lp4[j],y_lp4[i+j]);
+      xcorr[i] = MAX32(-1, sum);
+#ifdef FIXED_POINT
+      maxcorr = MAX32(maxcorr, sum);
+#endif
+   }
+   find_best_pitch(xcorr, y_lp4, len>>2, max_pitch>>2, best_pitch
+#ifdef FIXED_POINT
+                   , 0, maxcorr
+#endif
+                   );
+
+   /* Finer search with 2x decimation */
+#ifdef FIXED_POINT
+   maxcorr=1;
+#endif
+   for (i=0;i<max_pitch>>1;i++)
+   {
+      opus_val32 sum=0;
+      xcorr[i] = 0;
+      if (abs(i-2*best_pitch[0])>2 && abs(i-2*best_pitch[1])>2)
+         continue;
+      for (j=0;j<len>>1;j++)
+         sum += SHR32(MULT16_16(x_lp[j],y[i+j]), shift);
+      xcorr[i] = MAX32(-1, sum);
+#ifdef FIXED_POINT
+      maxcorr = MAX32(maxcorr, sum);
+#endif
+   }
+   find_best_pitch(xcorr, y, len>>1, max_pitch>>1, best_pitch
+#ifdef FIXED_POINT
+                   , shift+1, maxcorr
+#endif
+                   );
+
+   /* Refine by pseudo-interpolation */
+   if (best_pitch[0]>0 && best_pitch[0]<(max_pitch>>1)-1)
+   {
+      opus_val32 a, b, c;
+      a = xcorr[best_pitch[0]-1];
+      b = xcorr[best_pitch[0]];
+      c = xcorr[best_pitch[0]+1];
+      if ((c-a) > MULT16_32_Q15(QCONST16(.7f,15),b-a))
+         offset = 1;
+      else if ((a-c) > MULT16_32_Q15(QCONST16(.7f,15),b-c))
+         offset = -1;
+      else
+         offset = 0;
+   } else {
+      offset = 0;
+   }
+   *pitch = 2*best_pitch[0]-offset;
+
+   RESTORE_STACK;
+}
+
+static const int second_check[16] = {0, 0, 3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2};
+opus_val16 remove_doubling(opus_val16 *x, int maxperiod, int minperiod,
+      int N, int *T0_, int prev_period, opus_val16 prev_gain)
+{
+   int k, i, T, T0;
+   opus_val16 g, g0;
+   opus_val16 pg;
+   opus_val32 xy,xx,yy;
+   opus_val32 xcorr[3];
+   opus_val32 best_xy, best_yy;
+   int offset;
+   int minperiod0;
+
+   minperiod0 = minperiod;
+   maxperiod /= 2;
+   minperiod /= 2;
+   *T0_ /= 2;
+   prev_period /= 2;
+   N /= 2;
+   x += maxperiod;
+   if (*T0_>=maxperiod)
+      *T0_=maxperiod-1;
+
+   T = T0 = *T0_;
+   xx=xy=yy=0;
+   for (i=0;i<N;i++)
+   {
+      xy = MAC16_16(xy, x[i], x[i-T0]);
+      xx = MAC16_16(xx, x[i], x[i]);
+      yy = MAC16_16(yy, x[i-T0],x[i-T0]);
+   }
+   best_xy = xy;
+   best_yy = yy;
+#ifdef FIXED_POINT
+      {
+         opus_val32 x2y2;
+         int sh, t;
+         x2y2 = 1+HALF32(MULT32_32_Q31(xx,yy));
+         sh = celt_ilog2(x2y2)>>1;
+         t = VSHR32(x2y2, 2*(sh-7));
+         g = g0 = VSHR32(MULT16_32_Q15(celt_rsqrt_norm(t), xy),sh+1);
+      }
+#else
+      g = g0 = xy/celt_sqrt(1+xx*yy);
+#endif
+   /* Look for any pitch at T/k */
+   for (k=2;k<=15;k++)
+   {
+      int T1, T1b;
+      opus_val16 g1;
+      opus_val16 cont=0;
+      T1 = (2*T0+k)/(2*k);
+      if (T1 < minperiod)
+         break;
+      /* Look for another strong correlation at T1b */
+      if (k==2)
+      {
+         if (T1+T0>maxperiod)
+            T1b = T0;
+         else
+            T1b = T0+T1;
+      } else
+      {
+         T1b = (2*second_check[k]*T0+k)/(2*k);
+      }
+      xy=yy=0;
+      for (i=0;i<N;i++)
+      {
+         xy = MAC16_16(xy, x[i], x[i-T1]);
+         yy = MAC16_16(yy, x[i-T1], x[i-T1]);
+
+         xy = MAC16_16(xy, x[i], x[i-T1b]);
+         yy = MAC16_16(yy, x[i-T1b], x[i-T1b]);
+      }
+#ifdef FIXED_POINT
+      {
+         opus_val32 x2y2;
+         int sh, t;
+         x2y2 = 1+MULT32_32_Q31(xx,yy);
+         sh = celt_ilog2(x2y2)>>1;
+         t = VSHR32(x2y2, 2*(sh-7));
+         g1 = VSHR32(MULT16_32_Q15(celt_rsqrt_norm(t), xy),sh+1);
+      }
+#else
+      g1 = xy/celt_sqrt(1+2.f*xx*1.f*yy);
+#endif
+      if (abs(T1-prev_period)<=1)
+         cont = prev_gain;
+      else if (abs(T1-prev_period)<=2 && 5*k*k < T0)
+         cont = HALF32(prev_gain);
+      else
+         cont = 0;
+      if (g1 > QCONST16(.3f,15) + MULT16_16_Q15(QCONST16(.4f,15),g0)-cont)
+      {
+         best_xy = xy;
+         best_yy = yy;
+         T = T1;
+         g = g1;
+      }
+   }
+   best_xy = MAX32(0, best_xy);
+   if (best_yy <= best_xy)
+      pg = Q15ONE;
+   else
+      pg = SHR32(frac_div32(best_xy,best_yy+1),16);
+
+   for (k=0;k<3;k++)
+   {
+      int T1 = T+k-1;
+      xy = 0;
+      for (i=0;i<N;i++)
+         xy = MAC16_16(xy, x[i], x[i-T1]);
+      xcorr[k] = xy;
+   }
+   if ((xcorr[2]-xcorr[0]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[0]))
+      offset = 1;
+   else if ((xcorr[0]-xcorr[2]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[2]))
+      offset = -1;
+   else
+      offset = 0;
+   if (pg > g)
+      pg = g;
+   *T0_ = 2*T+offset;
+
+   if (*T0_<minperiod0)
+      *T0_=minperiod0;
+   return pg;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/pitch.h b/lib/rbcodec/codecs/libopus/celt/pitch.h
new file mode 100644
index 0000000000..2757071a6f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/pitch.h
@@ -0,0 +1,48 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file pitch.h
+   @brief Pitch analysis
+ */
+
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef PITCH_H
+#define PITCH_H
+
+#include "modes.h"
+
+void pitch_downsample(celt_sig * OPUS_RESTRICT x[], opus_val16 * OPUS_RESTRICT x_lp,
+      int len, int C);
+
+void pitch_search(const opus_val16 * OPUS_RESTRICT x_lp, opus_val16 * OPUS_RESTRICT y,
+                  int len, int max_pitch, int *pitch);
+
+opus_val16 remove_doubling(opus_val16 *x, int maxperiod, int minperiod,
+      int N, int *T0, int prev_period, opus_val16 prev_gain);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/quant_bands.c b/lib/rbcodec/codecs/libopus/celt/quant_bands.c
new file mode 100644
index 0000000000..5ad5311f84
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/quant_bands.c
@@ -0,0 +1,567 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "quant_bands.h"
+#include "laplace.h"
+#include <math.h>
+#include "os_support.h"
+#include "arch.h"
+#include "mathops.h"
+#include "stack_alloc.h"
+#include "rate.h"
+
+#ifdef FIXED_POINT
+/* Mean energy in each band quantized in Q6 */
+static const signed char eMeans[25] = {
+      103,100, 92, 85, 81,
+       77, 72, 70, 78, 75,
+       73, 71, 78, 74, 69,
+       72, 70, 74, 76, 71,
+       60, 60, 60, 60, 60
+};
+#else
+/* Mean energy in each band quantized in Q6 and converted back to float */
+static const opus_val16 eMeans[25] = {
+      6.437500f, 6.250000f, 5.750000f, 5.312500f, 5.062500f,
+      4.812500f, 4.500000f, 4.375000f, 4.875000f, 4.687500f,
+      4.562500f, 4.437500f, 4.875000f, 4.625000f, 4.312500f,
+      4.500000f, 4.375000f, 4.625000f, 4.750000f, 4.437500f,
+      3.750000f, 3.750000f, 3.750000f, 3.750000f, 3.750000f
+};
+#endif
+/* prediction coefficients: 0.9, 0.8, 0.65, 0.5 */
+#ifdef FIXED_POINT
+static const opus_val16 pred_coef[4] = {29440, 26112, 21248, 16384};
+static const opus_val16 beta_coef[4] = {30147, 22282, 12124, 6554};
+static const opus_val16 beta_intra = 4915;
+#else
+static const opus_val16 pred_coef[4] = {29440/32768., 26112/32768., 21248/32768., 16384/32768.};
+static const opus_val16 beta_coef[4] = {30147/32768., 22282/32768., 12124/32768., 6554/32768.};
+static const opus_val16 beta_intra = 4915/32768.;
+#endif
+
+/*Parameters of the Laplace-like probability models used for the coarse energy.
+  There is one pair of parameters for each frame size, prediction type
+   (inter/intra), and band number.
+  The first number of each pair is the probability of 0, and the second is the
+   decay rate, both in Q8 precision.*/
+static const unsigned char e_prob_model[4][2][42] = {
+   /*120 sample frames.*/
+   {
+      /*Inter*/
+      {
+          72, 127,  65, 129,  66, 128,  65, 128,  64, 128,  62, 128,  64, 128,
+          64, 128,  92,  78,  92,  79,  92,  78,  90,  79, 116,  41, 115,  40,
+         114,  40, 132,  26, 132,  26, 145,  17, 161,  12, 176,  10, 177,  11
+      },
+      /*Intra*/
+      {
+          24, 179,  48, 138,  54, 135,  54, 132,  53, 134,  56, 133,  55, 132,
+          55, 132,  61, 114,  70,  96,  74,  88,  75,  88,  87,  74,  89,  66,
+          91,  67, 100,  59, 108,  50, 120,  40, 122,  37,  97,  43,  78,  50
+      }
+   },
+   /*240 sample frames.*/
+   {
+      /*Inter*/
+      {
+          83,  78,  84,  81,  88,  75,  86,  74,  87,  71,  90,  73,  93,  74,
+          93,  74, 109,  40, 114,  36, 117,  34, 117,  34, 143,  17, 145,  18,
+         146,  19, 162,  12, 165,  10, 178,   7, 189,   6, 190,   8, 177,   9
+      },
+      /*Intra*/
+      {
+          23, 178,  54, 115,  63, 102,  66,  98,  69,  99,  74,  89,  71,  91,
+          73,  91,  78,  89,  86,  80,  92,  66,  93,  64, 102,  59, 103,  60,
+         104,  60, 117,  52, 123,  44, 138,  35, 133,  31,  97,  38,  77,  45
+      }
+   },
+   /*480 sample frames.*/
+   {
+      /*Inter*/
+      {
+          61,  90,  93,  60, 105,  42, 107,  41, 110,  45, 116,  38, 113,  38,
+         112,  38, 124,  26, 132,  27, 136,  19, 140,  20, 155,  14, 159,  16,
+         158,  18, 170,  13, 177,  10, 187,   8, 192,   6, 175,   9, 159,  10
+      },
+      /*Intra*/
+      {
+          21, 178,  59, 110,  71,  86,  75,  85,  84,  83,  91,  66,  88,  73,
+          87,  72,  92,  75,  98,  72, 105,  58, 107,  54, 115,  52, 114,  55,
+         112,  56, 129,  51, 132,  40, 150,  33, 140,  29,  98,  35,  77,  42
+      }
+   },
+   /*960 sample frames.*/
+   {
+      /*Inter*/
+      {
+          42, 121,  96,  66, 108,  43, 111,  40, 117,  44, 123,  32, 120,  36,
+         119,  33, 127,  33, 134,  34, 139,  21, 147,  23, 152,  20, 158,  25,
+         154,  26, 166,  21, 173,  16, 184,  13, 184,  10, 150,  13, 139,  15
+      },
+      /*Intra*/
+      {
+          22, 178,  63, 114,  74,  82,  84,  83,  92,  82, 103,  62,  96,  72,
+          96,  67, 101,  73, 107,  72, 113,  55, 118,  52, 125,  52, 118,  52,
+         117,  55, 135,  49, 137,  39, 157,  32, 145,  29,  97,  33,  77,  40
+      }
+   }
+};
+
+static const unsigned char small_energy_icdf[3]={2,1,0};
+
+static opus_val32 loss_distortion(const opus_val16 *eBands, opus_val16 *oldEBands, int start, int end, int len, int C)
+{
+   int c, i;
+   opus_val32 dist = 0;
+   c=0; do {
+      for (i=start;i<end;i++)
+      {
+         opus_val16 d = SUB16(SHR16(eBands[i+c*len], 3), SHR16(oldEBands[i+c*len], 3));
+         dist = MAC16_16(dist, d,d);
+      }
+   } while (++c<C);
+   return MIN32(200,SHR32(dist,2*DB_SHIFT-6));
+}
+
+static int quant_coarse_energy_impl(const CELTMode *m, int start, int end,
+      const opus_val16 *eBands, opus_val16 *oldEBands,
+      opus_int32 budget, opus_int32 tell,
+      const unsigned char *prob_model, opus_val16 *error, ec_enc *enc,
+      int C, int LM, int intra, opus_val16 max_decay)
+{
+   int i, c;
+   int badness = 0;
+   opus_val32 prev[2] = {0,0};
+   opus_val16 coef;
+   opus_val16 beta;
+
+   if (tell+3 <= budget)
+      ec_enc_bit_logp(enc, intra, 3);
+   if (intra)
+   {
+      coef = 0;
+      beta = beta_intra;
+   } else {
+      beta = beta_coef[LM];
+      coef = pred_coef[LM];
+   }
+
+   /* Encode at a fixed coarse resolution */
+   for (i=start;i<end;i++)
+   {
+      c=0;
+      do {
+         int bits_left;
+         int qi, qi0;
+         opus_val32 q;
+         opus_val16 x;
+         opus_val32 f, tmp;
+         opus_val16 oldE;
+         opus_val16 decay_bound;
+         x = eBands[i+c*m->nbEBands];
+         oldE = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]);
+#ifdef FIXED_POINT
+         f = SHL32(EXTEND32(x),7) - PSHR32(MULT16_16(coef,oldE), 8) - prev[c];
+         /* Rounding to nearest integer here is really important! */
+         qi = (f+QCONST32(.5f,DB_SHIFT+7))>>(DB_SHIFT+7);
+         decay_bound = EXTRACT16(MAX32(-QCONST16(28.f,DB_SHIFT),
+               SUB32((opus_val32)oldEBands[i+c*m->nbEBands],max_decay)));
+#else
+         f = x-coef*oldE-prev[c];
+         /* Rounding to nearest integer here is really important! */
+         qi = (int)floor(.5f+f);
+         decay_bound = MAX16(-QCONST16(28.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]) - max_decay;
+#endif
+         /* Prevent the energy from going down too quickly (e.g. for bands
+            that have just one bin) */
+         if (qi < 0 && x < decay_bound)
+         {
+            qi += (int)SHR16(SUB16(decay_bound,x), DB_SHIFT);
+            if (qi > 0)
+               qi = 0;
+         }
+         qi0 = qi;
+         /* If we don't have enough bits to encode all the energy, just assume
+             something safe. */
+         tell = ec_tell(enc);
+         bits_left = budget-tell-3*C*(end-i);
+         if (i!=start && bits_left < 30)
+         {
+            if (bits_left < 24)
+               qi = IMIN(1, qi);
+            if (bits_left < 16)
+               qi = IMAX(-1, qi);
+         }
+         if (budget-tell >= 15)
+         {
+            int pi;
+            pi = 2*IMIN(i,20);
+            ec_laplace_encode(enc, &qi,
+                  prob_model[pi]<<7, prob_model[pi+1]<<6);
+         }
+         else if(budget-tell >= 2)
+         {
+            qi = IMAX(-1, IMIN(qi, 1));
+            ec_enc_icdf(enc, 2*qi^-(qi<0), small_energy_icdf, 2);
+         }
+         else if(budget-tell >= 1)
+         {
+            qi = IMIN(0, qi);
+            ec_enc_bit_logp(enc, -qi, 1);
+         }
+         else
+            qi = -1;
+         error[i+c*m->nbEBands] = PSHR32(f,7) - SHL16(qi,DB_SHIFT);
+         badness += abs(qi0-qi);
+         q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT);
+
+         tmp = PSHR32(MULT16_16(coef,oldE),8) + prev[c] + SHL32(q,7);
+#ifdef FIXED_POINT
+         tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp);
+#endif
+         oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7);
+         prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8));
+      } while (++c < C);
+   }
+   return badness;
+}
+
+void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd,
+      const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget,
+      opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes,
+      int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate)
+{
+   int intra;
+   opus_val16 max_decay;
+   VARDECL(opus_val16, oldEBands_intra);
+   VARDECL(opus_val16, error_intra);
+   ec_enc enc_start_state;
+   opus_uint32 tell;
+   int badness1=0;
+   opus_int32 intra_bias;
+   opus_val32 new_distortion;
+   SAVE_STACK;
+
+   intra = force_intra || (!two_pass && *delayedIntra>2*C*(end-start) && nbAvailableBytes > (end-start)*C);
+   intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512));
+   new_distortion = loss_distortion(eBands, oldEBands, start, effEnd, m->nbEBands, C);
+
+   tell = ec_tell(enc);
+   if (tell+3 > budget)
+      two_pass = intra = 0;
+
+   /* Encode the global flags using a simple probability model
+      (first symbols in the stream) */
+
+#ifdef FIXED_POINT
+      max_decay = MIN32(QCONST16(16.f,DB_SHIFT), SHL32(EXTEND32(nbAvailableBytes),DB_SHIFT-3));
+#else
+   max_decay = MIN32(16.f, .125f*nbAvailableBytes);
+#endif
+
+   enc_start_state = *enc;
+
+   ALLOC(oldEBands_intra, C*m->nbEBands, opus_val16);
+   ALLOC(error_intra, C*m->nbEBands, opus_val16);
+   OPUS_COPY(oldEBands_intra, oldEBands, C*m->nbEBands);
+
+   if (two_pass || intra)
+   {
+      badness1 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands_intra, budget,
+            tell, e_prob_model[LM][1], error_intra, enc, C, LM, 1, max_decay);
+   }
+
+   if (!intra)
+   {
+      unsigned char *intra_buf;
+      ec_enc enc_intra_state;
+      opus_int32 tell_intra;
+      opus_uint32 nstart_bytes;
+      opus_uint32 nintra_bytes;
+      int badness2;
+      VARDECL(unsigned char, intra_bits);
+
+      tell_intra = ec_tell_frac(enc);
+
+      enc_intra_state = *enc;
+
+      nstart_bytes = ec_range_bytes(&enc_start_state);
+      nintra_bytes = ec_range_bytes(&enc_intra_state);
+      intra_buf = ec_get_buffer(&enc_intra_state) + nstart_bytes;
+      ALLOC(intra_bits, nintra_bytes-nstart_bytes, unsigned char);
+      /* Copy bits from intra bit-stream */
+      OPUS_COPY(intra_bits, intra_buf, nintra_bytes - nstart_bytes);
+
+      *enc = enc_start_state;
+
+      badness2 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands, budget,
+            tell, e_prob_model[LM][intra], error, enc, C, LM, 0, max_decay);
+
+      if (two_pass && (badness1 < badness2 || (badness1 == badness2 && ((opus_int32)ec_tell_frac(enc))+intra_bias > tell_intra)))
+      {
+         *enc = enc_intra_state;
+         /* Copy intra bits to bit-stream */
+         OPUS_COPY(intra_buf, intra_bits, nintra_bytes - nstart_bytes);
+         OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands);
+         OPUS_COPY(error, error_intra, C*m->nbEBands);
+         intra = 1;
+      }
+   } else {
+      OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands);
+      OPUS_COPY(error, error_intra, C*m->nbEBands);
+   }
+
+   if (intra)
+      *delayedIntra = new_distortion;
+   else
+      *delayedIntra = ADD32(MULT16_32_Q15(MULT16_16_Q15(pred_coef[LM], pred_coef[LM]),*delayedIntra),
+            new_distortion);
+
+   RESTORE_STACK;
+}
+
+void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C)
+{
+   int i, c;
+
+   /* Encode finer resolution */
+   for (i=start;i<end;i++)
+   {
+      opus_int16 frac = 1<<fine_quant[i];
+      if (fine_quant[i] <= 0)
+         continue;
+      c=0;
+      do {
+         int q2;
+         opus_val16 offset;
+#ifdef FIXED_POINT
+         /* Has to be without rounding */
+         q2 = (error[i+c*m->nbEBands]+QCONST16(.5f,DB_SHIFT))>>(DB_SHIFT-fine_quant[i]);
+#else
+         q2 = (int)floor((error[i+c*m->nbEBands]+.5f)*frac);
+#endif
+         if (q2 > frac-1)
+            q2 = frac-1;
+         if (q2<0)
+            q2 = 0;
+         ec_enc_bits(enc, q2, fine_quant[i]);
+#ifdef FIXED_POINT
+         offset = SUB16(SHR32(SHL32(EXTEND32(q2),DB_SHIFT)+QCONST16(.5f,DB_SHIFT),fine_quant[i]),QCONST16(.5f,DB_SHIFT));
+#else
+         offset = (q2+.5f)*(1<<(14-fine_quant[i]))*(1.f/16384) - .5f;
+#endif
+         oldEBands[i+c*m->nbEBands] += offset;
+         error[i+c*m->nbEBands] -= offset;
+         /*printf ("%f ", error[i] - offset);*/
+      } while (++c < C);
+   }
+}
+
+void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C)
+{
+   int i, prio, c;
+
+   /* Use up the remaining bits */
+   for (prio=0;prio<2;prio++)
+   {
+      for (i=start;i<end && bits_left>=C ;i++)
+      {
+         if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio)
+            continue;
+         c=0;
+         do {
+            int q2;
+            opus_val16 offset;
+            q2 = error[i+c*m->nbEBands]<0 ? 0 : 1;
+            ec_enc_bits(enc, q2, 1);
+#ifdef FIXED_POINT
+            offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1);
+#else
+            offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384);
+#endif
+            oldEBands[i+c*m->nbEBands] += offset;
+            bits_left--;
+         } while (++c < C);
+      }
+   }
+}
+
+void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM)
+{
+   const unsigned char *prob_model = e_prob_model[LM][intra];
+   int i, c;
+   opus_val32 prev[2] = {0, 0};
+   opus_val16 coef;
+   opus_val16 beta;
+   opus_int32 budget;
+   opus_int32 tell;
+
+   if (intra)
+   {
+      coef = 0;
+      beta = beta_intra;
+   } else {
+      beta = beta_coef[LM];
+      coef = pred_coef[LM];
+   }
+
+   budget = dec->storage*8;
+
+   /* Decode at a fixed coarse resolution */
+   for (i=start;i<end;i++)
+   {
+      c=0;
+      do {
+         int qi;
+         opus_val32 q;
+         opus_val32 tmp;
+         /* It would be better to express this invariant as a
+            test on C at function entry, but that isn't enough
+            to make the static analyzer happy. */
+         celt_assert(c<2);
+         tell = ec_tell(dec);
+         if(budget-tell>=15)
+         {
+            int pi;
+            pi = 2*IMIN(i,20);
+            qi = ec_laplace_decode(dec,
+                  prob_model[pi]<<7, prob_model[pi+1]<<6);
+         }
+         else if(budget-tell>=2)
+         {
+            qi = ec_dec_icdf(dec, small_energy_icdf, 2);
+            qi = (qi>>1)^-(qi&1);
+         }
+         else if(budget-tell>=1)
+         {
+            qi = -ec_dec_bit_logp(dec, 1);
+         }
+         else
+            qi = -1;
+         q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT);
+
+         oldEBands[i+c*m->nbEBands] = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]);
+         tmp = PSHR32(MULT16_16(coef,oldEBands[i+c*m->nbEBands]),8) + prev[c] + SHL32(q,7);
+#ifdef FIXED_POINT
+         tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp);
+#endif
+         oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7);
+         prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8));
+      } while (++c < C);
+   }
+}
+
+void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C)
+{
+   int i, c;
+   /* Decode finer resolution */
+   for (i=start;i<end;i++)
+   {
+      if (fine_quant[i] <= 0)
+         continue;
+      c=0;
+      do {
+         int q2;
+         opus_val16 offset;
+         q2 = ec_dec_bits(dec, fine_quant[i]);
+#ifdef FIXED_POINT
+         offset = SUB16(SHR32(SHL32(EXTEND32(q2),DB_SHIFT)+QCONST16(.5f,DB_SHIFT),fine_quant[i]),QCONST16(.5f,DB_SHIFT));
+#else
+         offset = (q2+.5f)*(1<<(14-fine_quant[i]))*(1.f/16384) - .5f;
+#endif
+         oldEBands[i+c*m->nbEBands] += offset;
+      } while (++c < C);
+   }
+}
+
+void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant,  int *fine_priority, int bits_left, ec_dec *dec, int C)
+{
+   int i, prio, c;
+
+   /* Use up the remaining bits */
+   for (prio=0;prio<2;prio++)
+   {
+      for (i=start;i<end && bits_left>=C ;i++)
+      {
+         if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio)
+            continue;
+         c=0;
+         do {
+            int q2;
+            opus_val16 offset;
+            q2 = ec_dec_bits(dec, 1);
+#ifdef FIXED_POINT
+            offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1);
+#else
+            offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384);
+#endif
+            oldEBands[i+c*m->nbEBands] += offset;
+            bits_left--;
+         } while (++c < C);
+      }
+   }
+}
+
+void log2Amp(const CELTMode *m, int start, int end,
+      celt_ener *eBands, const opus_val16 *oldEBands, int C)
+{
+   int c, i;
+   c=0;
+   do {
+      for (i=0;i<start;i++)
+         eBands[i+c*m->nbEBands] = 0;
+      for (;i<end;i++)
+      {
+         opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands],
+                         SHL16((opus_val16)eMeans[i],6));
+         eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4);
+      }
+      for (;i<m->nbEBands;i++)
+         eBands[i+c*m->nbEBands] = 0;
+   } while (++c < C);
+}
+
+void amp2Log2(const CELTMode *m, int effEnd, int end,
+      celt_ener *bandE, opus_val16 *bandLogE, int C)
+{
+   int c, i;
+   c=0;
+   do {
+      for (i=0;i<effEnd;i++)
+         bandLogE[i+c*m->nbEBands] =
+               celt_log2(SHL32(bandE[i+c*m->nbEBands],2))
+               - SHL16((opus_val16)eMeans[i],6);
+      for (i=effEnd;i<end;i++)
+         bandLogE[c*m->nbEBands+i] = -QCONST16(14.f,DB_SHIFT);
+   } while (++c < C);
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/quant_bands.h b/lib/rbcodec/codecs/libopus/celt/quant_bands.h
new file mode 100644
index 0000000000..bec2855cf0
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/quant_bands.h
@@ -0,0 +1,60 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef QUANT_BANDS
+#define QUANT_BANDS
+
+#include "arch.h"
+#include "modes.h"
+#include "entenc.h"
+#include "entdec.h"
+#include "mathops.h"
+
+void amp2Log2(const CELTMode *m, int effEnd, int end,
+      celt_ener *bandE, opus_val16 *bandLogE, int C);
+
+void log2Amp(const CELTMode *m, int start, int end,
+      celt_ener *eBands, const opus_val16 *oldEBands, int C);
+
+void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd,
+      const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget,
+      opus_val16 *error, ec_enc *enc, int C, int LM,
+      int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra,
+      int two_pass, int loss_rate);
+
+void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C);
+
+void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C);
+
+void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM);
+
+void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C);
+
+void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, int *fine_priority, int bits_left, ec_dec *dec, int C);
+
+#endif /* QUANT_BANDS */
diff --git a/lib/rbcodec/codecs/libopus/celt/rate.c b/lib/rbcodec/codecs/libopus/celt/rate.c
new file mode 100644
index 0000000000..3b056d8dc7
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/rate.c
@@ -0,0 +1,638 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include <math.h>
+#include "modes.h"
+#include "cwrs.h"
+#include "arch.h"
+#include "os_support.h"
+
+#include "entcode.h"
+#include "rate.h"
+
+static const unsigned char LOG2_FRAC_TABLE[24]={
+   0,
+   8,13,
+  16,19,21,23,
+  24,26,27,28,29,30,31,32,
+  32,33,34,34,35,36,36,37,37
+};
+
+#ifdef CUSTOM_MODES
+
+/*Determines if V(N,K) fits in a 32-bit unsigned integer.
+  N and K are themselves limited to 15 bits.*/
+static int fits_in32(int _n, int _k)
+{
+   static const opus_int16 maxN[15] = {
+      32767, 32767, 32767, 1476, 283, 109,  60,  40,
+       29,  24,  20,  18,  16,  14,  13};
+   static const opus_int16 maxK[15] = {
+      32767, 32767, 32767, 32767, 1172, 238,  95,  53,
+       36,  27,  22,  18,  16,  15,  13};
+   if (_n>=14)
+   {
+      if (_k>=14)
+         return 0;
+      else
+         return _n <= maxN[_k];
+   } else {
+      return _k <= maxK[_n];
+   }
+}
+
+void compute_pulse_cache(CELTMode *m, int LM)
+{
+   int C;
+   int i;
+   int j;
+   int curr=0;
+   int nbEntries=0;
+   int entryN[100], entryK[100], entryI[100];
+   const opus_int16 *eBands = m->eBands;
+   PulseCache *cache = &m->cache;
+   opus_int16 *cindex;
+   unsigned char *bits;
+   unsigned char *cap;
+
+   cindex = (opus_int16 *)opus_alloc(sizeof(cache->index[0])*m->nbEBands*(LM+2));
+   cache->index = cindex;
+
+   /* Scan for all unique band sizes */
+   for (i=0;i<=LM+1;i++)
+   {
+      for (j=0;j<m->nbEBands;j++)
+      {
+         int k;
+         int N = (eBands[j+1]-eBands[j])<<i>>1;
+         cindex[i*m->nbEBands+j] = -1;
+         /* Find other bands that have the same size */
+         for (k=0;k<=i;k++)
+         {
+            int n;
+            for (n=0;n<m->nbEBands && (k!=i || n<j);n++)
+            {
+               if (N == (eBands[n+1]-eBands[n])<<k>>1)
+               {
+                  cindex[i*m->nbEBands+j] = cindex[k*m->nbEBands+n];
+                  break;
+               }
+            }
+         }
+         if (cache->index[i*m->nbEBands+j] == -1 && N!=0)
+         {
+            int K;
+            entryN[nbEntries] = N;
+            K = 0;
+            while (fits_in32(N,get_pulses(K+1)) && K<MAX_PSEUDO)
+               K++;
+            entryK[nbEntries] = K;
+            cindex[i*m->nbEBands+j] = curr;
+            entryI[nbEntries] = curr;
+
+            curr += K+1;
+            nbEntries++;
+         }
+      }
+   }
+   bits = (unsigned char *)opus_alloc(sizeof(unsigned char)*curr);
+   cache->bits = bits;
+   cache->size = curr;
+   /* Compute the cache for all unique sizes */
+   for (i=0;i<nbEntries;i++)
+   {
+      unsigned char *ptr = bits+entryI[i];
+      opus_int16 tmp[MAX_PULSES+1];
+      get_required_bits(tmp, entryN[i], get_pulses(entryK[i]), BITRES);
+      for (j=1;j<=entryK[i];j++)
+         ptr[j] = tmp[get_pulses(j)]-1;
+      ptr[0] = entryK[i];
+   }
+
+   /* Compute the maximum rate for each band at which we'll reliably use as
+       many bits as we ask for. */
+   cache->caps = cap = (unsigned char *)opus_alloc(sizeof(cache->caps[0])*(LM+1)*2*m->nbEBands);
+   for (i=0;i<=LM;i++)
+   {
+      for (C=1;C<=2;C++)
+      {
+         for (j=0;j<m->nbEBands;j++)
+         {
+            int N0;
+            int max_bits;
+            N0 = m->eBands[j+1]-m->eBands[j];
+            /* N=1 bands only have a sign bit and fine bits. */
+            if (N0<<i == 1)
+               max_bits = C*(1+MAX_FINE_BITS)<<BITRES;
+            else
+            {
+               const unsigned char *pcache;
+               opus_int32           num;
+               opus_int32           den;
+               int                  LM0;
+               int                  N;
+               int                  offset;
+               int                  ndof;
+               int                  qb;
+               int                  k;
+               LM0 = 0;
+               /* Even-sized bands bigger than N=2 can be split one more time.
+                  As of commit 44203907 all bands >1 are even, including custom modes.*/
+               if (N0 > 2)
+               {
+                  N0>>=1;
+                  LM0--;
+               }
+               /* N0=1 bands can't be split down to N<2. */
+               else if (N0 <= 1)
+               {
+                  LM0=IMIN(i,1);
+                  N0<<=LM0;
+               }
+               /* Compute the cost for the lowest-level PVQ of a fully split
+                   band. */
+               pcache = bits + cindex[(LM0+1)*m->nbEBands+j];
+               max_bits = pcache[pcache[0]]+1;
+               /* Add in the cost of coding regular splits. */
+               N = N0;
+               for(k=0;k<i-LM0;k++){
+                  max_bits <<= 1;
+                  /* Offset the number of qtheta bits by log2(N)/2
+                      + QTHETA_OFFSET compared to their "fair share" of
+                      total/N */
+                  offset = ((m->logN[j]+((LM0+k)<<BITRES))>>1)-QTHETA_OFFSET;
+                  /* The number of qtheta bits we'll allocate if the remainder
+                      is to be max_bits.
+                     The average measured cost for theta is 0.89701 times qb,
+                      approximated here as 459/512. */
+                  num=459*(opus_int32)((2*N-1)*offset+max_bits);
+                  den=((opus_int32)(2*N-1)<<9)-459;
+                  qb = IMIN((num+(den>>1))/den, 57);
+                  celt_assert(qb >= 0);
+                  max_bits += qb;
+                  N <<= 1;
+               }
+               /* Add in the cost of a stereo split, if necessary. */
+               if (C==2)
+               {
+                  max_bits <<= 1;
+                  offset = ((m->logN[j]+(i<<BITRES))>>1)-(N==2?QTHETA_OFFSET_TWOPHASE:QTHETA_OFFSET);
+                  ndof = 2*N-1-(N==2);
+                  /* The average measured cost for theta with the step PDF is
+                      0.95164 times qb, approximated here as 487/512. */
+                  num = (N==2?512:487)*(opus_int32)(max_bits+ndof*offset);
+                  den = ((opus_int32)ndof<<9)-(N==2?512:487);
+                  qb = IMIN((num+(den>>1))/den, (N==2?64:61));
+                  celt_assert(qb >= 0);
+                  max_bits += qb;
+               }
+               /* Add the fine bits we'll use. */
+               /* Compensate for the extra DoF in stereo */
+               ndof = C*N + ((C==2 && N>2) ? 1 : 0);
+               /* Offset the number of fine bits by log2(N)/2 + FINE_OFFSET
+                   compared to their "fair share" of total/N */
+               offset = ((m->logN[j] + (i<<BITRES))>>1)-FINE_OFFSET;
+               /* N=2 is the only point that doesn't match the curve */
+               if (N==2)
+                  offset += 1<<BITRES>>2;
+               /* The number of fine bits we'll allocate if the remainder is
+                   to be max_bits. */
+               num = max_bits+ndof*offset;
+               den = (ndof-1)<<BITRES;
+               qb = IMIN((num+(den>>1))/den, MAX_FINE_BITS);
+               celt_assert(qb >= 0);
+               max_bits += C*qb<<BITRES;
+            }
+            max_bits = (4*max_bits/(C*((m->eBands[j+1]-m->eBands[j])<<i)))-64;
+            celt_assert(max_bits >= 0);
+            celt_assert(max_bits < 256);
+            *cap++ = (unsigned char)max_bits;
+         }
+      }
+   }
+}
+
+#endif /* CUSTOM_MODES */
+
+#define ALLOC_STEPS 6
+
+static inline int interp_bits2pulses(const CELTMode *m, int start, int end, int skip_start,
+      const int *bits1, const int *bits2, const int *thresh, const int *cap, opus_int32 total, opus_int32 *_balance,
+      int skip_rsv, int *intensity, int intensity_rsv, int *dual_stereo, int dual_stereo_rsv, int *bits,
+      int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev)
+{
+   opus_int32 psum;
+   int lo, hi;
+   int i, j;
+   int logM;
+   int stereo;
+   int codedBands=-1;
+   int alloc_floor;
+   opus_int32 left, percoeff;
+   int done;
+   opus_int32 balance;
+   SAVE_STACK;
+
+   alloc_floor = C<<BITRES;
+   stereo = C>1;
+
+   logM = LM<<BITRES;
+   lo = 0;
+   hi = 1<<ALLOC_STEPS;
+   for (i=0;i<ALLOC_STEPS;i++)
+   {
+      int mid = (lo+hi)>>1;
+      psum = 0;
+      done = 0;
+      for (j=end;j-->start;)
+      {
+         int tmp = bits1[j] + (mid*(opus_int32)bits2[j]>>ALLOC_STEPS);
+         if (tmp >= thresh[j] || done)
+         {
+            done = 1;
+            /* Don't allocate more than we can actually use */
+            psum += IMIN(tmp, cap[j]);
+         } else {
+            if (tmp >= alloc_floor)
+               psum += alloc_floor;
+         }
+      }
+      if (psum > total)
+         hi = mid;
+      else
+         lo = mid;
+   }
+   psum = 0;
+   /*printf ("interp bisection gave %d\n", lo);*/
+   done = 0;
+   for (j=end;j-->start;)
+   {
+      int tmp = bits1[j] + (lo*bits2[j]>>ALLOC_STEPS);
+      if (tmp < thresh[j] && !done)
+      {
+         if (tmp >= alloc_floor)
+            tmp = alloc_floor;
+         else
+            tmp = 0;
+      } else
+         done = 1;
+      /* Don't allocate more than we can actually use */
+      tmp = IMIN(tmp, cap[j]);
+      bits[j] = tmp;
+      psum += tmp;
+   }
+
+   /* Decide which bands to skip, working backwards from the end. */
+   for (codedBands=end;;codedBands--)
+   {
+      int band_width;
+      int band_bits;
+      int rem;
+      j = codedBands-1;
+      /* Never skip the first band, nor a band that has been boosted by
+          dynalloc.
+         In the first case, we'd be coding a bit to signal we're going to waste
+          all the other bits.
+         In the second case, we'd be coding a bit to redistribute all the bits
+          we just signaled should be cocentrated in this band. */
+      if (j<=skip_start)
+      {
+         /* Give the bit we reserved to end skipping back. */
+         total += skip_rsv;
+         break;
+      }
+      /*Figure out how many left-over bits we would be adding to this band.
+        This can include bits we've stolen back from higher, skipped bands.*/
+      left = total-psum;
+      percoeff = left/(m->eBands[codedBands]-m->eBands[start]);
+      left -= (m->eBands[codedBands]-m->eBands[start])*percoeff;
+      rem = IMAX(left-(m->eBands[j]-m->eBands[start]),0);
+      band_width = m->eBands[codedBands]-m->eBands[j];
+      band_bits = (int)(bits[j] + percoeff*band_width + rem);
+      /*Only code a skip decision if we're above the threshold for this band.
+        Otherwise it is force-skipped.
+        This ensures that we have enough bits to code the skip flag.*/
+      if (band_bits >= IMAX(thresh[j], alloc_floor+(1<<BITRES)))
+      {
+         if (encode)
+         {
+            /*This if() block is the only part of the allocation function that
+               is not a mandatory part of the bitstream: any bands we choose to
+               skip here must be explicitly signaled.*/
+            /*Choose a threshold with some hysteresis to keep bands from
+               fluctuating in and out.*/
+#ifdef FUZZING
+            if ((rand()&0x1) == 0)
+#else
+            if (band_bits > ((j<prev?7:9)*band_width<<LM<<BITRES)>>4)
+#endif
+            {
+               ec_enc_bit_logp(ec, 1, 1);
+               break;
+            }
+            ec_enc_bit_logp(ec, 0, 1);
+         } else if (ec_dec_bit_logp(ec, 1)) {
+            break;
+         }
+         /*We used a bit to skip this band.*/
+         psum += 1<<BITRES;
+         band_bits -= 1<<BITRES;
+      }
+      /*Reclaim the bits originally allocated to this band.*/
+      psum -= bits[j]+intensity_rsv;
+      if (intensity_rsv > 0)
+         intensity_rsv = LOG2_FRAC_TABLE[j-start];
+      psum += intensity_rsv;
+      if (band_bits >= alloc_floor)
+      {
+         /*If we have enough for a fine energy bit per channel, use it.*/
+         psum += alloc_floor;
+         bits[j] = alloc_floor;
+      } else {
+         /*Otherwise this band gets nothing at all.*/
+         bits[j] = 0;
+      }
+   }
+
+   celt_assert(codedBands > start);
+   /* Code the intensity and dual stereo parameters. */
+   if (intensity_rsv > 0)
+   {
+      if (encode)
+      {
+         *intensity = IMIN(*intensity, codedBands);
+         ec_enc_uint(ec, *intensity-start, codedBands+1-start);
+      }
+      else
+         *intensity = start+ec_dec_uint(ec, codedBands+1-start);
+   }
+   else
+      *intensity = 0;
+   if (*intensity <= start)
+   {
+      total += dual_stereo_rsv;
+      dual_stereo_rsv = 0;
+   }
+   if (dual_stereo_rsv > 0)
+   {
+      if (encode)
+         ec_enc_bit_logp(ec, *dual_stereo, 1);
+      else
+         *dual_stereo = ec_dec_bit_logp(ec, 1);
+   }
+   else
+      *dual_stereo = 0;
+
+   /* Allocate the remaining bits */
+   left = total-psum;
+   percoeff = left/(m->eBands[codedBands]-m->eBands[start]);
+   left -= (m->eBands[codedBands]-m->eBands[start])*percoeff;
+   for (j=start;j<codedBands;j++)
+      bits[j] += ((int)percoeff*(m->eBands[j+1]-m->eBands[j]));
+   for (j=start;j<codedBands;j++)
+   {
+      int tmp = (int)IMIN(left, m->eBands[j+1]-m->eBands[j]);
+      bits[j] += tmp;
+      left -= tmp;
+   }
+   /*for (j=0;j<end;j++)printf("%d ", bits[j]);printf("\n");*/
+
+   balance = 0;
+   for (j=start;j<codedBands;j++)
+   {
+      int N0, N, den;
+      int offset;
+      int NClogN;
+      opus_int32 excess, bit;
+
+      celt_assert(bits[j] >= 0);
+      N0 = m->eBands[j+1]-m->eBands[j];
+      N=N0<<LM;
+      bit = (opus_int32)bits[j]+balance;
+
+      if (N>1)
+      {
+         excess = MAX32(bit-cap[j],0);
+         bits[j] = bit-excess;
+
+         /* Compensate for the extra DoF in stereo */
+         den=(C*N+ ((C==2 && N>2 && !*dual_stereo && j<*intensity) ? 1 : 0));
+
+         NClogN = den*(m->logN[j] + logM);
+
+         /* Offset for the number of fine bits by log2(N)/2 + FINE_OFFSET
+            compared to their "fair share" of total/N */
+         offset = (NClogN>>1)-den*FINE_OFFSET;
+
+         /* N=2 is the only point that doesn't match the curve */
+         if (N==2)
+            offset += den<<BITRES>>2;
+
+         /* Changing the offset for allocating the second and third
+             fine energy bit */
+         if (bits[j] + offset < den*2<<BITRES)
+            offset += NClogN>>2;
+         else if (bits[j] + offset < den*3<<BITRES)
+            offset += NClogN>>3;
+
+         /* Divide with rounding */
+         ebits[j] = IMAX(0, (bits[j] + offset + (den<<(BITRES-1))) / (den<<BITRES));
+
+         /* Make sure not to bust */
+         if (C*ebits[j] > (bits[j]>>BITRES))
+            ebits[j] = bits[j] >> stereo >> BITRES;
+
+         /* More than that is useless because that's about as far as PVQ can go */
+         ebits[j] = IMIN(ebits[j], MAX_FINE_BITS);
+
+         /* If we rounded down or capped this band, make it a candidate for the
+             final fine energy pass */
+         fine_priority[j] = ebits[j]*(den<<BITRES) >= bits[j]+offset;
+
+         /* Remove the allocated fine bits; the rest are assigned to PVQ */
+         bits[j] -= C*ebits[j]<<BITRES;
+
+      } else {
+         /* For N=1, all bits go to fine energy except for a single sign bit */
+         excess = MAX32(0,bit-(C<<BITRES));
+         bits[j] = bit-excess;
+         ebits[j] = 0;
+         fine_priority[j] = 1;
+      }
+
+      /* Fine energy can't take advantage of the re-balancing in
+          quant_all_bands().
+         Instead, do the re-balancing here.*/
+      if(excess > 0)
+      {
+         int extra_fine;
+         int extra_bits;
+         extra_fine = IMIN(excess>>(stereo+BITRES),MAX_FINE_BITS-ebits[j]);
+         ebits[j] += extra_fine;
+         extra_bits = extra_fine*C<<BITRES;
+         fine_priority[j] = extra_bits >= excess-balance;
+         excess -= extra_bits;
+      }
+      balance = excess;
+
+      celt_assert(bits[j] >= 0);
+      celt_assert(ebits[j] >= 0);
+   }
+   /* Save any remaining bits over the cap for the rebalancing in
+       quant_all_bands(). */
+   *_balance = balance;
+
+   /* The skipped bands use all their bits for fine energy. */
+   for (;j<end;j++)
+   {
+      ebits[j] = bits[j] >> stereo >> BITRES;
+      celt_assert(C*ebits[j]<<BITRES == bits[j]);
+      bits[j] = 0;
+      fine_priority[j] = ebits[j]<1;
+   }
+   RESTORE_STACK;
+   return codedBands;
+}
+
+int compute_allocation(const CELTMode *m, int start, int end, const int *offsets, const int *cap, int alloc_trim, int *intensity, int *dual_stereo,
+      opus_int32 total, opus_int32 *balance, int *pulses, int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev)
+{
+   int lo, hi, len, j;
+   int codedBands;
+   int skip_start;
+   int skip_rsv;
+   int intensity_rsv;
+   int dual_stereo_rsv;
+   VARDECL(int, bits1);
+   VARDECL(int, bits2);
+   VARDECL(int, thresh);
+   VARDECL(int, trim_offset);
+   SAVE_STACK;
+
+   total = IMAX(total, 0);
+   len = m->nbEBands;
+   skip_start = start;
+   /* Reserve a bit to signal the end of manually skipped bands. */
+   skip_rsv = total >= 1<<BITRES ? 1<<BITRES : 0;
+   total -= skip_rsv;
+   /* Reserve bits for the intensity and dual stereo parameters. */
+   intensity_rsv = dual_stereo_rsv = 0;
+   if (C==2)
+   {
+      intensity_rsv = LOG2_FRAC_TABLE[end-start];
+      if (intensity_rsv>total)
+         intensity_rsv = 0;
+      else
+      {
+         total -= intensity_rsv;
+         dual_stereo_rsv = total>=1<<BITRES ? 1<<BITRES : 0;
+         total -= dual_stereo_rsv;
+      }
+   }
+   ALLOC(bits1, len, int);
+   ALLOC(bits2, len, int);
+   ALLOC(thresh, len, int);
+   ALLOC(trim_offset, len, int);
+
+   for (j=start;j<end;j++)
+   {
+      /* Below this threshold, we're sure not to allocate any PVQ bits */
+      thresh[j] = IMAX((C)<<BITRES, (3*(m->eBands[j+1]-m->eBands[j])<<LM<<BITRES)>>4);
+      /* Tilt of the allocation curve */
+      trim_offset[j] = C*(m->eBands[j+1]-m->eBands[j])*(alloc_trim-5-LM)*(end-j-1)
+            *(1<<(LM+BITRES))>>6;
+      /* Giving less resolution to single-coefficient bands because they get
+         more benefit from having one coarse value per coefficient*/
+      if ((m->eBands[j+1]-m->eBands[j])<<LM==1)
+         trim_offset[j] -= C<<BITRES;
+   }
+   lo = 1;
+   hi = m->nbAllocVectors - 1;
+   do
+   {
+      int done = 0;
+      int psum = 0;
+      int mid = (lo+hi) >> 1;
+      for (j=end;j-->start;)
+      {
+         int bitsj;
+         int N = m->eBands[j+1]-m->eBands[j];
+         bitsj = C*N*m->allocVectors[mid*len+j]<<LM>>2;
+         if (bitsj > 0)
+            bitsj = IMAX(0, bitsj + trim_offset[j]);
+         bitsj += offsets[j];
+         if (bitsj >= thresh[j] || done)
+         {
+            done = 1;
+            /* Don't allocate more than we can actually use */
+            psum += IMIN(bitsj, cap[j]);
+         } else {
+            if (bitsj >= C<<BITRES)
+               psum += C<<BITRES;
+         }
+      }
+      if (psum > total)
+         hi = mid - 1;
+      else
+         lo = mid + 1;
+      /*printf ("lo = %d, hi = %d\n", lo, hi);*/
+   }
+   while (lo <= hi);
+   hi = lo--;
+   /*printf ("interp between %d and %d\n", lo, hi);*/
+   for (j=start;j<end;j++)
+   {
+      int bits1j, bits2j;
+      int N = m->eBands[j+1]-m->eBands[j];
+      bits1j = C*N*m->allocVectors[lo*len+j]<<LM>>2;
+      bits2j = hi>=m->nbAllocVectors ?
+            cap[j] : C*N*m->allocVectors[hi*len+j]<<LM>>2;
+      if (bits1j > 0)
+         bits1j = IMAX(0, bits1j + trim_offset[j]);
+      if (bits2j > 0)
+         bits2j = IMAX(0, bits2j + trim_offset[j]);
+      if (lo > 0)
+         bits1j += offsets[j];
+      bits2j += offsets[j];
+      if (offsets[j]>0)
+         skip_start = j;
+      bits2j = IMAX(0,bits2j-bits1j);
+      bits1[j] = bits1j;
+      bits2[j] = bits2j;
+   }
+   codedBands = interp_bits2pulses(m, start, end, skip_start, bits1, bits2, thresh, cap,
+         total, balance, skip_rsv, intensity, intensity_rsv, dual_stereo, dual_stereo_rsv,
+         pulses, ebits, fine_priority, C, LM, ec, encode, prev);
+   RESTORE_STACK;
+   return codedBands;
+}
+
diff --git a/lib/rbcodec/codecs/libopus/celt/rate.h b/lib/rbcodec/codecs/libopus/celt/rate.h
new file mode 100644
index 0000000000..e0d5022326
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/rate.h
@@ -0,0 +1,101 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef RATE_H
+#define RATE_H
+
+#define MAX_PSEUDO 40
+#define LOG_MAX_PSEUDO 6
+
+#define MAX_PULSES 128
+
+#define MAX_FINE_BITS 8
+
+#define FINE_OFFSET 21
+#define QTHETA_OFFSET 4
+#define QTHETA_OFFSET_TWOPHASE 16
+
+#include "cwrs.h"
+#include "modes.h"
+
+void compute_pulse_cache(CELTMode *m, int LM);
+
+static inline int get_pulses(int i)
+{
+   return i<8 ? i : (8 + (i&7)) << ((i>>3)-1);
+}
+
+static inline int bits2pulses(const CELTMode *m, int band, int LM, int bits)
+{
+   int i;
+   int lo, hi;
+   const unsigned char *cache;
+
+   LM++;
+   cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band];
+
+   lo = 0;
+   hi = cache[0];
+   bits--;
+   for (i=0;i<LOG_MAX_PSEUDO;i++)
+   {
+      int mid = (lo+hi+1)>>1;
+      /* OPT: Make sure this is implemented with a conditional move */
+      if ((int)cache[mid] >= bits)
+         hi = mid;
+      else
+         lo = mid;
+   }
+   if (bits- (lo == 0 ? -1 : (int)cache[lo]) <= (int)cache[hi]-bits)
+      return lo;
+   else
+      return hi;
+}
+
+static inline int pulses2bits(const CELTMode *m, int band, int LM, int pulses)
+{
+   const unsigned char *cache;
+
+   LM++;
+   cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band];
+   return pulses == 0 ? 0 : cache[pulses]+1;
+}
+
+/** Compute the pulse allocation, i.e. how many pulses will go in each
+  * band.
+ @param m mode
+ @param offsets Requested increase or decrease in the number of bits for
+                each band
+ @param total Number of bands
+ @param pulses Number of pulses per band (returned)
+ @return Total number of bits allocated
+*/
+int compute_allocation(const CELTMode *m, int start, int end, const int *offsets, const int *cap, int alloc_trim, int *intensity, int *dual_stero,
+      opus_int32 total, opus_int32 *balance, int *pulses, int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/celt/stack_alloc.h b/lib/rbcodec/codecs/libopus/celt/stack_alloc.h
new file mode 100644
index 0000000000..a6f06d2263
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/stack_alloc.h
@@ -0,0 +1,149 @@
+/* Copyright (C) 2002-2003 Jean-Marc Valin
+   Copyright (C) 2007-2009 Xiph.Org Foundation */
+/**
+   @file stack_alloc.h
+   @brief Temporary memory allocation on stack
+*/
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef STACK_ALLOC_H
+#define STACK_ALLOC_H
+
+#if (!defined (VAR_ARRAYS) && !defined (USE_ALLOCA) && !defined (NONTHREADSAFE_PSEUDOSTACK))
+#error "Opus requires one of VAR_ARRAYS, USE_ALLOCA, or NONTHREADSAFE_PSEUDOSTACK be defined to select the temporary allocation mode."
+#endif
+
+#ifdef USE_ALLOCA
+# ifdef WIN32
+#  include <malloc.h>
+# else
+#  ifdef HAVE_ALLOCA_H
+#   include <alloca.h>
+#  else
+#   include <stdlib.h>
+#  endif
+# endif
+#endif
+
+/**
+ * @def ALIGN(stack, size)
+ *
+ * Aligns the stack to a 'size' boundary
+ *
+ * @param stack Stack
+ * @param size  New size boundary
+ */
+
+/**
+ * @def PUSH(stack, size, type)
+ *
+ * Allocates 'size' elements of type 'type' on the stack
+ *
+ * @param stack Stack
+ * @param size  Number of elements
+ * @param type  Type of element
+ */
+
+/**
+ * @def VARDECL(var)
+ *
+ * Declare variable on stack
+ *
+ * @param var Variable to declare
+ */
+
+/**
+ * @def ALLOC(var, size, type)
+ *
+ * Allocate 'size' elements of 'type' on stack
+ *
+ * @param var  Name of variable to allocate
+ * @param size Number of elements
+ * @param type Type of element
+ */
+
+#if defined(VAR_ARRAYS)
+
+#define VARDECL(type, var)
+#define ALLOC(var, size, type) type var[size]
+#define SAVE_STACK
+#define RESTORE_STACK
+#define ALLOC_STACK
+
+#elif defined(USE_ALLOCA)
+
+#define VARDECL(type, var) type *var
+
+# ifdef WIN32
+#  define ALLOC(var, size, type) var = ((type*)_alloca(sizeof(type)*(size)))
+# else
+#  define ALLOC(var, size, type) var = ((type*)alloca(sizeof(type)*(size)))
+# endif
+
+#define SAVE_STACK
+#define RESTORE_STACK
+#define ALLOC_STACK
+
+#else
+
+#ifdef CELT_C
+char *global_stack=0;
+#else
+extern char *global_stack;
+#endif /* CELT_C */
+
+#ifdef ENABLE_VALGRIND
+
+#include <valgrind/memcheck.h>
+
+#ifdef CELT_C
+char *global_stack_top=0;
+#else
+extern char *global_stack_top;
+#endif /* CELT_C */
+
+#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1))
+#define PUSH(stack, size, type) (VALGRIND_MAKE_MEM_NOACCESS(stack, global_stack_top-stack),ALIGN((stack),sizeof(type)/sizeof(char)),VALGRIND_MAKE_MEM_UNDEFINED(stack, ((size)*sizeof(type)/sizeof(char))),(stack)+=(2*(size)*sizeof(type)/sizeof(char)),(type*)((stack)-(2*(size)*sizeof(type)/sizeof(char))))
+#define RESTORE_STACK ((global_stack = _saved_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack))
+#define ALLOC_STACK char *_saved_stack; ((global_stack = (global_stack==0) ? ((global_stack_top=opus_alloc_scratch(GLOBAL_STACK_SIZE*2)+(GLOBAL_STACK_SIZE*2))-(GLOBAL_STACK_SIZE*2)) : global_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack)); _saved_stack = global_stack;
+
+#else
+
+#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1))
+#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)/sizeof(char)),(stack)+=(size)*(sizeof(type)/sizeof(char)),(type*)((stack)-(size)*(sizeof(type)/sizeof(char))))
+#define RESTORE_STACK (global_stack = _saved_stack)
+#define ALLOC_STACK char *_saved_stack; (global_stack = (global_stack==0) ? opus_alloc_scratch(GLOBAL_STACK_SIZE) : global_stack); _saved_stack = global_stack;
+
+#endif /* ENABLE_VALGRIND */
+
+#include "os_support.h"
+#define VARDECL(type, var) type *var
+#define ALLOC(var, size, type) var = PUSH(global_stack, size, type)
+#define SAVE_STACK char *_saved_stack = global_stack;
+
+#endif /* VAR_ARRAYS */
+
+#endif /* STACK_ALLOC_H */
diff --git a/lib/rbcodec/codecs/libopus/celt/static_modes_fixed.h b/lib/rbcodec/codecs/libopus/celt/static_modes_fixed.h
new file mode 100644
index 0000000000..216df9e605
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/static_modes_fixed.h
@@ -0,0 +1,595 @@
+/* The contents of this file was automatically generated by dump_modes.c
+   with arguments: 48000 960
+   It contains static definitions for some pre-defined modes. */
+#include "modes.h"
+#include "rate.h"
+
+#ifndef DEF_WINDOW120
+#define DEF_WINDOW120
+static const opus_val16 window120[120] = {
+2, 20, 55, 108, 178,
+266, 372, 494, 635, 792,
+966, 1157, 1365, 1590, 1831,
+2089, 2362, 2651, 2956, 3276,
+3611, 3961, 4325, 4703, 5094,
+5499, 5916, 6346, 6788, 7241,
+7705, 8179, 8663, 9156, 9657,
+10167, 10684, 11207, 11736, 12271,
+12810, 13353, 13899, 14447, 14997,
+15547, 16098, 16648, 17197, 17744,
+18287, 18827, 19363, 19893, 20418,
+20936, 21447, 21950, 22445, 22931,
+23407, 23874, 24330, 24774, 25208,
+25629, 26039, 26435, 26819, 27190,
+27548, 27893, 28224, 28541, 28845,
+29135, 29411, 29674, 29924, 30160,
+30384, 30594, 30792, 30977, 31151,
+31313, 31463, 31602, 31731, 31849,
+31958, 32057, 32148, 32229, 32303,
+32370, 32429, 32481, 32528, 32568,
+32604, 32634, 32661, 32683, 32701,
+32717, 32729, 32740, 32748, 32754,
+32758, 32762, 32764, 32766, 32767,
+32767, 32767, 32767, 32767, 32767,
+};
+#endif
+
+#ifndef DEF_LOGN400
+#define DEF_LOGN400
+static const opus_int16 logN400[21] = {
+0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 8, 16, 16, 16, 21, 21, 24, 29, 34, 36, };
+#endif
+
+#ifndef DEF_PULSE_CACHE50
+#define DEF_PULSE_CACHE50
+static const opus_int16 cache_index50[105] = {
+-1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41,
+82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41,
+41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41,
+41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305,
+318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240,
+305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240,
+240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387,
+};
+static const unsigned char cache_bits50[392] = {
+40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28,
+31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50,
+51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65,
+66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61,
+64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92,
+94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123,
+124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94,
+97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139,
+142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35,
+28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149,
+153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225,
+229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157,
+166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63,
+86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250,
+25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180,
+185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89,
+110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41,
+74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138,
+163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214,
+228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49,
+90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47,
+87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57,
+106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187,
+224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127,
+182, 234, };
+static const unsigned char cache_caps50[168] = {
+224, 224, 224, 224, 224, 224, 224, 224, 160, 160, 160, 160, 185, 185, 185,
+178, 178, 168, 134, 61, 37, 224, 224, 224, 224, 224, 224, 224, 224, 240,
+240, 240, 240, 207, 207, 207, 198, 198, 183, 144, 66, 40, 160, 160, 160,
+160, 160, 160, 160, 160, 185, 185, 185, 185, 193, 193, 193, 183, 183, 172,
+138, 64, 38, 240, 240, 240, 240, 240, 240, 240, 240, 207, 207, 207, 207,
+204, 204, 204, 193, 193, 180, 143, 66, 40, 185, 185, 185, 185, 185, 185,
+185, 185, 193, 193, 193, 193, 193, 193, 193, 183, 183, 172, 138, 65, 39,
+207, 207, 207, 207, 207, 207, 207, 207, 204, 204, 204, 204, 201, 201, 201,
+188, 188, 176, 141, 66, 40, 193, 193, 193, 193, 193, 193, 193, 193, 193,
+193, 193, 193, 194, 194, 194, 184, 184, 173, 139, 65, 39, 204, 204, 204,
+204, 204, 204, 204, 204, 201, 201, 201, 201, 198, 198, 198, 187, 187, 175,
+140, 66, 40, };
+#endif
+
+#ifndef FFT_TWIDDLES48000_960
+#define FFT_TWIDDLES48000_960
+static const kiss_twiddle_cpx fft_twiddles48000_960[480] = {
+{32767, 0}, {32766, -429},
+{32757, -858}, {32743, -1287},
+{32724, -1715}, {32698, -2143},
+{32667, -2570}, {32631, -2998},
+{32588, -3425}, {32541, -3851},
+{32488, -4277}, {32429, -4701},
+{32364, -5125}, {32295, -5548},
+{32219, -5971}, {32138, -6393},
+{32051, -6813}, {31960, -7231},
+{31863, -7650}, {31760, -8067},
+{31652, -8481}, {31539, -8895},
+{31419, -9306}, {31294, -9716},
+{31165, -10126}, {31030, -10532},
+{30889, -10937}, {30743, -11340},
+{30592, -11741}, {30436, -12141},
+{30274, -12540}, {30107, -12935},
+{29936, -13328}, {29758, -13718},
+{29577, -14107}, {29390, -14493},
+{29197, -14875}, {29000, -15257},
+{28797, -15635}, {28590, -16010},
+{28379, -16384}, {28162, -16753},
+{27940, -17119}, {27714, -17484},
+{27482, -17845}, {27246, -18205},
+{27006, -18560}, {26760, -18911},
+{26510, -19260}, {26257, -19606},
+{25997, -19947}, {25734, -20286},
+{25466, -20621}, {25194, -20952},
+{24918, -21281}, {24637, -21605},
+{24353, -21926}, {24063, -22242},
+{23770, -22555}, {23473, -22865},
+{23171, -23171}, {22866, -23472},
+{22557, -23769}, {22244, -24063},
+{21927, -24352}, {21606, -24636},
+{21282, -24917}, {20954, -25194},
+{20622, -25465}, {20288, -25733},
+{19949, -25997}, {19607, -26255},
+{19261, -26509}, {18914, -26760},
+{18561, -27004}, {18205, -27246},
+{17846, -27481}, {17485, -27713},
+{17122, -27940}, {16755, -28162},
+{16385, -28378}, {16012, -28590},
+{15636, -28797}, {15258, -28999},
+{14878, -29197}, {14494, -29389},
+{14108, -29576}, {13720, -29757},
+{13329, -29934}, {12937, -30107},
+{12540, -30274}, {12142, -30435},
+{11744, -30592}, {11342, -30743},
+{10939, -30889}, {10534, -31030},
+{10127, -31164}, {9718, -31294},
+{9307, -31418}, {8895, -31537},
+{8482, -31652}, {8067, -31759},
+{7650, -31862}, {7233, -31960},
+{6815, -32051}, {6393, -32138},
+{5973, -32219}, {5549, -32294},
+{5127, -32364}, {4703, -32429},
+{4278, -32487}, {3852, -32541},
+{3426, -32588}, {2999, -32630},
+{2572, -32667}, {2144, -32698},
+{1716, -32724}, {1287, -32742},
+{860, -32757}, {430, -32766},
+{0, -32767}, {-429, -32766},
+{-858, -32757}, {-1287, -32743},
+{-1715, -32724}, {-2143, -32698},
+{-2570, -32667}, {-2998, -32631},
+{-3425, -32588}, {-3851, -32541},
+{-4277, -32488}, {-4701, -32429},
+{-5125, -32364}, {-5548, -32295},
+{-5971, -32219}, {-6393, -32138},
+{-6813, -32051}, {-7231, -31960},
+{-7650, -31863}, {-8067, -31760},
+{-8481, -31652}, {-8895, -31539},
+{-9306, -31419}, {-9716, -31294},
+{-10126, -31165}, {-10532, -31030},
+{-10937, -30889}, {-11340, -30743},
+{-11741, -30592}, {-12141, -30436},
+{-12540, -30274}, {-12935, -30107},
+{-13328, -29936}, {-13718, -29758},
+{-14107, -29577}, {-14493, -29390},
+{-14875, -29197}, {-15257, -29000},
+{-15635, -28797}, {-16010, -28590},
+{-16384, -28379}, {-16753, -28162},
+{-17119, -27940}, {-17484, -27714},
+{-17845, -27482}, {-18205, -27246},
+{-18560, -27006}, {-18911, -26760},
+{-19260, -26510}, {-19606, -26257},
+{-19947, -25997}, {-20286, -25734},
+{-20621, -25466}, {-20952, -25194},
+{-21281, -24918}, {-21605, -24637},
+{-21926, -24353}, {-22242, -24063},
+{-22555, -23770}, {-22865, -23473},
+{-23171, -23171}, {-23472, -22866},
+{-23769, -22557}, {-24063, -22244},
+{-24352, -21927}, {-24636, -21606},
+{-24917, -21282}, {-25194, -20954},
+{-25465, -20622}, {-25733, -20288},
+{-25997, -19949}, {-26255, -19607},
+{-26509, -19261}, {-26760, -18914},
+{-27004, -18561}, {-27246, -18205},
+{-27481, -17846}, {-27713, -17485},
+{-27940, -17122}, {-28162, -16755},
+{-28378, -16385}, {-28590, -16012},
+{-28797, -15636}, {-28999, -15258},
+{-29197, -14878}, {-29389, -14494},
+{-29576, -14108}, {-29757, -13720},
+{-29934, -13329}, {-30107, -12937},
+{-30274, -12540}, {-30435, -12142},
+{-30592, -11744}, {-30743, -11342},
+{-30889, -10939}, {-31030, -10534},
+{-31164, -10127}, {-31294, -9718},
+{-31418, -9307}, {-31537, -8895},
+{-31652, -8482}, {-31759, -8067},
+{-31862, -7650}, {-31960, -7233},
+{-32051, -6815}, {-32138, -6393},
+{-32219, -5973}, {-32294, -5549},
+{-32364, -5127}, {-32429, -4703},
+{-32487, -4278}, {-32541, -3852},
+{-32588, -3426}, {-32630, -2999},
+{-32667, -2572}, {-32698, -2144},
+{-32724, -1716}, {-32742, -1287},
+{-32757, -860}, {-32766, -430},
+{-32767, 0}, {-32766, 429},
+{-32757, 858}, {-32743, 1287},
+{-32724, 1715}, {-32698, 2143},
+{-32667, 2570}, {-32631, 2998},
+{-32588, 3425}, {-32541, 3851},
+{-32488, 4277}, {-32429, 4701},
+{-32364, 5125}, {-32295, 5548},
+{-32219, 5971}, {-32138, 6393},
+{-32051, 6813}, {-31960, 7231},
+{-31863, 7650}, {-31760, 8067},
+{-31652, 8481}, {-31539, 8895},
+{-31419, 9306}, {-31294, 9716},
+{-31165, 10126}, {-31030, 10532},
+{-30889, 10937}, {-30743, 11340},
+{-30592, 11741}, {-30436, 12141},
+{-30274, 12540}, {-30107, 12935},
+{-29936, 13328}, {-29758, 13718},
+{-29577, 14107}, {-29390, 14493},
+{-29197, 14875}, {-29000, 15257},
+{-28797, 15635}, {-28590, 16010},
+{-28379, 16384}, {-28162, 16753},
+{-27940, 17119}, {-27714, 17484},
+{-27482, 17845}, {-27246, 18205},
+{-27006, 18560}, {-26760, 18911},
+{-26510, 19260}, {-26257, 19606},
+{-25997, 19947}, {-25734, 20286},
+{-25466, 20621}, {-25194, 20952},
+{-24918, 21281}, {-24637, 21605},
+{-24353, 21926}, {-24063, 22242},
+{-23770, 22555}, {-23473, 22865},
+{-23171, 23171}, {-22866, 23472},
+{-22557, 23769}, {-22244, 24063},
+{-21927, 24352}, {-21606, 24636},
+{-21282, 24917}, {-20954, 25194},
+{-20622, 25465}, {-20288, 25733},
+{-19949, 25997}, {-19607, 26255},
+{-19261, 26509}, {-18914, 26760},
+{-18561, 27004}, {-18205, 27246},
+{-17846, 27481}, {-17485, 27713},
+{-17122, 27940}, {-16755, 28162},
+{-16385, 28378}, {-16012, 28590},
+{-15636, 28797}, {-15258, 28999},
+{-14878, 29197}, {-14494, 29389},
+{-14108, 29576}, {-13720, 29757},
+{-13329, 29934}, {-12937, 30107},
+{-12540, 30274}, {-12142, 30435},
+{-11744, 30592}, {-11342, 30743},
+{-10939, 30889}, {-10534, 31030},
+{-10127, 31164}, {-9718, 31294},
+{-9307, 31418}, {-8895, 31537},
+{-8482, 31652}, {-8067, 31759},
+{-7650, 31862}, {-7233, 31960},
+{-6815, 32051}, {-6393, 32138},
+{-5973, 32219}, {-5549, 32294},
+{-5127, 32364}, {-4703, 32429},
+{-4278, 32487}, {-3852, 32541},
+{-3426, 32588}, {-2999, 32630},
+{-2572, 32667}, {-2144, 32698},
+{-1716, 32724}, {-1287, 32742},
+{-860, 32757}, {-430, 32766},
+{0, 32767}, {429, 32766},
+{858, 32757}, {1287, 32743},
+{1715, 32724}, {2143, 32698},
+{2570, 32667}, {2998, 32631},
+{3425, 32588}, {3851, 32541},
+{4277, 32488}, {4701, 32429},
+{5125, 32364}, {5548, 32295},
+{5971, 32219}, {6393, 32138},
+{6813, 32051}, {7231, 31960},
+{7650, 31863}, {8067, 31760},
+{8481, 31652}, {8895, 31539},
+{9306, 31419}, {9716, 31294},
+{10126, 31165}, {10532, 31030},
+{10937, 30889}, {11340, 30743},
+{11741, 30592}, {12141, 30436},
+{12540, 30274}, {12935, 30107},
+{13328, 29936}, {13718, 29758},
+{14107, 29577}, {14493, 29390},
+{14875, 29197}, {15257, 29000},
+{15635, 28797}, {16010, 28590},
+{16384, 28379}, {16753, 28162},
+{17119, 27940}, {17484, 27714},
+{17845, 27482}, {18205, 27246},
+{18560, 27006}, {18911, 26760},
+{19260, 26510}, {19606, 26257},
+{19947, 25997}, {20286, 25734},
+{20621, 25466}, {20952, 25194},
+{21281, 24918}, {21605, 24637},
+{21926, 24353}, {22242, 24063},
+{22555, 23770}, {22865, 23473},
+{23171, 23171}, {23472, 22866},
+{23769, 22557}, {24063, 22244},
+{24352, 21927}, {24636, 21606},
+{24917, 21282}, {25194, 20954},
+{25465, 20622}, {25733, 20288},
+{25997, 19949}, {26255, 19607},
+{26509, 19261}, {26760, 18914},
+{27004, 18561}, {27246, 18205},
+{27481, 17846}, {27713, 17485},
+{27940, 17122}, {28162, 16755},
+{28378, 16385}, {28590, 16012},
+{28797, 15636}, {28999, 15258},
+{29197, 14878}, {29389, 14494},
+{29576, 14108}, {29757, 13720},
+{29934, 13329}, {30107, 12937},
+{30274, 12540}, {30435, 12142},
+{30592, 11744}, {30743, 11342},
+{30889, 10939}, {31030, 10534},
+{31164, 10127}, {31294, 9718},
+{31418, 9307}, {31537, 8895},
+{31652, 8482}, {31759, 8067},
+{31862, 7650}, {31960, 7233},
+{32051, 6815}, {32138, 6393},
+{32219, 5973}, {32294, 5549},
+{32364, 5127}, {32429, 4703},
+{32487, 4278}, {32541, 3852},
+{32588, 3426}, {32630, 2999},
+{32667, 2572}, {32698, 2144},
+{32724, 1716}, {32742, 1287},
+{32757, 860}, {32766, 430},
+};
+#ifndef FFT_BITREV480
+#define FFT_BITREV480
+static const opus_int16 fft_bitrev480[480] = {
+0, 120, 240, 360, 30, 150, 270, 390, 60, 180, 300, 420, 90, 210, 330,
+450, 15, 135, 255, 375, 45, 165, 285, 405, 75, 195, 315, 435, 105, 225,
+345, 465, 5, 125, 245, 365, 35, 155, 275, 395, 65, 185, 305, 425, 95,
+215, 335, 455, 20, 140, 260, 380, 50, 170, 290, 410, 80, 200, 320, 440,
+110, 230, 350, 470, 10, 130, 250, 370, 40, 160, 280, 400, 70, 190, 310,
+430, 100, 220, 340, 460, 25, 145, 265, 385, 55, 175, 295, 415, 85, 205,
+325, 445, 115, 235, 355, 475, 1, 121, 241, 361, 31, 151, 271, 391, 61,
+181, 301, 421, 91, 211, 331, 451, 16, 136, 256, 376, 46, 166, 286, 406,
+76, 196, 316, 436, 106, 226, 346, 466, 6, 126, 246, 366, 36, 156, 276,
+396, 66, 186, 306, 426, 96, 216, 336, 456, 21, 141, 261, 381, 51, 171,
+291, 411, 81, 201, 321, 441, 111, 231, 351, 471, 11, 131, 251, 371, 41,
+161, 281, 401, 71, 191, 311, 431, 101, 221, 341, 461, 26, 146, 266, 386,
+56, 176, 296, 416, 86, 206, 326, 446, 116, 236, 356, 476, 2, 122, 242,
+362, 32, 152, 272, 392, 62, 182, 302, 422, 92, 212, 332, 452, 17, 137,
+257, 377, 47, 167, 287, 407, 77, 197, 317, 437, 107, 227, 347, 467, 7,
+127, 247, 367, 37, 157, 277, 397, 67, 187, 307, 427, 97, 217, 337, 457,
+22, 142, 262, 382, 52, 172, 292, 412, 82, 202, 322, 442, 112, 232, 352,
+472, 12, 132, 252, 372, 42, 162, 282, 402, 72, 192, 312, 432, 102, 222,
+342, 462, 27, 147, 267, 387, 57, 177, 297, 417, 87, 207, 327, 447, 117,
+237, 357, 477, 3, 123, 243, 363, 33, 153, 273, 393, 63, 183, 303, 423,
+93, 213, 333, 453, 18, 138, 258, 378, 48, 168, 288, 408, 78, 198, 318,
+438, 108, 228, 348, 468, 8, 128, 248, 368, 38, 158, 278, 398, 68, 188,
+308, 428, 98, 218, 338, 458, 23, 143, 263, 383, 53, 173, 293, 413, 83,
+203, 323, 443, 113, 233, 353, 473, 13, 133, 253, 373, 43, 163, 283, 403,
+73, 193, 313, 433, 103, 223, 343, 463, 28, 148, 268, 388, 58, 178, 298,
+418, 88, 208, 328, 448, 118, 238, 358, 478, 4, 124, 244, 364, 34, 154,
+274, 394, 64, 184, 304, 424, 94, 214, 334, 454, 19, 139, 259, 379, 49,
+169, 289, 409, 79, 199, 319, 439, 109, 229, 349, 469, 9, 129, 249, 369,
+39, 159, 279, 399, 69, 189, 309, 429, 99, 219, 339, 459, 24, 144, 264,
+384, 54, 174, 294, 414, 84, 204, 324, 444, 114, 234, 354, 474, 14, 134,
+254, 374, 44, 164, 284, 404, 74, 194, 314, 434, 104, 224, 344, 464, 29,
+149, 269, 389, 59, 179, 299, 419, 89, 209, 329, 449, 119, 239, 359, 479,
+};
+#endif
+
+#ifndef FFT_BITREV240
+#define FFT_BITREV240
+static const opus_int16 fft_bitrev240[240] = {
+0, 60, 120, 180, 15, 75, 135, 195, 30, 90, 150, 210, 45, 105, 165,
+225, 5, 65, 125, 185, 20, 80, 140, 200, 35, 95, 155, 215, 50, 110,
+170, 230, 10, 70, 130, 190, 25, 85, 145, 205, 40, 100, 160, 220, 55,
+115, 175, 235, 1, 61, 121, 181, 16, 76, 136, 196, 31, 91, 151, 211,
+46, 106, 166, 226, 6, 66, 126, 186, 21, 81, 141, 201, 36, 96, 156,
+216, 51, 111, 171, 231, 11, 71, 131, 191, 26, 86, 146, 206, 41, 101,
+161, 221, 56, 116, 176, 236, 2, 62, 122, 182, 17, 77, 137, 197, 32,
+92, 152, 212, 47, 107, 167, 227, 7, 67, 127, 187, 22, 82, 142, 202,
+37, 97, 157, 217, 52, 112, 172, 232, 12, 72, 132, 192, 27, 87, 147,
+207, 42, 102, 162, 222, 57, 117, 177, 237, 3, 63, 123, 183, 18, 78,
+138, 198, 33, 93, 153, 213, 48, 108, 168, 228, 8, 68, 128, 188, 23,
+83, 143, 203, 38, 98, 158, 218, 53, 113, 173, 233, 13, 73, 133, 193,
+28, 88, 148, 208, 43, 103, 163, 223, 58, 118, 178, 238, 4, 64, 124,
+184, 19, 79, 139, 199, 34, 94, 154, 214, 49, 109, 169, 229, 9, 69,
+129, 189, 24, 84, 144, 204, 39, 99, 159, 219, 54, 114, 174, 234, 14,
+74, 134, 194, 29, 89, 149, 209, 44, 104, 164, 224, 59, 119, 179, 239,
+};
+#endif
+
+#ifndef FFT_BITREV120
+#define FFT_BITREV120
+static const opus_int16 fft_bitrev120[120] = {
+0, 30, 60, 90, 15, 45, 75, 105, 5, 35, 65, 95, 20, 50, 80,
+110, 10, 40, 70, 100, 25, 55, 85, 115, 1, 31, 61, 91, 16, 46,
+76, 106, 6, 36, 66, 96, 21, 51, 81, 111, 11, 41, 71, 101, 26,
+56, 86, 116, 2, 32, 62, 92, 17, 47, 77, 107, 7, 37, 67, 97,
+22, 52, 82, 112, 12, 42, 72, 102, 27, 57, 87, 117, 3, 33, 63,
+93, 18, 48, 78, 108, 8, 38, 68, 98, 23, 53, 83, 113, 13, 43,
+73, 103, 28, 58, 88, 118, 4, 34, 64, 94, 19, 49, 79, 109, 9,
+39, 69, 99, 24, 54, 84, 114, 14, 44, 74, 104, 29, 59, 89, 119,
+};
+#endif
+
+#ifndef FFT_BITREV60
+#define FFT_BITREV60
+static const opus_int16 fft_bitrev60[60] = {
+0, 15, 30, 45, 5, 20, 35, 50, 10, 25, 40, 55, 1, 16, 31,
+46, 6, 21, 36, 51, 11, 26, 41, 56, 2, 17, 32, 47, 7, 22,
+37, 52, 12, 27, 42, 57, 3, 18, 33, 48, 8, 23, 38, 53, 13,
+28, 43, 58, 4, 19, 34, 49, 9, 24, 39, 54, 14, 29, 44, 59,
+};
+#endif
+
+#ifndef FFT_STATE48000_960_0
+#define FFT_STATE48000_960_0
+static const kiss_fft_state fft_state48000_960_0 = {
+480,    /* nfft */
+-1,     /* shift */
+{4, 120, 4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, }, /* factors */
+fft_bitrev480,  /* bitrev */
+fft_twiddles48000_960,  /* bitrev */
+};
+#endif
+
+#ifndef FFT_STATE48000_960_1
+#define FFT_STATE48000_960_1
+static const kiss_fft_state fft_state48000_960_1 = {
+240,    /* nfft */
+1,      /* shift */
+{4, 60, 4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, },   /* factors */
+fft_bitrev240,  /* bitrev */
+fft_twiddles48000_960,  /* bitrev */
+};
+#endif
+
+#ifndef FFT_STATE48000_960_2
+#define FFT_STATE48000_960_2
+static const kiss_fft_state fft_state48000_960_2 = {
+120,    /* nfft */
+2,      /* shift */
+{4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, },   /* factors */
+fft_bitrev120,  /* bitrev */
+fft_twiddles48000_960,  /* bitrev */
+};
+#endif
+
+#ifndef FFT_STATE48000_960_3
+#define FFT_STATE48000_960_3
+static const kiss_fft_state fft_state48000_960_3 = {
+60,     /* nfft */
+3,      /* shift */
+{4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, },    /* factors */
+fft_bitrev60,   /* bitrev */
+fft_twiddles48000_960,  /* bitrev */
+};
+#endif
+
+#endif
+
+#ifndef MDCT_TWIDDLES960
+#define MDCT_TWIDDLES960
+static const opus_val16 mdct_twiddles960[481] = {
+32767, 32767, 32767, 32767, 32766,
+32763, 32762, 32759, 32757, 32753,
+32751, 32747, 32743, 32738, 32733,
+32729, 32724, 32717, 32711, 32705,
+32698, 32690, 32683, 32676, 32667,
+32658, 32650, 32640, 32631, 32620,
+32610, 32599, 32588, 32577, 32566,
+32554, 32541, 32528, 32515, 32502,
+32487, 32474, 32459, 32444, 32429,
+32413, 32397, 32381, 32364, 32348,
+32331, 32313, 32294, 32277, 32257,
+32239, 32219, 32200, 32180, 32159,
+32138, 32118, 32096, 32074, 32051,
+32029, 32006, 31984, 31960, 31936,
+31912, 31888, 31863, 31837, 31812,
+31786, 31760, 31734, 31707, 31679,
+31652, 31624, 31596, 31567, 31539,
+31508, 31479, 31450, 31419, 31388,
+31357, 31326, 31294, 31262, 31230,
+31198, 31164, 31131, 31097, 31063,
+31030, 30994, 30959, 30924, 30889,
+30853, 30816, 30779, 30743, 30705,
+30668, 30629, 30592, 30553, 30515,
+30475, 30435, 30396, 30356, 30315,
+30274, 30233, 30191, 30149, 30107,
+30065, 30022, 29979, 29936, 29891,
+29847, 29803, 29758, 29713, 29668,
+29622, 29577, 29529, 29483, 29436,
+29390, 29341, 29293, 29246, 29197,
+29148, 29098, 29050, 29000, 28949,
+28899, 28848, 28797, 28746, 28694,
+28642, 28590, 28537, 28485, 28432,
+28378, 28324, 28271, 28217, 28162,
+28106, 28051, 27995, 27940, 27884,
+27827, 27770, 27713, 27657, 27598,
+27540, 27481, 27423, 27365, 27305,
+27246, 27187, 27126, 27066, 27006,
+26945, 26883, 26822, 26760, 26698,
+26636, 26574, 26510, 26448, 26383,
+26320, 26257, 26191, 26127, 26062,
+25997, 25931, 25866, 25800, 25734,
+25667, 25601, 25533, 25466, 25398,
+25330, 25262, 25194, 25125, 25056,
+24987, 24917, 24848, 24778, 24707,
+24636, 24566, 24495, 24424, 24352,
+24280, 24208, 24135, 24063, 23990,
+23917, 23842, 23769, 23695, 23622,
+23546, 23472, 23398, 23322, 23246,
+23171, 23095, 23018, 22942, 22866,
+22788, 22711, 22634, 22557, 22478,
+22400, 22322, 22244, 22165, 22085,
+22006, 21927, 21846, 21766, 21687,
+21606, 21524, 21443, 21363, 21282,
+21199, 21118, 21035, 20954, 20870,
+20788, 20705, 20621, 20538, 20455,
+20371, 20286, 20202, 20118, 20034,
+19947, 19863, 19777, 19692, 19606,
+19520, 19434, 19347, 19260, 19174,
+19088, 18999, 18911, 18825, 18737,
+18648, 18560, 18472, 18384, 18294,
+18205, 18116, 18025, 17936, 17846,
+17757, 17666, 17576, 17485, 17395,
+17303, 17212, 17122, 17030, 16937,
+16846, 16755, 16662, 16569, 16477,
+16385, 16291, 16198, 16105, 16012,
+15917, 15824, 15730, 15636, 15541,
+15447, 15352, 15257, 15162, 15067,
+14973, 14875, 14781, 14685, 14589,
+14493, 14396, 14300, 14204, 14107,
+14010, 13914, 13815, 13718, 13621,
+13524, 13425, 13328, 13230, 13133,
+13033, 12935, 12836, 12738, 12638,
+12540, 12441, 12341, 12241, 12142,
+12044, 11943, 11843, 11744, 11643,
+11542, 11442, 11342, 11241, 11139,
+11039, 10939, 10836, 10736, 10635,
+10534, 10431, 10330, 10228, 10127,
+10024, 9921, 9820, 9718, 9614,
+9512, 9410, 9306, 9204, 9101,
+8998, 8895, 8791, 8689, 8585,
+8481, 8377, 8274, 8171, 8067,
+7962, 7858, 7753, 7650, 7545,
+7441, 7336, 7231, 7129, 7023,
+6917, 6813, 6709, 6604, 6498,
+6393, 6288, 6182, 6077, 5973,
+5867, 5760, 5656, 5549, 5445,
+5339, 5232, 5127, 5022, 4914,
+4809, 4703, 4596, 4490, 4384,
+4278, 4171, 4065, 3958, 3852,
+3745, 3640, 3532, 3426, 3318,
+3212, 3106, 2998, 2891, 2786,
+2679, 2570, 2465, 2358, 2251,
+2143, 2037, 1929, 1823, 1715,
+1609, 1501, 1393, 1287, 1180,
+1073, 964, 858, 751, 644,
+535, 429, 322, 214, 107,
+0, };
+#endif
+
+static const CELTMode mode48000_960_120 = {
+48000,  /* Fs */
+120,    /* overlap */
+21,     /* nbEBands */
+21,     /* effEBands */
+{27853, 0, 4096, 8192, },       /* preemph */
+eband5ms,       /* eBands */
+3,      /* maxLM */
+8,      /* nbShortMdcts */
+120,    /* shortMdctSize */
+11,     /* nbAllocVectors */
+band_allocation,        /* allocVectors */
+logN400,        /* logN */
+window120,      /* window */
+{1920, 3, {&fft_state48000_960_0, &fft_state48000_960_1, &fft_state48000_960_2, &fft_state48000_960_3, }, mdct_twiddles960},    /* mdct */
+{392, cache_index50, cache_bits50, cache_caps50},       /* cache */
+};
+
+/* List of all the available modes */
+#define TOTAL_MODES 1
+static const CELTMode * const static_mode_list[TOTAL_MODES] = {
+&mode48000_960_120,
+};
diff --git a/lib/rbcodec/codecs/libopus/celt/vq.c b/lib/rbcodec/codecs/libopus/celt/vq.c
new file mode 100644
index 0000000000..6a00edf9cd
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/vq.c
@@ -0,0 +1,415 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "mathops.h"
+#include "cwrs.h"
+#include "vq.h"
+#include "arch.h"
+#include "os_support.h"
+#include "bands.h"
+#include "rate.h"
+
+static void exp_rotation1(celt_norm *X, int len, int stride, opus_val16 c, opus_val16 s)
+{
+   int i;
+   celt_norm *Xptr;
+   Xptr = X;
+   for (i=0;i<len-stride;i++)
+   {
+      celt_norm x1, x2;
+      x1 = Xptr[0];
+      x2 = Xptr[stride];
+      Xptr[stride] = EXTRACT16(SHR32(MULT16_16(c,x2) + MULT16_16(s,x1), 15));
+      *Xptr++      = EXTRACT16(SHR32(MULT16_16(c,x1) - MULT16_16(s,x2), 15));
+   }
+   Xptr = &X[len-2*stride-1];
+   for (i=len-2*stride-1;i>=0;i--)
+   {
+      celt_norm x1, x2;
+      x1 = Xptr[0];
+      x2 = Xptr[stride];
+      Xptr[stride] = EXTRACT16(SHR32(MULT16_16(c,x2) + MULT16_16(s,x1), 15));
+      *Xptr--      = EXTRACT16(SHR32(MULT16_16(c,x1) - MULT16_16(s,x2), 15));
+   }
+}
+
+static void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int spread)
+{
+   static const int SPREAD_FACTOR[3]={15,10,5};
+   int i;
+   opus_val16 c, s;
+   opus_val16 gain, theta;
+   int stride2=0;
+   int factor;
+
+   if (2*K>=len || spread==SPREAD_NONE)
+      return;
+   factor = SPREAD_FACTOR[spread-1];
+
+   gain = celt_div((opus_val32)MULT16_16(Q15_ONE,len),(opus_val32)(len+factor*K));
+   theta = HALF16(MULT16_16_Q15(gain,gain));
+
+   c = celt_cos_norm(EXTEND32(theta));
+   s = celt_cos_norm(EXTEND32(SUB16(Q15ONE,theta))); /*  sin(theta) */
+
+   if (len>=8*stride)
+   {
+      stride2 = 1;
+      /* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding.
+         It's basically incrementing long as (stride2+0.5)^2 < len/stride. */
+      while ((stride2*stride2+stride2)*stride + (stride>>2) < len)
+         stride2++;
+   }
+   /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for
+      extract_collapse_mask().*/
+   len /= stride;
+   for (i=0;i<stride;i++)
+   {
+      if (dir < 0)
+      {
+         if (stride2)
+            exp_rotation1(X+i*len, len, stride2, s, c);
+         exp_rotation1(X+i*len, len, 1, c, s);
+      } else {
+         exp_rotation1(X+i*len, len, 1, c, -s);
+         if (stride2)
+            exp_rotation1(X+i*len, len, stride2, s, -c);
+      }
+   }
+}
+
+/** Takes the pitch vector and the decoded residual vector, computes the gain
+    that will give ||p+g*y||=1 and mixes the residual with the pitch. */
+static void normalise_residual(int * OPUS_RESTRICT iy, celt_norm * OPUS_RESTRICT X,
+      int N, opus_val32 Ryy, opus_val16 gain)
+{
+   int i;
+#ifdef FIXED_POINT
+   int k;
+#endif
+   opus_val32 t;
+   opus_val16 g;
+
+#ifdef FIXED_POINT
+   k = celt_ilog2(Ryy)>>1;
+#endif
+   t = VSHR32(Ryy, 2*(k-7));
+   g = MULT16_16_P15(celt_rsqrt_norm(t),gain);
+
+   i=0;
+   do
+      X[i] = EXTRACT16(PSHR32(MULT16_16(g, iy[i]), k+1));
+   while (++i < N);
+}
+
+static unsigned extract_collapse_mask(int *iy, int N, int B)
+{
+   unsigned collapse_mask;
+   int N0;
+   int i;
+   if (B<=1)
+      return 1;
+   /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for
+      exp_rotation().*/
+   N0 = N/B;
+   collapse_mask = 0;
+   i=0; do {
+      int j;
+      j=0; do {
+         collapse_mask |= (iy[i*N0+j]!=0)<<i;
+      } while (++j<N0);
+   } while (++i<B);
+   return collapse_mask;
+}
+
+unsigned alg_quant(celt_norm *X, int N, int K, int spread, int B, ec_enc *enc
+#ifdef RESYNTH
+   , opus_val16 gain
+#endif
+   )
+{
+   VARDECL(celt_norm, y);
+   VARDECL(int, iy);
+   VARDECL(opus_val16, signx);
+   int i, j;
+   opus_val16 s;
+   int pulsesLeft;
+   opus_val32 sum;
+   opus_val32 xy;
+   opus_val16 yy;
+   unsigned collapse_mask;
+   SAVE_STACK;
+
+   celt_assert2(K>0, "alg_quant() needs at least one pulse");
+   celt_assert2(N>1, "alg_quant() needs at least two dimensions");
+
+   ALLOC(y, N, celt_norm);
+   ALLOC(iy, N, int);
+   ALLOC(signx, N, opus_val16);
+
+   exp_rotation(X, N, 1, B, K, spread);
+
+   /* Get rid of the sign */
+   sum = 0;
+   j=0; do {
+      if (X[j]>0)
+         signx[j]=1;
+      else {
+         signx[j]=-1;
+         X[j]=-X[j];
+      }
+      iy[j] = 0;
+      y[j] = 0;
+   } while (++j<N);
+
+   xy = yy = 0;
+
+   pulsesLeft = K;
+
+   /* Do a pre-search by projecting on the pyramid */
+   if (K > (N>>1))
+   {
+      opus_val16 rcp;
+      j=0; do {
+         sum += X[j];
+      }  while (++j<N);
+
+      /* If X is too small, just replace it with a pulse at 0 */
+#ifdef FIXED_POINT
+      if (sum <= K)
+#else
+      /* Prevents infinities and NaNs from causing too many pulses
+         to be allocated. 64 is an approximation of infinity here. */
+      if (!(sum > EPSILON && sum < 64))
+#endif
+      {
+         X[0] = QCONST16(1.f,14);
+         j=1; do
+            X[j]=0;
+         while (++j<N);
+         sum = QCONST16(1.f,14);
+      }
+      rcp = EXTRACT16(MULT16_32_Q16(K-1, celt_rcp(sum)));
+      j=0; do {
+#ifdef FIXED_POINT
+         /* It's really important to round *towards zero* here */
+         iy[j] = MULT16_16_Q15(X[j],rcp);
+#else
+         iy[j] = (int)floor(rcp*X[j]);
+#endif
+         y[j] = (celt_norm)iy[j];
+         yy = MAC16_16(yy, y[j],y[j]);
+         xy = MAC16_16(xy, X[j],y[j]);
+         y[j] *= 2;
+         pulsesLeft -= iy[j];
+      }  while (++j<N);
+   }
+   celt_assert2(pulsesLeft>=1, "Allocated too many pulses in the quick pass");
+
+   /* This should never happen, but just in case it does (e.g. on silence)
+      we fill the first bin with pulses. */
+#ifdef FIXED_POINT_DEBUG
+   celt_assert2(pulsesLeft<=N+3, "Not enough pulses in the quick pass");
+#endif
+   if (pulsesLeft > N+3)
+   {
+      opus_val16 tmp = (opus_val16)pulsesLeft;
+      yy = MAC16_16(yy, tmp, tmp);
+      yy = MAC16_16(yy, tmp, y[0]);
+      iy[0] += pulsesLeft;
+      pulsesLeft=0;
+   }
+
+   s = 1;
+   for (i=0;i<pulsesLeft;i++)
+   {
+      int best_id;
+      opus_val32 best_num = -VERY_LARGE16;
+      opus_val16 best_den = 0;
+#ifdef FIXED_POINT
+      int rshift;
+#endif
+#ifdef FIXED_POINT
+      rshift = 1+celt_ilog2(K-pulsesLeft+i+1);
+#endif
+      best_id = 0;
+      /* The squared magnitude term gets added anyway, so we might as well
+         add it outside the loop */
+      yy = ADD32(yy, 1);
+      j=0;
+      do {
+         opus_val16 Rxy, Ryy;
+         /* Temporary sums of the new pulse(s) */
+         Rxy = EXTRACT16(SHR32(ADD32(xy, EXTEND32(X[j])),rshift));
+         /* We're multiplying y[j] by two so we don't have to do it here */
+         Ryy = ADD16(yy, y[j]);
+
+         /* Approximate score: we maximise Rxy/sqrt(Ryy) (we're guaranteed that
+            Rxy is positive because the sign is pre-computed) */
+         Rxy = MULT16_16_Q15(Rxy,Rxy);
+         /* The idea is to check for num/den >= best_num/best_den, but that way
+            we can do it without any division */
+         /* OPT: Make sure to use conditional moves here */
+         if (MULT16_16(best_den, Rxy) > MULT16_16(Ryy, best_num))
+         {
+            best_den = Ryy;
+            best_num = Rxy;
+            best_id = j;
+         }
+      } while (++j<N);
+
+      /* Updating the sums of the new pulse(s) */
+      xy = ADD32(xy, EXTEND32(X[best_id]));
+      /* We're multiplying y[j] by two so we don't have to do it here */
+      yy = ADD16(yy, y[best_id]);
+
+      /* Only now that we've made the final choice, update y/iy */
+      /* Multiplying y[j] by 2 so we don't have to do it everywhere else */
+      y[best_id] += 2*s;
+      iy[best_id]++;
+   }
+
+   /* Put the original sign back */
+   j=0;
+   do {
+      X[j] = MULT16_16(signx[j],X[j]);
+      if (signx[j] < 0)
+         iy[j] = -iy[j];
+   } while (++j<N);
+   encode_pulses(iy, N, K, enc);
+
+#ifdef RESYNTH
+   normalise_residual(iy, X, N, yy, gain);
+   exp_rotation(X, N, -1, B, K, spread);
+#endif
+
+   collapse_mask = extract_collapse_mask(iy, N, B);
+   RESTORE_STACK;
+   return collapse_mask;
+}
+
+/** Decode pulse vector and combine the result with the pitch vector to produce
+    the final normalised signal in the current band. */
+unsigned alg_unquant(celt_norm *X, int N, int K, int spread, int B,
+      ec_dec *dec, opus_val16 gain)
+{
+   int i;
+   opus_val32 Ryy;
+   unsigned collapse_mask;
+   VARDECL(int, iy);
+   SAVE_STACK;
+
+   celt_assert2(K>0, "alg_unquant() needs at least one pulse");
+   celt_assert2(N>1, "alg_unquant() needs at least two dimensions");
+   ALLOC(iy, N, int);
+   decode_pulses(iy, N, K, dec);
+   Ryy = 0;
+   i=0;
+   do {
+      Ryy = MAC16_16(Ryy, iy[i], iy[i]);
+   } while (++i < N);
+   normalise_residual(iy, X, N, Ryy, gain);
+   exp_rotation(X, N, -1, B, K, spread);
+   collapse_mask = extract_collapse_mask(iy, N, B);
+   RESTORE_STACK;
+   return collapse_mask;
+}
+
+void renormalise_vector(celt_norm *X, int N, opus_val16 gain)
+{
+   int i;
+#ifdef FIXED_POINT
+   int k;
+#endif
+   opus_val32 E = EPSILON;
+   opus_val16 g;
+   opus_val32 t;
+   celt_norm *xptr = X;
+   for (i=0;i<N;i++)
+   {
+      E = MAC16_16(E, *xptr, *xptr);
+      xptr++;
+   }
+#ifdef FIXED_POINT
+   k = celt_ilog2(E)>>1;
+#endif
+   t = VSHR32(E, 2*(k-7));
+   g = MULT16_16_P15(celt_rsqrt_norm(t),gain);
+
+   xptr = X;
+   for (i=0;i<N;i++)
+   {
+      *xptr = EXTRACT16(PSHR32(MULT16_16(g, *xptr), k+1));
+      xptr++;
+   }
+   /*return celt_sqrt(E);*/
+}
+
+int stereo_itheta(celt_norm *X, celt_norm *Y, int stereo, int N)
+{
+   int i;
+   int itheta;
+   opus_val16 mid, side;
+   opus_val32 Emid, Eside;
+
+   Emid = Eside = EPSILON;
+   if (stereo)
+   {
+      for (i=0;i<N;i++)
+      {
+         celt_norm m, s;
+         m = ADD16(SHR16(X[i],1),SHR16(Y[i],1));
+         s = SUB16(SHR16(X[i],1),SHR16(Y[i],1));
+         Emid = MAC16_16(Emid, m, m);
+         Eside = MAC16_16(Eside, s, s);
+      }
+   } else {
+      for (i=0;i<N;i++)
+      {
+         celt_norm m, s;
+         m = X[i];
+         s = Y[i];
+         Emid = MAC16_16(Emid, m, m);
+         Eside = MAC16_16(Eside, s, s);
+      }
+   }
+   mid = celt_sqrt(Emid);
+   side = celt_sqrt(Eside);
+#ifdef FIXED_POINT
+   /* 0.63662 = 2/pi */
+   itheta = MULT16_16_Q15(QCONST16(0.63662f,15),celt_atan2p(side, mid));
+#else
+   itheta = (int)floor(.5f+16384*0.63662f*atan2(side,mid));
+#endif
+
+   return itheta;
+}
diff --git a/lib/rbcodec/codecs/libopus/celt/vq.h b/lib/rbcodec/codecs/libopus/celt/vq.h
new file mode 100644
index 0000000000..1ceeeeb268
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/celt/vq.h
@@ -0,0 +1,73 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/**
+   @file vq.h
+   @brief Vector quantisation of the residual
+ */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef VQ_H
+#define VQ_H
+
+#include "entenc.h"
+#include "entdec.h"
+#include "modes.h"
+
+/** Algebraic pulse-vector quantiser. The signal x is replaced by the sum of
+  * the pitch and a combination of pulses such that its norm is still equal
+  * to 1. This is the function that will typically require the most CPU.
+ * @param x Residual signal to quantise/encode (returns quantised version)
+ * @param W Perceptual weight to use when optimising (currently unused)
+ * @param N Number of samples to encode
+ * @param K Number of pulses to use
+ * @param p Pitch vector (it is assumed that p+x is a unit vector)
+ * @param enc Entropy encoder state
+ * @ret A mask indicating which blocks in the band received pulses
+*/
+unsigned alg_quant(celt_norm *X, int N, int K, int spread, int B,
+      ec_enc *enc
+#ifdef RESYNTH
+      , opus_val16 gain
+#endif
+      );
+
+/** Algebraic pulse decoder
+ * @param x Decoded normalised spectrum (returned)
+ * @param N Number of samples to decode
+ * @param K Number of pulses to use
+ * @param p Pitch vector (automatically added to x)
+ * @param dec Entropy decoder state
+ * @ret A mask indicating which blocks in the band received pulses
+ */
+unsigned alg_unquant(celt_norm *X, int N, int K, int spread, int B,
+      ec_dec *dec, opus_val16 gain);
+
+void renormalise_vector(celt_norm *X, int N, opus_val16 gain);
+
+int stereo_itheta(celt_norm *X, celt_norm *Y, int stereo, int N);
+
+#endif /* VQ_H */
diff --git a/lib/rbcodec/codecs/libopus/libopus.make b/lib/rbcodec/codecs/libopus/libopus.make
new file mode 100644
index 0000000000..1df52dbe4f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/libopus.make
@@ -0,0 +1,24 @@
+#             __________               __   ___.
+#   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+#   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+#   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+#   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+#                     \/            \/     \/    \/            \/
+# $Id$
+#
+
+# libopus
+OPUSLIB := $(CODECDIR)/libopus.a
+OPUSLIB_SRC := $(call preprocess, $(RBCODECLIB_DIR)/codecs/libopus/SOURCES)
+OPUSLIB_OBJ := $(call c2obj, $(OPUSLIB_SRC))
+
+# codec specific compilation flags
+$(OPUSLIB) : CODECFLAGS += -DHAVE_CONFIG_H \
+   -I$(RBCODECLIB_DIR)/codecs/libopus \
+   -I$(RBCODECLIB_DIR)/codecs/libopus/celt \
+   -I$(RBCODECLIB_DIR)/codecs/libopus/silk \
+   -I$(RBCODECLIB_DIR)/codecs/libopus/silk/fixed
+
+$(OPUSLIB): $(OPUSLIB_OBJ)
+        $(SILENT)$(shell rm -f $@)
+        $(call PRINTS,AR $(@F))$(AR) rcs $@ $^ >/dev/null
diff --git a/lib/rbcodec/codecs/libopus/ogg/framing.c b/lib/rbcodec/codecs/libopus/ogg/framing.c
new file mode 100644
index 0000000000..f007de176a
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/ogg/framing.c
@@ -0,0 +1,1025 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE Ogg CONTAINER SOURCE CODE.              *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2010             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: code raw packets into framed OggSquish stream and
+           decode Ogg streams back into raw packets
+ last mod: $Id: framing.c 18052 2011-08-04 17:57:02Z giles $
+
+ note: The CRC code is directly derived from public domain code by
+ Ross Williams (ross@guest.adelaide.edu.au).  See docs/framing.html
+ for details.
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <ogg/ogg.h>
+
+/* A complete description of Ogg framing exists in docs/framing.html */
+
+int ogg_page_version(const ogg_page *og){
+  return((int)(og->header[4]));
+}
+
+int ogg_page_continued(const ogg_page *og){
+  return((int)(og->header[5]&0x01));
+}
+
+int ogg_page_bos(const ogg_page *og){
+  return((int)(og->header[5]&0x02));
+}
+
+int ogg_page_eos(const ogg_page *og){
+  return((int)(og->header[5]&0x04));
+}
+
+ogg_int64_t ogg_page_granulepos(const ogg_page *og){
+  unsigned char *page=og->header;
+  ogg_int64_t granulepos=page[13]&(0xff);
+  granulepos= (granulepos<<8)|(page[12]&0xff);
+  granulepos= (granulepos<<8)|(page[11]&0xff);
+  granulepos= (granulepos<<8)|(page[10]&0xff);
+  granulepos= (granulepos<<8)|(page[9]&0xff);
+  granulepos= (granulepos<<8)|(page[8]&0xff);
+  granulepos= (granulepos<<8)|(page[7]&0xff);
+  granulepos= (granulepos<<8)|(page[6]&0xff);
+  return(granulepos);
+}
+
+int ogg_page_serialno(const ogg_page *og){
+  return(og->header[14] |
+         (og->header[15]<<8) |
+         (og->header[16]<<16) |
+         (og->header[17]<<24));
+}
+
+long ogg_page_pageno(const ogg_page *og){
+  return(og->header[18] |
+         (og->header[19]<<8) |
+         (og->header[20]<<16) |
+         (og->header[21]<<24));
+}
+
+
+
+/* returns the number of packets that are completed on this page (if
+   the leading packet is begun on a previous page, but ends on this
+   page, it's counted */
+
+/* NOTE:
+   If a page consists of a packet begun on a previous page, and a new
+   packet begun (but not completed) on this page, the return will be:
+     ogg_page_packets(page)   ==1,
+     ogg_page_continued(page) !=0
+
+   If a page happens to be a single packet that was begun on a
+   previous page, and spans to the next page (in the case of a three or
+   more page packet), the return will be:
+     ogg_page_packets(page)   ==0,
+     ogg_page_continued(page) !=0
+*/
+
+int ogg_page_packets(const ogg_page *og){
+  int i,n=og->header[26],count=0;
+  for(i=0;i<n;i++)
+    if(og->header[27+i]<255)count++;
+  return(count);
+}
+
+
+#if 0
+/* helper to initialize lookup for direct-table CRC (illustrative; we
+   use the static init below) */
+
+static ogg_uint32_t _ogg_crc_entry(unsigned long index){
+  int           i;
+  unsigned long r;
+
+  r = index << 24;
+  for (i=0; i<8; i++)
+    if (r & 0x80000000UL)
+      r = (r << 1) ^ 0x04c11db7; /* The same as the ethernet generator
+                                    polynomial, although we use an
+                                    unreflected alg and an init/final
+                                    of 0, not 0xffffffff */
+    else
+       r<<=1;
+ return (r & 0xffffffffUL);
+}
+#endif
+
+static const ogg_uint32_t crc_lookup[256]={
+  0x00000000,0x04c11db7,0x09823b6e,0x0d4326d9,
+  0x130476dc,0x17c56b6b,0x1a864db2,0x1e475005,
+  0x2608edb8,0x22c9f00f,0x2f8ad6d6,0x2b4bcb61,
+  0x350c9b64,0x31cd86d3,0x3c8ea00a,0x384fbdbd,
+  0x4c11db70,0x48d0c6c7,0x4593e01e,0x4152fda9,
+  0x5f15adac,0x5bd4b01b,0x569796c2,0x52568b75,
+  0x6a1936c8,0x6ed82b7f,0x639b0da6,0x675a1011,
+  0x791d4014,0x7ddc5da3,0x709f7b7a,0x745e66cd,
+  0x9823b6e0,0x9ce2ab57,0x91a18d8e,0x95609039,
+  0x8b27c03c,0x8fe6dd8b,0x82a5fb52,0x8664e6e5,
+  0xbe2b5b58,0xbaea46ef,0xb7a96036,0xb3687d81,
+  0xad2f2d84,0xa9ee3033,0xa4ad16ea,0xa06c0b5d,
+  0xd4326d90,0xd0f37027,0xddb056fe,0xd9714b49,
+  0xc7361b4c,0xc3f706fb,0xceb42022,0xca753d95,
+  0xf23a8028,0xf6fb9d9f,0xfbb8bb46,0xff79a6f1,
+  0xe13ef6f4,0xe5ffeb43,0xe8bccd9a,0xec7dd02d,
+  0x34867077,0x30476dc0,0x3d044b19,0x39c556ae,
+  0x278206ab,0x23431b1c,0x2e003dc5,0x2ac12072,
+  0x128e9dcf,0x164f8078,0x1b0ca6a1,0x1fcdbb16,
+  0x018aeb13,0x054bf6a4,0x0808d07d,0x0cc9cdca,
+  0x7897ab07,0x7c56b6b0,0x71159069,0x75d48dde,
+  0x6b93dddb,0x6f52c06c,0x6211e6b5,0x66d0fb02,
+  0x5e9f46bf,0x5a5e5b08,0x571d7dd1,0x53dc6066,
+  0x4d9b3063,0x495a2dd4,0x44190b0d,0x40d816ba,
+  0xaca5c697,0xa864db20,0xa527fdf9,0xa1e6e04e,
+  0xbfa1b04b,0xbb60adfc,0xb6238b25,0xb2e29692,
+  0x8aad2b2f,0x8e6c3698,0x832f1041,0x87ee0df6,
+  0x99a95df3,0x9d684044,0x902b669d,0x94ea7b2a,
+  0xe0b41de7,0xe4750050,0xe9362689,0xedf73b3e,
+  0xf3b06b3b,0xf771768c,0xfa325055,0xfef34de2,
+  0xc6bcf05f,0xc27dede8,0xcf3ecb31,0xcbffd686,
+  0xd5b88683,0xd1799b34,0xdc3abded,0xd8fba05a,
+  0x690ce0ee,0x6dcdfd59,0x608edb80,0x644fc637,
+  0x7a089632,0x7ec98b85,0x738aad5c,0x774bb0eb,
+  0x4f040d56,0x4bc510e1,0x46863638,0x42472b8f,
+  0x5c007b8a,0x58c1663d,0x558240e4,0x51435d53,
+  0x251d3b9e,0x21dc2629,0x2c9f00f0,0x285e1d47,
+  0x36194d42,0x32d850f5,0x3f9b762c,0x3b5a6b9b,
+  0x0315d626,0x07d4cb91,0x0a97ed48,0x0e56f0ff,
+  0x1011a0fa,0x14d0bd4d,0x19939b94,0x1d528623,
+  0xf12f560e,0xf5ee4bb9,0xf8ad6d60,0xfc6c70d7,
+  0xe22b20d2,0xe6ea3d65,0xeba91bbc,0xef68060b,
+  0xd727bbb6,0xd3e6a601,0xdea580d8,0xda649d6f,
+  0xc423cd6a,0xc0e2d0dd,0xcda1f604,0xc960ebb3,
+  0xbd3e8d7e,0xb9ff90c9,0xb4bcb610,0xb07daba7,
+  0xae3afba2,0xaafbe615,0xa7b8c0cc,0xa379dd7b,
+  0x9b3660c6,0x9ff77d71,0x92b45ba8,0x9675461f,
+  0x8832161a,0x8cf30bad,0x81b02d74,0x857130c3,
+  0x5d8a9099,0x594b8d2e,0x5408abf7,0x50c9b640,
+  0x4e8ee645,0x4a4ffbf2,0x470cdd2b,0x43cdc09c,
+  0x7b827d21,0x7f436096,0x7200464f,0x76c15bf8,
+  0x68860bfd,0x6c47164a,0x61043093,0x65c52d24,
+  0x119b4be9,0x155a565e,0x18197087,0x1cd86d30,
+  0x029f3d35,0x065e2082,0x0b1d065b,0x0fdc1bec,
+  0x3793a651,0x3352bbe6,0x3e119d3f,0x3ad08088,
+  0x2497d08d,0x2056cd3a,0x2d15ebe3,0x29d4f654,
+  0xc5a92679,0xc1683bce,0xcc2b1d17,0xc8ea00a0,
+  0xd6ad50a5,0xd26c4d12,0xdf2f6bcb,0xdbee767c,
+  0xe3a1cbc1,0xe760d676,0xea23f0af,0xeee2ed18,
+  0xf0a5bd1d,0xf464a0aa,0xf9278673,0xfde69bc4,
+  0x89b8fd09,0x8d79e0be,0x803ac667,0x84fbdbd0,
+  0x9abc8bd5,0x9e7d9662,0x933eb0bb,0x97ffad0c,
+  0xafb010b1,0xab710d06,0xa6322bdf,0xa2f33668,
+  0xbcb4666d,0xb8757bda,0xb5365d03,0xb1f740b4};
+
+/* init the encode/decode logical stream state */
+
+int ogg_stream_init(ogg_stream_state *os,int serialno){
+  if(os){
+    memset(os,0,sizeof(*os));
+    os->body_storage=16*1024;
+    os->lacing_storage=1024;
+
+    os->body_data=_ogg_malloc(os->body_storage*sizeof(*os->body_data));
+    os->lacing_vals=_ogg_malloc(os->lacing_storage*sizeof(*os->lacing_vals));
+    os->granule_vals=_ogg_malloc(os->lacing_storage*sizeof(*os->granule_vals));
+
+    if(!os->body_data || !os->lacing_vals || !os->granule_vals){
+      ogg_stream_clear(os);
+      return -1;
+    }
+
+    os->serialno=serialno;
+
+    return(0);
+  }
+  return(-1);
+}
+
+/* async/delayed error detection for the ogg_stream_state */
+int ogg_stream_check(ogg_stream_state *os){
+  if(!os || !os->body_data) return -1;
+  return 0;
+}
+
+/* _clear does not free os, only the non-flat storage within */
+int ogg_stream_clear(ogg_stream_state *os){
+  if(os){
+    if(os->body_data)_ogg_free(os->body_data);
+    if(os->lacing_vals)_ogg_free(os->lacing_vals);
+    if(os->granule_vals)_ogg_free(os->granule_vals);
+
+    memset(os,0,sizeof(*os));
+  }
+  return(0);
+}
+
+int ogg_stream_destroy(ogg_stream_state *os){
+  if(os){
+    ogg_stream_clear(os);
+    _ogg_free(os);
+  }
+  return(0);
+}
+
+/* Helpers for ogg_stream_encode; this keeps the structure and
+   what's happening fairly clear */
+
+static int _os_body_expand(ogg_stream_state *os,int needed){
+  if(os->body_storage<=os->body_fill+needed){
+    void *ret;
+    ret=_ogg_realloc(os->body_data,(os->body_storage+needed+1024)*
+                     sizeof(*os->body_data));
+    if(!ret){
+      ogg_stream_clear(os);
+      return -1;
+    }
+    os->body_storage+=(needed+1024);
+    os->body_data=ret;
+  }
+  return 0;
+}
+
+static int _os_lacing_expand(ogg_stream_state *os,int needed){
+  if(os->lacing_storage<=os->lacing_fill+needed){
+    void *ret;
+    ret=_ogg_realloc(os->lacing_vals,(os->lacing_storage+needed+32)*
+                     sizeof(*os->lacing_vals));
+    if(!ret){
+      ogg_stream_clear(os);
+      return -1;
+    }
+    os->lacing_vals=ret;
+    ret=_ogg_realloc(os->granule_vals,(os->lacing_storage+needed+32)*
+                     sizeof(*os->granule_vals));
+    if(!ret){
+      ogg_stream_clear(os);
+      return -1;
+    }
+    os->granule_vals=ret;
+    os->lacing_storage+=(needed+32);
+  }
+  return 0;
+}
+
+/* checksum the page */
+/* Direct table CRC; note that this will be faster in the future if we
+   perform the checksum simultaneously with other copies */
+
+void ogg_page_checksum_set(ogg_page *og){
+  if(og){
+    ogg_uint32_t crc_reg=0;
+    int i;
+
+    /* safety; needed for API behavior, but not framing code */
+    og->header[22]=0;
+    og->header[23]=0;
+    og->header[24]=0;
+    og->header[25]=0;
+
+    for(i=0;i<og->header_len;i++)
+      crc_reg=(crc_reg<<8)^crc_lookup[((crc_reg >> 24)&0xff)^og->header[i]];
+    for(i=0;i<og->body_len;i++)
+      crc_reg=(crc_reg<<8)^crc_lookup[((crc_reg >> 24)&0xff)^og->body[i]];
+
+    og->header[22]=(unsigned char)(crc_reg&0xff);
+    og->header[23]=(unsigned char)((crc_reg>>8)&0xff);
+    og->header[24]=(unsigned char)((crc_reg>>16)&0xff);
+    og->header[25]=(unsigned char)((crc_reg>>24)&0xff);
+  }
+}
+
+/* submit data to the internal buffer of the framing engine */
+int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov, int count,
+                       long e_o_s, ogg_int64_t granulepos){
+
+  int bytes = 0, lacing_vals, i;
+
+  if(ogg_stream_check(os)) return -1;
+  if(!iov) return 0;
+
+  for (i = 0; i < count; ++i) bytes += (int)iov[i].iov_len;
+  lacing_vals=bytes/255+1;
+
+  if(os->body_returned){
+    /* advance packet data according to the body_returned pointer. We
+       had to keep it around to return a pointer into the buffer last
+       call */
+
+    os->body_fill-=os->body_returned;
+    if(os->body_fill)
+      memmove(os->body_data,os->body_data+os->body_returned,
+              os->body_fill);
+    os->body_returned=0;
+  }
+
+  /* make sure we have the buffer storage */
+  if(_os_body_expand(os,bytes) || _os_lacing_expand(os,lacing_vals))
+    return -1;
+
+  /* Copy in the submitted packet.  Yes, the copy is a waste; this is
+     the liability of overly clean abstraction for the time being.  It
+     will actually be fairly easy to eliminate the extra copy in the
+     future */
+
+  for (i = 0; i < count; ++i) {
+    memcpy(os->body_data+os->body_fill, iov[i].iov_base, iov[i].iov_len);
+    os->body_fill += (int)iov[i].iov_len;
+  }
+
+  /* Store lacing vals for this packet */
+  for(i=0;i<lacing_vals-1;i++){
+    os->lacing_vals[os->lacing_fill+i]=255;
+    os->granule_vals[os->lacing_fill+i]=os->granulepos;
+  }
+  os->lacing_vals[os->lacing_fill+i]=bytes%255;
+  os->granulepos=os->granule_vals[os->lacing_fill+i]=granulepos;
+
+  /* flag the first segment as the beginning of the packet */
+  os->lacing_vals[os->lacing_fill]|= 0x100;
+
+  os->lacing_fill+=lacing_vals;
+
+  /* for the sake of completeness */
+  os->packetno++;
+
+  if(e_o_s)os->e_o_s=1;
+
+  return(0);
+}
+
+int ogg_stream_packetin(ogg_stream_state *os,ogg_packet *op){
+  ogg_iovec_t iov;
+  iov.iov_base = op->packet;
+  iov.iov_len = op->bytes;
+  return ogg_stream_iovecin(os, &iov, 1, op->e_o_s, op->granulepos);
+}
+
+/* Conditionally flush a page; force==0 will only flush nominal-size
+   pages, force==1 forces us to flush a page regardless of page size
+   so long as there's any data available at all. */
+static int ogg_stream_flush_i(ogg_stream_state *os,ogg_page *og, int force, int nfill){
+  int i;
+  int vals=0;
+  int maxvals=(os->lacing_fill>255?255:os->lacing_fill);
+  int bytes=0;
+  long acc=0;
+  ogg_int64_t granule_pos=-1;
+
+  if(ogg_stream_check(os)) return(0);
+  if(maxvals==0) return(0);
+
+  /* construct a page */
+  /* decide how many segments to include */
+
+  /* If this is the initial header case, the first page must only include
+     the initial header packet */
+  if(os->b_o_s==0){  /* 'initial header page' case */
+    granule_pos=0;
+    for(vals=0;vals<maxvals;vals++){
+      if((os->lacing_vals[vals]&0x0ff)<255){
+        vals++;
+        break;
+      }
+    }
+  }else{
+
+    /* The extra packets_done, packet_just_done logic here attempts to do two things:
+       1) Don't unneccessarily span pages.
+       2) Unless necessary, don't flush pages if there are less than four packets on
+          them; this expands page size to reduce unneccessary overhead if incoming packets
+          are large.
+       These are not necessary behaviors, just 'always better than naive flushing'
+       without requiring an application to explicitly request a specific optimized
+       behavior. We'll want an explicit behavior setup pathway eventually as well. */
+
+    int packets_done=0;
+    int packet_just_done=0;
+    for(vals=0;vals<maxvals;vals++){
+      if(acc>nfill && packet_just_done>=4){
+        force=1;
+        break;
+      }
+      acc+=os->lacing_vals[vals]&0x0ff;
+      if((os->lacing_vals[vals]&0xff)<255){
+        granule_pos=os->granule_vals[vals];
+        packet_just_done=++packets_done;
+      }else
+        packet_just_done=0;
+    }
+    if(vals==255)force=1;
+  }
+
+  if(!force) return(0);
+
+  /* construct the header in temp storage */
+  memcpy(os->header,"OggS",4);
+
+  /* stream structure version */
+  os->header[4]=0x00;
+
+  /* continued packet flag? */
+  os->header[5]=0x00;
+  if((os->lacing_vals[0]&0x100)==0)os->header[5]|=0x01;
+  /* first page flag? */
+  if(os->b_o_s==0)os->header[5]|=0x02;
+  /* last page flag? */
+  if(os->e_o_s && os->lacing_fill==vals)os->header[5]|=0x04;
+  os->b_o_s=1;
+
+  /* 64 bits of PCM position */
+  for(i=6;i<14;i++){
+    os->header[i]=(unsigned char)(granule_pos&0xff);
+    granule_pos>>=8;
+  }
+
+  /* 32 bits of stream serial number */
+  {
+    long serialno=os->serialno;
+    for(i=14;i<18;i++){
+      os->header[i]=(unsigned char)(serialno&0xff);
+      serialno>>=8;
+    }
+  }
+
+  /* 32 bits of page counter (we have both counter and page header
+     because this val can roll over) */
+  if(os->pageno==-1)os->pageno=0; /* because someone called
+                                     stream_reset; this would be a
+                                     strange thing to do in an
+                                     encode stream, but it has
+                                     plausible uses */
+  {
+    long pageno=os->pageno++;
+    for(i=18;i<22;i++){
+      os->header[i]=(unsigned char)(pageno&0xff);
+      pageno>>=8;
+    }
+  }
+
+  /* zero for computation; filled in later */
+  os->header[22]=0;
+  os->header[23]=0;
+  os->header[24]=0;
+  os->header[25]=0;
+
+  /* segment table */
+  os->header[26]=(unsigned char)(vals&0xff);
+  for(i=0;i<vals;i++)
+    bytes+=os->header[i+27]=(unsigned char)(os->lacing_vals[i]&0xff);
+
+  /* set pointers in the ogg_page struct */
+  og->header=os->header;
+  og->header_len=os->header_fill=vals+27;
+  og->body=os->body_data+os->body_returned;
+  og->body_len=bytes;
+
+  /* advance the lacing data and set the body_returned pointer */
+
+  os->lacing_fill-=vals;
+  memmove(os->lacing_vals,os->lacing_vals+vals,os->lacing_fill*sizeof(*os->lacing_vals));
+  memmove(os->granule_vals,os->granule_vals+vals,os->lacing_fill*sizeof(*os->granule_vals));
+  os->body_returned+=bytes;
+
+  /* calculate the checksum */
+
+  ogg_page_checksum_set(og);
+
+  /* done */
+  return(1);
+}
+
+/* This will flush remaining packets into a page (returning nonzero),
+   even if there is not enough data to trigger a flush normally
+   (undersized page). If there are no packets or partial packets to
+   flush, ogg_stream_flush returns 0.  Note that ogg_stream_flush will
+   try to flush a normal sized page like ogg_stream_pageout; a call to
+   ogg_stream_flush does not guarantee that all packets have flushed.
+   Only a return value of 0 from ogg_stream_flush indicates all packet
+   data is flushed into pages.
+
+   since ogg_stream_flush will flush the last page in a stream even if
+   it's undersized, you almost certainly want to use ogg_stream_pageout
+   (and *not* ogg_stream_flush) unless you specifically need to flush
+   a page regardless of size in the middle of a stream. */
+
+int ogg_stream_flush(ogg_stream_state *os,ogg_page *og){
+  return ogg_stream_flush_i(os,og,1,4096);
+}
+
+/* Like the above, but an argument is provided to adjust the nominal
+   page size for applications which are smart enough to provide their
+   own delay based flushing */
+
+int ogg_stream_flush_fill(ogg_stream_state *os,ogg_page *og, int nfill){
+  return ogg_stream_flush_i(os,og,1,nfill);
+}
+
+/* This constructs pages from buffered packet segments.  The pointers
+returned are to static buffers; do not free. The returned buffers are
+good only until the next call (using the same ogg_stream_state) */
+
+int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og){
+  int force=0;
+  if(ogg_stream_check(os)) return 0;
+
+  if((os->e_o_s&&os->lacing_fill) ||          /* 'were done, now flush' case */
+     (os->lacing_fill&&!os->b_o_s))           /* 'initial header page' case */
+    force=1;
+
+  return(ogg_stream_flush_i(os,og,force,4096));
+}
+
+/* Like the above, but an argument is provided to adjust the nominal
+page size for applications which are smart enough to provide their
+own delay based flushing */
+
+int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill){
+  int force=0;
+  if(ogg_stream_check(os)) return 0;
+
+  if((os->e_o_s&&os->lacing_fill) ||          /* 'were done, now flush' case */
+     (os->lacing_fill&&!os->b_o_s))           /* 'initial header page' case */
+    force=1;
+
+  return(ogg_stream_flush_i(os,og,force,nfill));
+}
+
+int ogg_stream_eos(ogg_stream_state *os){
+  if(ogg_stream_check(os)) return 1;
+  return os->e_o_s;
+}
+
+/* DECODING PRIMITIVES: packet streaming layer **********************/
+
+/* This has two layers to place more of the multi-serialno and paging
+   control in the application's hands.  First, we expose a data buffer
+   using ogg_sync_buffer().  The app either copies into the
+   buffer, or passes it directly to read(), etc.  We then call
+   ogg_sync_wrote() to tell how many bytes we just added.
+
+   Pages are returned (pointers into the buffer in ogg_sync_state)
+   by ogg_sync_pageout().  The page is then submitted to
+   ogg_stream_pagein() along with the appropriate
+   ogg_stream_state* (ie, matching serialno).  We then get raw
+   packets out calling ogg_stream_packetout() with a
+   ogg_stream_state. */
+
+/* initialize the struct to a known state */
+int ogg_sync_init(ogg_sync_state *oy){
+  if(oy){
+    oy->storage = -1; /* used as a readiness flag */
+    memset(oy,0,sizeof(*oy));
+  }
+  return(0);
+}
+
+/* clear non-flat storage within */
+int ogg_sync_clear(ogg_sync_state *oy){
+  if(oy){
+    if(oy->data)_ogg_free(oy->data);
+    memset(oy,0,sizeof(*oy));
+  }
+  return(0);
+}
+
+int ogg_sync_destroy(ogg_sync_state *oy){
+  if(oy){
+    ogg_sync_clear(oy);
+    _ogg_free(oy);
+  }
+  return(0);
+}
+
+int ogg_sync_check(ogg_sync_state *oy){
+  if(oy->storage<0) return -1;
+  return 0;
+}
+
+char *ogg_sync_buffer(ogg_sync_state *oy, long size){
+  if(ogg_sync_check(oy)) return NULL;
+
+  /* first, clear out any space that has been previously returned */
+  if(oy->returned){
+    oy->fill-=oy->returned;
+    if(oy->fill>0)
+      memmove(oy->data,oy->data+oy->returned,oy->fill);
+    oy->returned=0;
+  }
+
+  if(size>oy->storage-oy->fill){
+    /* We need to extend the internal buffer */
+    long newsize=size+oy->fill+4096; /* an extra page to be nice */
+    void *ret;
+
+    if(oy->data)
+      ret=_ogg_realloc(oy->data,newsize);
+    else
+      ret=_ogg_malloc(newsize);
+    if(!ret){
+      ogg_sync_clear(oy);
+      return NULL;
+    }
+    oy->data=ret;
+    oy->storage=newsize;
+  }
+
+  /* expose a segment at least as large as requested at the fill mark */
+  return((char *)oy->data+oy->fill);
+}
+
+int ogg_sync_wrote(ogg_sync_state *oy, long bytes){
+  if(ogg_sync_check(oy))return -1;
+  if(oy->fill+bytes>oy->storage)return -1;
+  oy->fill+=bytes;
+  return(0);
+}
+
+/* sync the stream.  This is meant to be useful for finding page
+   boundaries.
+
+   return values for this:
+  -n) skipped n bytes
+   0) page not ready; more data (no bytes skipped)
+   n) page synced at current location; page length n bytes
+
+*/
+
+long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og){
+  unsigned char *page=oy->data+oy->returned;
+  unsigned char *next;
+  long bytes=oy->fill-oy->returned;
+
+  if(ogg_sync_check(oy))return 0;
+
+  if(oy->headerbytes==0){
+    int headerbytes,i;
+    if(bytes<27)return(0); /* not enough for a header */
+
+    /* verify capture pattern */
+    if(memcmp(page,"OggS",4))goto sync_fail;
+
+    headerbytes=page[26]+27;
+    if(bytes<headerbytes)return(0); /* not enough for header + seg table */
+
+    /* count up body length in the segment table */
+
+    for(i=0;i<page[26];i++)
+      oy->bodybytes+=page[27+i];
+    oy->headerbytes=headerbytes;
+  }
+
+  if(oy->bodybytes+oy->headerbytes>bytes)return(0);
+
+  /* The whole test page is buffered.  Verify the checksum */
+  if (0) {
+    /* Grab the checksum bytes, set the header field to zero */
+    char chksum[4];
+    ogg_page log;
+
+    memcpy(chksum,page+22,4);
+    memset(page+22,0,4);
+
+    /* set up a temp page struct and recompute the checksum */
+    log.header=page;
+    log.header_len=oy->headerbytes;
+    log.body=page+oy->headerbytes;
+    log.body_len=oy->bodybytes;
+    ogg_page_checksum_set(&log);
+
+    /* Compare */
+    if(memcmp(chksum,page+22,4)){
+      /* D'oh.  Mismatch! Corrupt page (or miscapture and not a page
+         at all) */
+      /* replace the computed checksum with the one actually read in */
+      memcpy(page+22,chksum,4);
+
+      /* Bad checksum. Lose sync */
+      goto sync_fail;
+    }
+  }
+
+  /* yes, have a whole page all ready to go */
+  {
+    unsigned char *page=oy->data+oy->returned;
+    long bytes;
+
+    if(og){
+      og->header=page;
+      og->header_len=oy->headerbytes;
+      og->body=page+oy->headerbytes;
+      og->body_len=oy->bodybytes;
+    }
+
+    oy->unsynced=0;
+    oy->returned+=(bytes=oy->headerbytes+oy->bodybytes);
+    oy->headerbytes=0;
+    oy->bodybytes=0;
+    return(bytes);
+  }
+
+ sync_fail:
+
+  oy->headerbytes=0;
+  oy->bodybytes=0;
+
+  /* search for possible capture */
+  next=memchr(page+1,'O',bytes-1);
+  if(!next)
+    next=oy->data+oy->fill;
+
+  oy->returned=(int)(next-oy->data);
+  return((long)-(next-page));
+}
+
+/* sync the stream and get a page.  Keep trying until we find a page.
+   Suppress 'sync errors' after reporting the first.
+
+   return values:
+   -1) recapture (hole in data)
+    0) need more data
+    1) page returned
+
+   Returns pointers into buffered data; invalidated by next call to
+   _stream, _clear, _init, or _buffer */
+
+int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og){
+
+  if(ogg_sync_check(oy))return 0;
+
+  /* all we need to do is verify a page at the head of the stream
+     buffer.  If it doesn't verify, we look for the next potential
+     frame */
+
+  for(;;){
+    long ret=ogg_sync_pageseek(oy,og);
+    if(ret>0){
+      /* have a page */
+      return(1);
+    }
+    if(ret==0){
+      /* need more data */
+      return(0);
+    }
+
+    /* head did not start a synced page... skipped some bytes */
+    if(!oy->unsynced){
+      oy->unsynced=1;
+      return(-1);
+    }
+
+    /* loop. keep looking */
+  }
+}
+
+/* add the incoming page to the stream state; we decompose the page
+   into packet segments here as well. */
+
+int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og){
+  unsigned char *header=og->header;
+  unsigned char *body=og->body;
+  long           bodysize=og->body_len;
+  int            segptr=0;
+
+  int version=ogg_page_version(og);
+  int continued=ogg_page_continued(og);
+  int bos=ogg_page_bos(og);
+  int eos=ogg_page_eos(og);
+  ogg_int64_t granulepos=ogg_page_granulepos(og);
+  int serialno=ogg_page_serialno(og);
+  long pageno=ogg_page_pageno(og);
+  int segments=header[26];
+
+  if(ogg_stream_check(os)) return -1;
+
+  /* clean up 'returned data' */
+  {
+    long lr=os->lacing_returned;
+    long br=os->body_returned;
+
+    /* body data */
+    if(br){
+      os->body_fill-=br;
+      if(os->body_fill)
+        memmove(os->body_data,os->body_data+br,os->body_fill);
+      os->body_returned=0;
+    }
+
+    if(lr){
+      /* segment table */
+      if(os->lacing_fill-lr){
+        memmove(os->lacing_vals,os->lacing_vals+lr,
+                (os->lacing_fill-lr)*sizeof(*os->lacing_vals));
+        memmove(os->granule_vals,os->granule_vals+lr,
+                (os->lacing_fill-lr)*sizeof(*os->granule_vals));
+      }
+      os->lacing_fill-=lr;
+      os->lacing_packet-=lr;
+      os->lacing_returned=0;
+    }
+  }
+
+  /* check the serial number */
+  if(serialno!=os->serialno)return(-1);
+  if(version>0)return(-1);
+
+  if(_os_lacing_expand(os,segments+1)) return -1;
+
+  /* are we in sequence? */
+  if(pageno!=os->pageno){
+    int i;
+
+    /* unroll previous partial packet (if any) */
+    for(i=os->lacing_packet;i<os->lacing_fill;i++)
+      os->body_fill-=os->lacing_vals[i]&0xff;
+    os->lacing_fill=os->lacing_packet;
+
+    /* make a note of dropped data in segment table */
+    if(os->pageno!=-1){
+      os->lacing_vals[os->lacing_fill++]=0x400;
+      os->lacing_packet++;
+    }
+  }
+
+  /* are we a 'continued packet' page?  If so, we may need to skip
+     some segments */
+  if(continued){
+    if(os->lacing_fill<1 ||
+       os->lacing_vals[os->lacing_fill-1]==0x400){
+      bos=0;
+      for(;segptr<segments;segptr++){
+        int val=header[27+segptr];
+        body+=val;
+        bodysize-=val;
+        if(val<255){
+          segptr++;
+          break;
+        }
+      }
+    }
+  }
+
+  if(bodysize){
+    if(_os_body_expand(os,bodysize)) return -1;
+    memcpy(os->body_data+os->body_fill,body,bodysize);
+    os->body_fill+=bodysize;
+  }
+
+  {
+    int saved=-1;
+    while(segptr<segments){
+      int val=header[27+segptr];
+      os->lacing_vals[os->lacing_fill]=val;
+      os->granule_vals[os->lacing_fill]=-1;
+
+      if(bos){
+        os->lacing_vals[os->lacing_fill]|=0x100;
+        bos=0;
+      }
+
+      if(val<255)saved=os->lacing_fill;
+
+      os->lacing_fill++;
+      segptr++;
+
+      if(val<255)os->lacing_packet=os->lacing_fill;
+    }
+
+    /* set the granulepos on the last granuleval of the last full packet */
+    if(saved!=-1){
+      os->granule_vals[saved]=granulepos;
+    }
+
+  }
+
+  if(eos){
+    os->e_o_s=1;
+    if(os->lacing_fill>0)
+      os->lacing_vals[os->lacing_fill-1]|=0x200;
+  }
+
+  os->pageno=pageno+1;
+
+  return(0);
+}
+
+/* clear things to an initial state.  Good to call, eg, before seeking */
+int ogg_sync_reset(ogg_sync_state *oy){
+  if(ogg_sync_check(oy))return -1;
+
+  oy->fill=0;
+  oy->returned=0;
+  oy->unsynced=0;
+  oy->headerbytes=0;
+  oy->bodybytes=0;
+  return(0);
+}
+
+int ogg_stream_reset(ogg_stream_state *os){
+  if(ogg_stream_check(os)) return -1;
+
+  os->body_fill=0;
+  os->body_returned=0;
+
+  os->lacing_fill=0;
+  os->lacing_packet=0;
+  os->lacing_returned=0;
+
+  os->header_fill=0;
+
+  os->e_o_s=0;
+  os->b_o_s=0;
+  os->pageno=-1;
+  os->packetno=0;
+  os->granulepos=0;
+
+  return(0);
+}
+
+int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno){
+  if(ogg_stream_check(os)) return -1;
+  ogg_stream_reset(os);
+  os->serialno=serialno;
+  return(0);
+}
+
+static int _packetout(ogg_stream_state *os,ogg_packet *op,int adv){
+
+  /* The last part of decode. We have the stream broken into packet
+     segments.  Now we need to group them into packets (or return the
+     out of sync markers) */
+
+  int ptr=os->lacing_returned;
+
+  if(os->lacing_packet<=ptr)return(0);
+
+  if(os->lacing_vals[ptr]&0x400){
+    /* we need to tell the codec there's a gap; it might need to
+       handle previous packet dependencies. */
+    os->lacing_returned++;
+    os->packetno++;
+    return(-1);
+  }
+
+  if(!op && !adv)return(1); /* just using peek as an inexpensive way
+                               to ask if there's a whole packet
+                               waiting */
+
+  /* Gather the whole packet. We'll have no holes or a partial packet */
+  {
+    int size=os->lacing_vals[ptr]&0xff;
+    long bytes=size;
+    int eos=os->lacing_vals[ptr]&0x200; /* last packet of the stream? */
+    int bos=os->lacing_vals[ptr]&0x100; /* first packet of the stream? */
+
+    while(size==255){
+      int val=os->lacing_vals[++ptr];
+      size=val&0xff;
+      if(val&0x200)eos=0x200;
+      bytes+=size;
+    }
+
+    if(op){
+      op->e_o_s=eos;
+      op->b_o_s=bos;
+      op->packet=os->body_data+os->body_returned;
+      op->packetno=os->packetno;
+      op->granulepos=os->granule_vals[ptr];
+      op->bytes=bytes;
+    }
+
+    if(adv){
+      os->body_returned+=bytes;
+      os->lacing_returned=ptr+1;
+      os->packetno++;
+    }
+  }
+  return(1);
+}
+
+int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op){
+  if(ogg_stream_check(os)) return 0;
+  return _packetout(os,op,1);
+}
+
+int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op){
+  if(ogg_stream_check(os)) return 0;
+  return _packetout(os,op,0);
+}
+
+void ogg_packet_clear(ogg_packet *op) {
+  _ogg_free(op->packet);
+  memset(op, 0, sizeof(*op));
+}
+
diff --git a/lib/rbcodec/codecs/libopus/ogg/ogg.h b/lib/rbcodec/codecs/libopus/ogg/ogg.h
new file mode 100644
index 0000000000..00975ca354
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/ogg/ogg.h
@@ -0,0 +1,210 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: toplevel libogg include
+ last mod: $Id: ogg.h 18044 2011-08-01 17:55:20Z gmaxwell $
+
+ ********************************************************************/
+#ifndef _OGG_H
+#define _OGG_H
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include <stddef.h>
+#include "os_types.h"
+
+typedef struct {
+  void *iov_base;
+  size_t iov_len;
+} ogg_iovec_t;
+
+typedef struct {
+  long endbyte;
+  int  endbit;
+
+  unsigned char *buffer;
+  unsigned char *ptr;
+  long storage;
+} oggpack_buffer;
+
+/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/
+
+typedef struct {
+  unsigned char *header;
+  long header_len;
+  unsigned char *body;
+  long body_len;
+} ogg_page;
+
+/* ogg_stream_state contains the current encode/decode state of a logical
+   Ogg bitstream **********************************************************/
+
+typedef struct {
+  unsigned char   *body_data;    /* bytes from packet bodies */
+  long    body_storage;          /* storage elements allocated */
+  long    body_fill;             /* elements stored; fill mark */
+  long    body_returned;         /* elements of fill returned */
+
+
+  int     *lacing_vals;      /* The values that will go to the segment table */
+  ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact
+                                this way, but it is simple coupled to the
+                                lacing fifo */
+  long    lacing_storage;
+  long    lacing_fill;
+  long    lacing_packet;
+  long    lacing_returned;
+
+  unsigned char    header[282];      /* working space for header encode */
+  int              header_fill;
+
+  int     e_o_s;          /* set when we have buffered the last packet in the
+                             logical bitstream */
+  int     b_o_s;          /* set after we've written the initial page
+                             of a logical bitstream */
+  long    serialno;
+  long    pageno;
+  ogg_int64_t  packetno;  /* sequence number for decode; the framing
+                             knows where there's a hole in the data,
+                             but we need coupling so that the codec
+                             (which is in a separate abstraction
+                             layer) also knows about the gap */
+  ogg_int64_t   granulepos;
+
+} ogg_stream_state;
+
+/* ogg_packet is used to encapsulate the data and metadata belonging
+   to a single raw Ogg/Vorbis packet *************************************/
+
+typedef struct {
+  unsigned char *packet;
+  long  bytes;
+  long  b_o_s;
+  long  e_o_s;
+
+  ogg_int64_t  granulepos;
+
+  ogg_int64_t  packetno;     /* sequence number for decode; the framing
+                                knows where there's a hole in the data,
+                                but we need coupling so that the codec
+                                (which is in a separate abstraction
+                                layer) also knows about the gap */
+} ogg_packet;
+
+typedef struct {
+  unsigned char *data;
+  int storage;
+  int fill;
+  int returned;
+
+  int unsynced;
+  int headerbytes;
+  int bodybytes;
+} ogg_sync_state;
+
+/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/
+
+extern void  oggpack_writeinit(oggpack_buffer *b);
+extern int   oggpack_writecheck(oggpack_buffer *b);
+extern void  oggpack_writetrunc(oggpack_buffer *b,long bits);
+extern void  oggpack_writealign(oggpack_buffer *b);
+extern void  oggpack_writecopy(oggpack_buffer *b,void *source,long bits);
+extern void  oggpack_reset(oggpack_buffer *b);
+extern void  oggpack_writeclear(oggpack_buffer *b);
+extern void  oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
+extern void  oggpack_write(oggpack_buffer *b,unsigned long value,int bits);
+extern long  oggpack_look(oggpack_buffer *b,int bits);
+extern long  oggpack_look1(oggpack_buffer *b);
+extern void  oggpack_adv(oggpack_buffer *b,int bits);
+extern void  oggpack_adv1(oggpack_buffer *b);
+extern long  oggpack_read(oggpack_buffer *b,int bits);
+extern long  oggpack_read1(oggpack_buffer *b);
+extern long  oggpack_bytes(oggpack_buffer *b);
+extern long  oggpack_bits(oggpack_buffer *b);
+extern unsigned char *oggpack_get_buffer(oggpack_buffer *b);
+
+extern void  oggpackB_writeinit(oggpack_buffer *b);
+extern int   oggpackB_writecheck(oggpack_buffer *b);
+extern void  oggpackB_writetrunc(oggpack_buffer *b,long bits);
+extern void  oggpackB_writealign(oggpack_buffer *b);
+extern void  oggpackB_writecopy(oggpack_buffer *b,void *source,long bits);
+extern void  oggpackB_reset(oggpack_buffer *b);
+extern void  oggpackB_writeclear(oggpack_buffer *b);
+extern void  oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
+extern void  oggpackB_write(oggpack_buffer *b,unsigned long value,int bits);
+extern long  oggpackB_look(oggpack_buffer *b,int bits);
+extern long  oggpackB_look1(oggpack_buffer *b);
+extern void  oggpackB_adv(oggpack_buffer *b,int bits);
+extern void  oggpackB_adv1(oggpack_buffer *b);
+extern long  oggpackB_read(oggpack_buffer *b,int bits);
+extern long  oggpackB_read1(oggpack_buffer *b);
+extern long  oggpackB_bytes(oggpack_buffer *b);
+extern long  oggpackB_bits(oggpack_buffer *b);
+extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b);
+
+/* Ogg BITSTREAM PRIMITIVES: encoding **************************/
+
+extern int      ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op);
+extern int      ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov,
+                                   int count, long e_o_s, ogg_int64_t granulepos);
+extern int      ogg_stream_pageout(ogg_stream_state *os, ogg_page *og);
+extern int      ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill);
+extern int      ogg_stream_flush(ogg_stream_state *os, ogg_page *og);
+extern int      ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int nfill);
+
+/* Ogg BITSTREAM PRIMITIVES: decoding **************************/
+
+extern int      ogg_sync_init(ogg_sync_state *oy);
+extern int      ogg_sync_clear(ogg_sync_state *oy);
+extern int      ogg_sync_reset(ogg_sync_state *oy);
+extern int      ogg_sync_destroy(ogg_sync_state *oy);
+extern int      ogg_sync_check(ogg_sync_state *oy);
+
+extern char    *ogg_sync_buffer(ogg_sync_state *oy, long size);
+extern int      ogg_sync_wrote(ogg_sync_state *oy, long bytes);
+extern long     ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og);
+extern int      ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og);
+extern int      ogg_stream_pagein(ogg_stream_state *os, ogg_page *og);
+extern int      ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op);
+extern int      ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op);
+
+/* Ogg BITSTREAM PRIMITIVES: general ***************************/
+
+extern int      ogg_stream_init(ogg_stream_state *os,int serialno);
+extern int      ogg_stream_clear(ogg_stream_state *os);
+extern int      ogg_stream_reset(ogg_stream_state *os);
+extern int      ogg_stream_reset_serialno(ogg_stream_state *os,int serialno);
+extern int      ogg_stream_destroy(ogg_stream_state *os);
+extern int      ogg_stream_check(ogg_stream_state *os);
+extern int      ogg_stream_eos(ogg_stream_state *os);
+
+extern void     ogg_page_checksum_set(ogg_page *og);
+
+extern int      ogg_page_version(const ogg_page *og);
+extern int      ogg_page_continued(const ogg_page *og);
+extern int      ogg_page_bos(const ogg_page *og);
+extern int      ogg_page_eos(const ogg_page *og);
+extern ogg_int64_t  ogg_page_granulepos(const ogg_page *og);
+extern int      ogg_page_serialno(const ogg_page *og);
+extern long     ogg_page_pageno(const ogg_page *og);
+extern int      ogg_page_packets(const ogg_page *og);
+
+extern void     ogg_packet_clear(ogg_packet *op);
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  /* _OGG_H */
diff --git a/lib/rbcodec/codecs/libopus/ogg/os_types.h b/lib/rbcodec/codecs/libopus/ogg/os_types.h
new file mode 100644
index 0000000000..55f0bf559c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/ogg/os_types.h
@@ -0,0 +1,56 @@
+#include "config.h"
+#include <stdint.h>
+#include "codeclib.h"
+
+#ifdef SIMULATOR
+
+#include <stdio.h>
+
+static inline void* _ogg_malloc(size_t size)
+{
+    void *buf;
+
+    printf("ogg_malloc %d", size);
+    buf = codec_malloc(size);
+    printf(" = %p\n", buf);
+    
+    return buf;
+}
+
+static inline void* _ogg_calloc(size_t nmemb, size_t size)
+{
+    printf("ogg_calloc %d %d\n", nmemb, size);
+    return codec_calloc(nmemb, size);
+}
+
+static inline void* _ogg_realloc(void *ptr, size_t size)
+{
+    void *buf;
+
+    printf("ogg_realloc %p %d", ptr, size);
+    buf = codec_realloc(ptr, size);
+    printf(" = %p\n", buf);
+    return buf;
+}
+
+static inline void _ogg_free(void *ptr)
+{
+    printf("ogg_free %p\n", ptr);
+    codec_free(ptr);
+}
+
+#else
+
+#define _ogg_malloc  codec_malloc
+#define _ogg_calloc  codec_calloc
+#define _ogg_realloc codec_realloc
+#define _ogg_free    codec_free
+
+#endif
+
+typedef int16_t ogg_int16_t;
+typedef uint16_t ogg_uint16_t;
+typedef int32_t ogg_int32_t;
+typedef uint32_t ogg_uint32_t;
+typedef int64_t ogg_int64_t;
+
diff --git a/lib/rbcodec/codecs/libopus/opus.h b/lib/rbcodec/codecs/libopus/opus.h
new file mode 100644
index 0000000000..c242fec0e7
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus.h
@@ -0,0 +1,882 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended)</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_frame() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 1 byte, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the variable bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the variable bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case or failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occured. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum frame size (120 ms), this function will
+  *  not be capable of decoding some packets.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm,
+  *  if less than the maximum frame size (120ms) some frames can not be decoded
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>short[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   short size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms.
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */
diff --git a/lib/rbcodec/codecs/libopus/opus_config.h b/lib/rbcodec/codecs/libopus/opus_config.h
new file mode 100644
index 0000000000..86210df52b
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_config.h
@@ -0,0 +1,42 @@
+#ifndef CONFIG_H
+#define CONFIG_H
+
+#include "config.h"
+#include "codeclib.h"
+
+/* general stuff */
+#define OPUS_BUILD
+
+/* alloc stuff */
+#define NONTHREADSAFE_PSEUDOSTACK
+
+#define OVERRIDE_OPUS_ALLOC
+#define OVERRIDE_OPUS_FREE
+#define OVERRIDE_OPUS_ALLOC_SCRATCH
+
+#define opus_alloc          codec_malloc
+#define opus_free           codec_free
+#define opus_alloc_scratch  codec_malloc
+
+/* lrint */
+#define HAVE_LRINTF 0
+#define HAVE_LRINT  0
+
+/* embedded stuff */
+#define FIXED_POINT
+#define DISABLE_FLOAT_API
+#define EMBEDDED_ARM 1
+
+/* undefinitions */
+#ifdef ABS
+#undef ABS
+#endif
+#ifdef MIN
+#undef MIN
+#endif
+#ifdef MAX
+#undef MAX
+#endif
+
+#endif /* CONFIG_H */
+
diff --git a/lib/rbcodec/codecs/libopus/opus_custom.h b/lib/rbcodec/codecs/libopus/opus_custom.h
new file mode 100644
index 0000000000..e7861d6f0a
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_custom.h
@@ -0,0 +1,329 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+#define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+#define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+#define OPUS_CUSTOM_EXPORT
+#ifdef CELT_C
+#define OPUS_CUSTOM_EXPORT_STATIC static inline
+#else
+#define OPUS_CUSTOM_EXPORT_STATIC
+#endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */
diff --git a/lib/rbcodec/codecs/libopus/opus_decoder.c b/lib/rbcodec/codecs/libopus/opus_decoder.c
new file mode 100644
index 0000000000..7103b183b8
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_decoder.c
@@ -0,0 +1,999 @@
+/* Copyright (c) 2010 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#ifndef OPUS_BUILD
+#error "OPUS_BUILD _MUST_ be defined to build Opus and you probably want a decent config.h, see README for more details."
+#endif
+
+#include <stdarg.h>
+#include "celt.h"
+#include "opus.h"
+#include "entdec.h"
+#include "modes.h"
+#include "API.h"
+#include "stack_alloc.h"
+#include "float_cast.h"
+#include "opus_private.h"
+#include "os_support.h"
+#include "structs.h"
+#include "define.h"
+#include "mathops.h"
+
+struct OpusDecoder {
+   int          celt_dec_offset;
+   int          silk_dec_offset;
+   int          channels;
+   opus_int32   Fs;          /** Sampling rate (at the API level) */
+   silk_DecControlStruct DecControl;
+   int          decode_gain;
+
+   /* Everything beyond this point gets cleared on a reset */
+#define OPUS_DECODER_RESET_START stream_channels
+   int          stream_channels;
+
+   int          bandwidth;
+   int          mode;
+   int          prev_mode;
+   int          frame_size;
+   int          prev_redundancy;
+
+   opus_uint32  rangeFinal;
+};
+
+#ifdef FIXED_POINT
+static inline opus_int16 SAT16(opus_int32 x) {
+   return x > 32767 ? 32767 : x < -32768 ? -32768 : (opus_int16)x;
+}
+#endif
+
+
+int opus_decoder_get_size(int channels)
+{
+   int silkDecSizeBytes, celtDecSizeBytes;
+   int ret;
+   if (channels<1 || channels > 2)
+      return 0;
+   ret = silk_Get_Decoder_Size( &silkDecSizeBytes );
+   if(ret)
+      return 0;
+   silkDecSizeBytes = align(silkDecSizeBytes);
+   celtDecSizeBytes = celt_decoder_get_size(channels);
+   return align(sizeof(OpusDecoder))+silkDecSizeBytes+celtDecSizeBytes;
+}
+
+int opus_decoder_init(OpusDecoder *st, opus_int32 Fs, int channels)
+{
+   void *silk_dec;
+   CELTDecoder *celt_dec;
+   int ret, silkDecSizeBytes;
+
+   if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)
+    || (channels!=1&&channels!=2))
+      return OPUS_BAD_ARG;
+
+   OPUS_CLEAR((char*)st, opus_decoder_get_size(channels));
+   /* Initialize SILK encoder */
+   ret = silk_Get_Decoder_Size(&silkDecSizeBytes);
+   if (ret)
+      return OPUS_INTERNAL_ERROR;
+
+   silkDecSizeBytes = align(silkDecSizeBytes);
+   st->silk_dec_offset = align(sizeof(OpusDecoder));
+   st->celt_dec_offset = st->silk_dec_offset+silkDecSizeBytes;
+   silk_dec = (char*)st+st->silk_dec_offset;
+   celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
+   st->stream_channels = st->channels = channels;
+
+   st->Fs = Fs;
+   st->DecControl.API_sampleRate = st->Fs;
+   st->DecControl.nChannelsAPI      = st->channels;
+
+   /* Reset decoder */
+   ret = silk_InitDecoder( silk_dec );
+   if(ret)return OPUS_INTERNAL_ERROR;
+
+   /* Initialize CELT decoder */
+   ret = celt_decoder_init(celt_dec, Fs, channels);
+   if(ret!=OPUS_OK)return OPUS_INTERNAL_ERROR;
+
+   celt_decoder_ctl(celt_dec, CELT_SET_SIGNALLING(0));
+
+   st->prev_mode = 0;
+   st->frame_size = Fs/400;
+   return OPUS_OK;
+}
+
+OpusDecoder *opus_decoder_create(opus_int32 Fs, int channels, int *error)
+{
+   int ret;
+   OpusDecoder *st;
+   if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)
+    || (channels!=1&&channels!=2))
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+   st = (OpusDecoder *)opus_alloc(opus_decoder_get_size(channels));
+   if (st == NULL)
+   {
+      if (error)
+         *error = OPUS_ALLOC_FAIL;
+      return NULL;
+   }
+   ret = opus_decoder_init(st, Fs, channels);
+   if (error)
+      *error = ret;
+   if (ret != OPUS_OK)
+   {
+      opus_free(st);
+      st = NULL;
+   }
+   return st;
+}
+
+static void smooth_fade(const opus_val16 *in1, const opus_val16 *in2,
+      opus_val16 *out, int overlap, int channels,
+      const opus_val16 *window, opus_int32 Fs)
+{
+   int i, c;
+   int inc = 48000/Fs;
+   for (c=0;c<channels;c++)
+   {
+      for (i=0;i<overlap;i++)
+      {
+         opus_val16 w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+         out[i*channels+c] = SHR32(MAC16_16(MULT16_16(w,in2[i*channels+c]),
+                                   Q15ONE-w, in1[i*channels+c]), 15);
+      }
+   }
+}
+
+static int opus_packet_get_mode(const unsigned char *data)
+{
+   int mode;
+   if (data[0]&0x80)
+   {
+      mode = MODE_CELT_ONLY;
+   } else if ((data[0]&0x60) == 0x60)
+   {
+      mode = MODE_HYBRID;
+   } else {
+      mode = MODE_SILK_ONLY;
+   }
+   return mode;
+}
+
+static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
+{
+   void *silk_dec;
+   CELTDecoder *celt_dec;
+   int i, silk_ret=0, celt_ret=0;
+   ec_dec dec;
+   opus_int32 silk_frame_size;
+   VARDECL(opus_int16, pcm_silk);
+   VARDECL(opus_val16, pcm_transition);
+   VARDECL(opus_val16, redundant_audio);
+
+   int audiosize;
+   int mode;
+   int transition=0;
+   int start_band;
+   int redundancy=0;
+   int redundancy_bytes = 0;
+   int celt_to_silk=0;
+   int c;
+   int F2_5, F5, F10, F20;
+   const opus_val16 *window;
+   opus_uint32 redundant_rng = 0;
+   ALLOC_STACK;
+
+   silk_dec = (char*)st+st->silk_dec_offset;
+   celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
+   F20 = st->Fs/50;
+   F10 = F20>>1;
+   F5 = F10>>1;
+   F2_5 = F5>>1;
+   if (frame_size < F2_5)
+   {
+      RESTORE_STACK;
+      return OPUS_BUFFER_TOO_SMALL;
+   }
+   /* Limit frame_size to avoid excessive stack allocations. */
+   frame_size = IMIN(frame_size, st->Fs/25*3);
+   /* Payloads of 1 (2 including ToC) or 0 trigger the PLC/DTX */
+   if (len<=1)
+   {
+      data = NULL;
+      /* In that case, don't conceal more than what the ToC says */
+      frame_size = IMIN(frame_size, st->frame_size);
+   }
+   if (data != NULL)
+   {
+      audiosize = st->frame_size;
+      mode = st->mode;
+      ec_dec_init(&dec,(unsigned char*)data,len);
+   } else {
+      audiosize = frame_size;
+
+      if (st->prev_mode == 0)
+      {
+         /* If we haven't got any packet yet, all we can do is return zeros */
+         for (i=0;i<audiosize*st->channels;i++)
+            pcm[i] = 0;
+         RESTORE_STACK;
+         return audiosize;
+      } else {
+         mode = st->prev_mode;
+      }
+   }
+
+   /* For CELT/hybrid PLC of more than 20 ms, do multiple calls */
+   if (data==NULL && frame_size > F20 && mode != MODE_SILK_ONLY)
+   {
+      int nb_samples = 0;
+      do {
+         int ret = opus_decode_frame(st, NULL, 0, pcm, F20, 0);
+         if (ret != F20)
+         {
+            RESTORE_STACK;
+            return OPUS_INTERNAL_ERROR;
+         }
+         pcm += F20*st->channels;
+         nb_samples += F20;
+      } while (nb_samples < frame_size);
+      RESTORE_STACK;
+      return frame_size;
+   }
+   ALLOC(pcm_transition, F5*st->channels, opus_val16);
+
+   if (data!=NULL && st->prev_mode > 0 && (
+       (mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY && !st->prev_redundancy)
+    || (mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) )
+      )
+   {
+      transition = 1;
+      if (mode == MODE_CELT_ONLY)
+         opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0);
+   }
+   if (audiosize > frame_size)
+   {
+      /*fprintf(stderr, "PCM buffer too small: %d vs %d (mode = %d)\n", audiosize, frame_size, mode);*/
+      RESTORE_STACK;
+      return OPUS_BAD_ARG;
+   } else {
+      frame_size = audiosize;
+   }
+
+   ALLOC(pcm_silk, IMAX(F10, frame_size)*st->channels, opus_int16);
+   ALLOC(redundant_audio, F5*st->channels, opus_val16);
+
+   /* SILK processing */
+   if (mode != MODE_CELT_ONLY)
+   {
+      int lost_flag, decoded_samples;
+      opus_int16 *pcm_ptr = pcm_silk;
+
+      if (st->prev_mode==MODE_CELT_ONLY)
+         silk_InitDecoder( silk_dec );
+
+      /* The SILK PLC cannot produce frames of less than 10 ms */
+      st->DecControl.payloadSize_ms = IMAX(10, 1000 * audiosize / st->Fs);
+
+      if (data != NULL)
+      {
+        st->DecControl.nChannelsInternal = st->stream_channels;
+        if( mode == MODE_SILK_ONLY ) {
+           if( st->bandwidth == OPUS_BANDWIDTH_NARROWBAND ) {
+              st->DecControl.internalSampleRate = 8000;
+           } else if( st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) {
+              st->DecControl.internalSampleRate = 12000;
+           } else if( st->bandwidth == OPUS_BANDWIDTH_WIDEBAND ) {
+              st->DecControl.internalSampleRate = 16000;
+           } else {
+              st->DecControl.internalSampleRate = 16000;
+              silk_assert( 0 );
+           }
+        } else {
+           /* Hybrid mode */
+           st->DecControl.internalSampleRate = 16000;
+        }
+     }
+
+     lost_flag = data == NULL ? 1 : 2 * decode_fec;
+     decoded_samples = 0;
+     do {
+        /* Call SILK decoder */
+        int first_frame = decoded_samples == 0;
+        silk_ret = silk_Decode( silk_dec, &st->DecControl,
+                                lost_flag, first_frame, &dec, pcm_ptr, &silk_frame_size );
+        if( silk_ret ) {
+           if (lost_flag) {
+              /* PLC failure should not be fatal */
+              silk_frame_size = frame_size;
+              for (i=0;i<frame_size*st->channels;i++)
+                 pcm_ptr[i] = 0;
+           } else {
+             RESTORE_STACK;
+             return OPUS_INVALID_PACKET;
+           }
+        }
+        pcm_ptr += silk_frame_size * st->channels;
+        decoded_samples += silk_frame_size;
+      } while( decoded_samples < frame_size );
+   }
+
+   start_band = 0;
+   if (!decode_fec && mode != MODE_CELT_ONLY && data != NULL
+    && ec_tell(&dec)+17+20*(st->mode == MODE_HYBRID) <= 8*len)
+   {
+      /* Check if we have a redundant 0-8 kHz band */
+      if (mode == MODE_HYBRID)
+         redundancy = ec_dec_bit_logp(&dec, 12);
+      else
+         redundancy = 1;
+      if (redundancy)
+      {
+         celt_to_silk = ec_dec_bit_logp(&dec, 1);
+         /* redundancy_bytes will be at least two, in the non-hybrid
+            case due to the ec_tell() check above */
+         redundancy_bytes = mode==MODE_HYBRID ?
+               (opus_int32)ec_dec_uint(&dec, 256)+2 :
+               len-((ec_tell(&dec)+7)>>3);
+         len -= redundancy_bytes;
+         /* This is a sanity check. It should never happen for a valid
+            packet, so the exact behaviour is not normative. */
+         if (len*8 < ec_tell(&dec))
+         {
+            len = 0;
+            redundancy_bytes = 0;
+            redundancy = 0;
+         }
+         /* Shrink decoder because of raw bits */
+         dec.storage -= redundancy_bytes;
+      }
+   }
+   if (mode != MODE_CELT_ONLY)
+      start_band = 17;
+
+   {
+      int endband=21;
+
+      switch(st->bandwidth)
+      {
+      case OPUS_BANDWIDTH_NARROWBAND:
+         endband = 13;
+         break;
+      case OPUS_BANDWIDTH_MEDIUMBAND:
+      case OPUS_BANDWIDTH_WIDEBAND:
+         endband = 17;
+         break;
+      case OPUS_BANDWIDTH_SUPERWIDEBAND:
+         endband = 19;
+         break;
+      case OPUS_BANDWIDTH_FULLBAND:
+         endband = 21;
+         break;
+      }
+      celt_decoder_ctl(celt_dec, CELT_SET_END_BAND(endband));
+      celt_decoder_ctl(celt_dec, CELT_SET_CHANNELS(st->stream_channels));
+   }
+
+   if (redundancy)
+      transition = 0;
+
+   if (transition && mode != MODE_CELT_ONLY)
+      opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0);
+
+   /* 5 ms redundant frame for CELT->SILK*/
+   if (redundancy && celt_to_silk)
+   {
+      celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
+      celt_decode_with_ec(celt_dec, data+len, redundancy_bytes,
+                          redundant_audio, F5, NULL);
+      celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng));
+   }
+
+   /* MUST be after PLC */
+   celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(start_band));
+
+   if (mode != MODE_SILK_ONLY)
+   {
+      int celt_frame_size = IMIN(F20, frame_size);
+      /* Make sure to discard any previous CELT state */
+      if (mode != st->prev_mode && st->prev_mode > 0 && !st->prev_redundancy)
+         celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
+      /* Decode CELT */
+      celt_ret = celt_decode_with_ec(celt_dec, decode_fec ? NULL : data,
+                                     len, pcm, celt_frame_size, &dec);
+   } else {
+      unsigned char silence[2] = {0xFF, 0xFF};
+      for (i=0;i<frame_size*st->channels;i++)
+         pcm[i] = 0;
+      /* For hybrid -> SILK transitions, we let the CELT MDCT
+         do a fade-out by decoding a silence frame */
+      if (st->prev_mode == MODE_HYBRID && !(redundancy && celt_to_silk && st->prev_redundancy) )
+      {
+         celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
+         celt_decode_with_ec(celt_dec, silence, 2, pcm, F2_5, NULL);
+      }
+   }
+
+   if (mode != MODE_CELT_ONLY)
+   {
+#ifdef FIXED_POINT
+      for (i=0;i<frame_size*st->channels;i++)
+         pcm[i] = SAT16(pcm[i] + pcm_silk[i]);
+#else
+      for (i=0;i<frame_size*st->channels;i++)
+         pcm[i] = pcm[i] + (opus_val16)((1.f/32768.f)*pcm_silk[i]);
+#endif
+   }
+
+   {
+      const CELTMode *celt_mode;
+      celt_decoder_ctl(celt_dec, CELT_GET_MODE(&celt_mode));
+      window = celt_mode->window;
+   }
+
+   /* 5 ms redundant frame for SILK->CELT */
+   if (redundancy && !celt_to_silk)
+   {
+      celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
+      celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
+
+      celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, redundant_audio, F5, NULL);
+      celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng));
+      smooth_fade(pcm+st->channels*(frame_size-F2_5), redundant_audio+st->channels*F2_5,
+                  pcm+st->channels*(frame_size-F2_5), F2_5, st->channels, window, st->Fs);
+   }
+   if (redundancy && celt_to_silk)
+   {
+      for (c=0;c<st->channels;c++)
+      {
+         for (i=0;i<F2_5;i++)
+            pcm[st->channels*i+c] = redundant_audio[st->channels*i+c];
+      }
+      smooth_fade(redundant_audio+st->channels*F2_5, pcm+st->channels*F2_5,
+                  pcm+st->channels*F2_5, F2_5, st->channels, window, st->Fs);
+   }
+   if (transition)
+   {
+      if (audiosize >= F5)
+      {
+         for (i=0;i<st->channels*F2_5;i++)
+            pcm[i] = pcm_transition[i];
+         smooth_fade(pcm_transition+st->channels*F2_5, pcm+st->channels*F2_5,
+                     pcm+st->channels*F2_5, F2_5,
+                     st->channels, window, st->Fs);
+      } else {
+         /* Not enough time to do a clean transition, but we do it anyway
+            This will not preserve amplitude perfectly and may introduce
+            a bit of temporal aliasing, but it shouldn't be too bad and
+            that's pretty much the best we can do. In any case, generating this
+            transition it pretty silly in the first place */
+         smooth_fade(pcm_transition, pcm,
+                     pcm, F2_5,
+                     st->channels, window, st->Fs);
+      }
+   }
+
+   if(st->decode_gain)
+   {
+      opus_val32 gain;
+      gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain));
+      for (i=0;i<frame_size*st->channels;i++)
+      {
+         opus_val32 x;
+         x = MULT16_32_P16(pcm[i],gain);
+         pcm[i] = SATURATE(x, 32767);
+      }
+   }
+
+   if (len <= 1)
+      st->rangeFinal = 0;
+   else
+      st->rangeFinal = dec.rng ^ redundant_rng;
+
+   st->prev_mode = mode;
+   st->prev_redundancy = redundancy && !celt_to_silk;
+   RESTORE_STACK;
+   return celt_ret < 0 ? celt_ret : audiosize;
+
+}
+
+static int parse_size(const unsigned char *data, opus_int32 len, short *size)
+{
+   if (len<1)
+   {
+      *size = -1;
+      return -1;
+   } else if (data[0]<252)
+   {
+      *size = data[0];
+      return 1;
+   } else if (len<2)
+   {
+      *size = -1;
+      return -1;
+   } else {
+      *size = 4*data[1] + data[0];
+      return 2;
+   }
+}
+
+static int opus_packet_parse_impl(const unsigned char *data, opus_int32 len,
+      int self_delimited, unsigned char *out_toc,
+      const unsigned char *frames[48], short size[48], int *payload_offset)
+{
+   int i, bytes;
+   int count;
+   int cbr;
+   unsigned char ch, toc;
+   int framesize;
+   int last_size;
+   const unsigned char *data0 = data;
+
+   if (size==NULL)
+      return OPUS_BAD_ARG;
+
+   framesize = opus_packet_get_samples_per_frame(data, 48000);
+
+   cbr = 0;
+   toc = *data++;
+   len--;
+   last_size = len;
+   switch (toc&0x3)
+   {
+   /* One frame */
+   case 0:
+      count=1;
+      break;
+   /* Two CBR frames */
+   case 1:
+      count=2;
+      cbr = 1;
+      if (!self_delimited)
+      {
+         if (len&0x1)
+            return OPUS_INVALID_PACKET;
+         size[0] = last_size = len/2;
+      }
+      break;
+   /* Two VBR frames */
+   case 2:
+      count = 2;
+      bytes = parse_size(data, len, size);
+      len -= bytes;
+      if (size[0]<0 || size[0] > len)
+         return OPUS_INVALID_PACKET;
+      data += bytes;
+      last_size = len-size[0];
+      break;
+   /* Multiple CBR/VBR frames (from 0 to 120 ms) */
+   default: /*case 3:*/
+      if (len<1)
+         return OPUS_INVALID_PACKET;
+      /* Number of frames encoded in bits 0 to 5 */
+      ch = *data++;
+      count = ch&0x3F;
+      if (count <= 0 || framesize*count > 5760)
+         return OPUS_INVALID_PACKET;
+      len--;
+      /* Padding flag is bit 6 */
+      if (ch&0x40)
+      {
+         int padding=0;
+         int p;
+         do {
+            if (len<=0)
+               return OPUS_INVALID_PACKET;
+            p = *data++;
+            len--;
+            padding += p==255 ? 254: p;
+         } while (p==255);
+         len -= padding;
+      }
+      if (len<0)
+         return OPUS_INVALID_PACKET;
+      /* VBR flag is bit 7 */
+      cbr = !(ch&0x80);
+      if (!cbr)
+      {
+         /* VBR case */
+         last_size = len;
+         for (i=0;i<count-1;i++)
+         {
+            bytes = parse_size(data, len, size+i);
+            len -= bytes;
+            if (size[i]<0 || size[i] > len)
+               return OPUS_INVALID_PACKET;
+            data += bytes;
+            last_size -= bytes+size[i];
+         }
+         if (last_size<0)
+            return OPUS_INVALID_PACKET;
+      } else if (!self_delimited)
+      {
+         /* CBR case */
+         last_size = len/count;
+         if (last_size*count!=len)
+            return OPUS_INVALID_PACKET;
+         for (i=0;i<count-1;i++)
+            size[i] = last_size;
+      }
+      break;
+   }
+   /* Self-delimited framing has an extra size for the last frame. */
+   if (self_delimited)
+   {
+      bytes = parse_size(data, len, size+count-1);
+      len -= bytes;
+      if (size[count-1]<0 || size[count-1] > len)
+         return OPUS_INVALID_PACKET;
+      data += bytes;
+      /* For CBR packets, apply the size to all the frames. */
+      if (cbr)
+      {
+         if (size[count-1]*count > len)
+            return OPUS_INVALID_PACKET;
+         for (i=0;i<count-1;i++)
+            size[i] = size[count-1];
+      } else if(size[count-1] > last_size)
+         return OPUS_INVALID_PACKET;
+   } else
+   {
+      /* Because it's not encoded explicitly, it's possible the size of the
+         last packet (or all the packets, for the CBR case) is larger than
+         1275. Reject them here.*/
+      if (last_size > 1275)
+         return OPUS_INVALID_PACKET;
+      size[count-1] = last_size;
+   }
+
+   if (frames)
+   {
+      for (i=0;i<count;i++)
+      {
+         frames[i] = data;
+         data += size[i];
+      }
+   }
+
+   if (out_toc)
+      *out_toc = toc;
+
+   if (payload_offset)
+      *payload_offset = data-data0;
+
+   return count;
+}
+
+int opus_packet_parse(const unsigned char *data, opus_int32 len,
+      unsigned char *out_toc, const unsigned char *frames[48],
+      short size[48], int *payload_offset)
+{
+   return opus_packet_parse_impl(data, len, 0, out_toc,
+                                 frames, size, payload_offset);
+}
+
+int opus_decode_native(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec,
+      int self_delimited, int *packet_offset)
+{
+   int i, nb_samples;
+   int count, offset;
+   unsigned char toc;
+   int tot_offset;
+   /* 48 x 2.5 ms = 120 ms */
+   short size[48];
+   if (decode_fec<0 || decode_fec>1)
+      return OPUS_BAD_ARG;
+   if (len==0 || data==NULL)
+      return opus_decode_frame(st, NULL, 0, pcm, frame_size, 0);
+   else if (len<0)
+      return OPUS_BAD_ARG;
+
+   tot_offset = 0;
+   st->mode = opus_packet_get_mode(data);
+   st->bandwidth = opus_packet_get_bandwidth(data);
+   st->frame_size = opus_packet_get_samples_per_frame(data, st->Fs);
+   st->stream_channels = opus_packet_get_nb_channels(data);
+
+   count = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL, size, &offset);
+   if (count < 0)
+      return count;
+
+   data += offset;
+   tot_offset += offset;
+
+   if (count*st->frame_size > frame_size)
+      return OPUS_BUFFER_TOO_SMALL;
+   nb_samples=0;
+   for (i=0;i<count;i++)
+   {
+      int ret;
+      ret = opus_decode_frame(st, data, size[i], pcm, frame_size-nb_samples, decode_fec);
+      if (ret<0)
+         return ret;
+      data += size[i];
+      tot_offset += size[i];
+      pcm += ret*st->channels;
+      nb_samples += ret;
+   }
+   if (packet_offset != NULL)
+      *packet_offset = tot_offset;
+   return nb_samples;
+}
+
+#ifdef FIXED_POINT
+
+int opus_decode(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
+{
+   return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_decode_float(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, float *pcm, int frame_size, int decode_fec)
+{
+   VARDECL(opus_int16, out);
+   int ret, i;
+   ALLOC_STACK;
+
+   ALLOC(out, frame_size*st->channels, opus_int16);
+
+   ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL);
+   if (ret > 0)
+   {
+      for (i=0;i<ret*st->channels;i++)
+         pcm[i] = (1.f/32768.f)*(out[i]);
+   }
+   RESTORE_STACK;
+   return ret;
+}
+#endif
+
+
+#else
+int opus_decode(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec)
+{
+   VARDECL(float, out);
+   int ret, i;
+   ALLOC_STACK;
+
+   if(frame_size<0)
+   {
+      RESTORE_STACK;
+      return OPUS_BAD_ARG;
+   }
+
+   ALLOC(out, frame_size*st->channels, float);
+
+   ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL);
+   if (ret > 0)
+   {
+      for (i=0;i<ret*st->channels;i++)
+         pcm[i] = FLOAT2INT16(out[i]);
+   }
+   RESTORE_STACK;
+   return ret;
+}
+
+int opus_decode_float(OpusDecoder *st, const unsigned char *data,
+      opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
+{
+   return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL);
+}
+
+#endif
+
+int opus_decoder_ctl(OpusDecoder *st, int request, ...)
+{
+   int ret = OPUS_OK;
+   va_list ap;
+   void *silk_dec;
+   CELTDecoder *celt_dec;
+
+   silk_dec = (char*)st+st->silk_dec_offset;
+   celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
+
+
+   va_start(ap, request);
+
+   switch (request)
+   {
+   case OPUS_GET_BANDWIDTH_REQUEST:
+   {
+      opus_int32 *value = va_arg(ap, opus_int32*);
+      *value = st->bandwidth;
+   }
+   break;
+   case OPUS_GET_FINAL_RANGE_REQUEST:
+   {
+      opus_uint32 *value = va_arg(ap, opus_uint32*);
+      *value = st->rangeFinal;
+   }
+   break;
+   case OPUS_RESET_STATE:
+   {
+      OPUS_CLEAR((char*)&st->OPUS_DECODER_RESET_START,
+            sizeof(OpusDecoder)-
+            ((char*)&st->OPUS_DECODER_RESET_START - (char*)st));
+
+      celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
+      silk_InitDecoder( silk_dec );
+      st->stream_channels = st->channels;
+      st->frame_size = st->Fs/400;
+   }
+   break;
+   case OPUS_GET_SAMPLE_RATE_REQUEST:
+   {
+      opus_int32 *value = va_arg(ap, opus_int32*);
+      if (value==NULL)
+      {
+         ret = OPUS_BAD_ARG;
+         break;
+      }
+      *value = st->Fs;
+   }
+   break;
+   case OPUS_GET_PITCH_REQUEST:
+   {
+      opus_int32 *value = va_arg(ap, opus_int32*);
+      if (value==NULL)
+      {
+         ret = OPUS_BAD_ARG;
+         break;
+      }
+      if (st->prev_mode == MODE_CELT_ONLY)
+         celt_decoder_ctl(celt_dec, OPUS_GET_PITCH(value));
+      else
+         *value = st->DecControl.prevPitchLag;
+   }
+   break;
+   case OPUS_GET_GAIN_REQUEST:
+   {
+      opus_int32 *value = va_arg(ap, opus_int32*);
+      if (value==NULL)
+      {
+         ret = OPUS_BAD_ARG;
+         break;
+      }
+      *value = st->decode_gain;
+   }
+   break;
+   case OPUS_SET_GAIN_REQUEST:
+   {
+       opus_int32 value = va_arg(ap, opus_int32);
+       if (value<-32768 || value>32767)
+       {
+          ret = OPUS_BAD_ARG;
+          break;
+       }
+       st->decode_gain = value;
+   }
+   break;
+   default:
+      /*fprintf(stderr, "unknown opus_decoder_ctl() request: %d", request);*/
+      ret = OPUS_UNIMPLEMENTED;
+      break;
+   }
+
+   va_end(ap);
+   return ret;
+}
+
+void opus_decoder_destroy(OpusDecoder *st)
+{
+   opus_free(st);
+}
+
+
+int opus_packet_get_bandwidth(const unsigned char *data)
+{
+   int bandwidth;
+   if (data[0]&0x80)
+   {
+      bandwidth = OPUS_BANDWIDTH_MEDIUMBAND + ((data[0]>>5)&0x3);
+      if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+         bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+   } else if ((data[0]&0x60) == 0x60)
+   {
+      bandwidth = (data[0]&0x10) ? OPUS_BANDWIDTH_FULLBAND :
+                                   OPUS_BANDWIDTH_SUPERWIDEBAND;
+   } else {
+      bandwidth = OPUS_BANDWIDTH_NARROWBAND + ((data[0]>>5)&0x3);
+   }
+   return bandwidth;
+}
+
+int opus_packet_get_samples_per_frame(const unsigned char *data,
+      opus_int32 Fs)
+{
+   int audiosize;
+   if (data[0]&0x80)
+   {
+      audiosize = ((data[0]>>3)&0x3);
+      audiosize = (Fs<<audiosize)/400;
+   } else if ((data[0]&0x60) == 0x60)
+   {
+      audiosize = (data[0]&0x08) ? Fs/50 : Fs/100;
+   } else {
+      audiosize = ((data[0]>>3)&0x3);
+      if (audiosize == 3)
+         audiosize = Fs*60/1000;
+      else
+         audiosize = (Fs<<audiosize)/100;
+   }
+   return audiosize;
+}
+
+int opus_packet_get_nb_channels(const unsigned char *data)
+{
+   return (data[0]&0x4) ? 2 : 1;
+}
+
+int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len)
+{
+   int count;
+   if (len<1)
+      return OPUS_BAD_ARG;
+   count = packet[0]&0x3;
+   if (count==0)
+      return 1;
+   else if (count!=3)
+      return 2;
+   else if (len<2)
+      return OPUS_INVALID_PACKET;
+   else
+      return packet[1]&0x3F;
+}
+
+int opus_decoder_get_nb_samples(const OpusDecoder *dec,
+      const unsigned char packet[], opus_int32 len)
+{
+   int samples;
+   int count = opus_packet_get_nb_frames(packet, len);
+
+   if (count<0)
+      return count;
+
+   samples = count*opus_packet_get_samples_per_frame(packet, dec->Fs);
+   /* Can't have more than 120 ms */
+   if (samples*25 > dec->Fs*3)
+      return OPUS_INVALID_PACKET;
+   else
+      return samples;
+}
diff --git a/lib/rbcodec/codecs/libopus/opus_defines.h b/lib/rbcodec/codecs/libopus/opus_defines.h
new file mode 100644
index 0000000000..830d225f14
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_defines.h
@@ -0,0 +1,644 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** The mode struct passed is invalid @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#if defined(__GNUC__) && defined(OPUS_BUILD)
+# define OPUS_EXPORT __attribute__ ((visibility ("default")))
+#elif defined(WIN32) && !defined(__MINGW32__)
+# ifdef OPUS_BUILD
+#   define OPUS_EXPORT __declspec(dllexport)
+# else
+#   define OPUS_EXPORT
+# endif
+#else
+# define OPUS_EXPORT
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @warning Only the MDCT mode of Opus can provide hard CBR behavior.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values with trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the lossless case, but greater quality
+  * under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the depth of signal being encoded.
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */
diff --git a/lib/rbcodec/codecs/libopus/opus_header.c b/lib/rbcodec/codecs/libopus/opus_header.c
new file mode 100644
index 0000000000..ed07c9ab50
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_header.c
@@ -0,0 +1,286 @@
+/* Copyright (C)2012 Xiph.Org Foundation
+   File: opus_header.c
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+# include "opus_config.h"
+#endif
+
+#include "opus_header.h"
+#include <string.h>
+#include <stdio.h>
+
+/* Header contents:
+  - "OpusHead" (64 bits)
+  - version number (8 bits)
+  - Channels C (8 bits)
+  - Pre-skip (16 bits)
+  - Sampling rate (32 bits)
+  - Gain in dB (16 bits, S7.8)
+  - Mapping (8 bits, 0=single stream (mono/stereo) 1=Vorbis mapping,
+             2..254: reserved, 255: multistream with no mapping)
+
+  - if (mapping != 0)
+     - N = totel number of streams (8 bits)
+     - M = number of paired streams (8 bits)
+     - C times channel origin
+          - if (C<2*M)
+             - stream = byte/2
+             - if (byte&0x1 == 0)
+                 - left
+               else
+                 - right
+          - else
+             - stream = byte-M
+*/
+
+typedef struct {
+   unsigned char *data;
+   int maxlen;
+   int pos;
+} Packet;
+
+typedef struct {
+   const unsigned char *data;
+   int maxlen;
+   int pos;
+} ROPacket;
+
+static int write_uint32(Packet *p, ogg_uint32_t val)
+{
+   if (p->pos>p->maxlen-4)
+      return 0;
+   p->data[p->pos  ] = (val    ) & 0xFF;
+   p->data[p->pos+1] = (val>> 8) & 0xFF;
+   p->data[p->pos+2] = (val>>16) & 0xFF;
+   p->data[p->pos+3] = (val>>24) & 0xFF;
+   p->pos += 4;
+   return 1;
+}
+
+static int write_uint16(Packet *p, ogg_uint16_t val)
+{
+   if (p->pos>p->maxlen-2)
+      return 0;
+   p->data[p->pos  ] = (val    ) & 0xFF;
+   p->data[p->pos+1] = (val>> 8) & 0xFF;
+   p->pos += 2;
+   return 1;
+}
+
+static int write_chars(Packet *p, const unsigned char *str, int nb_chars)
+{
+   int i;
+   if (p->pos>p->maxlen-nb_chars)
+      return 0;
+   for (i=0;i<nb_chars;i++)
+      p->data[p->pos++] = str[i];
+   return 1;
+}
+
+static int read_uint32(ROPacket *p, ogg_uint32_t *val)
+{
+   if (p->pos>p->maxlen-4)
+      return 0;
+   *val =  (ogg_uint32_t)p->data[p->pos  ];
+   *val |= (ogg_uint32_t)p->data[p->pos+1]<< 8;
+   *val |= (ogg_uint32_t)p->data[p->pos+2]<<16;
+   *val |= (ogg_uint32_t)p->data[p->pos+3]<<24;
+   p->pos += 4;
+   return 1;
+}
+
+static int read_uint16(ROPacket *p, ogg_uint16_t *val)
+{
+   if (p->pos>p->maxlen-2)
+      return 0;
+   *val =  (ogg_uint16_t)p->data[p->pos  ];
+   *val |= (ogg_uint16_t)p->data[p->pos+1]<<8;
+   p->pos += 2;
+   return 1;
+}
+
+static int read_chars(ROPacket *p, unsigned char *str, int nb_chars)
+{
+   int i;
+   if (p->pos>p->maxlen-nb_chars)
+      return 0;
+   for (i=0;i<nb_chars;i++)
+      str[i] = p->data[p->pos++];
+   return 1;
+}
+
+int opus_header_parse(const unsigned char *packet, int len, OpusHeader *h)
+{
+   int i;
+   char str[9];
+   ROPacket p;
+   unsigned char ch;
+   ogg_uint16_t shortval;
+
+   p.data = packet;
+   p.maxlen = len;
+   p.pos = 0;
+   str[8] = 0;
+   if (len<19)return 0;
+   read_chars(&p, (unsigned char*)str, 8);
+   if (memcmp(str, "OpusHead", 8)!=0)
+      return 0;
+
+   if (!read_chars(&p, &ch, 1))
+      return 0;
+   h->version = ch;
+   if((h->version&240) != 0) /* Only major version 0 supported. */
+      return 0;
+
+   if (!read_chars(&p, &ch, 1))
+      return 0;
+   h->channels = ch;
+   if (h->channels == 0)
+      return 0;
+
+   if (!read_uint16(&p, &shortval))
+      return 0;
+   h->preskip = shortval;
+
+   if (!read_uint32(&p, &h->input_sample_rate))
+      return 0;
+
+   if (!read_uint16(&p, &shortval))
+      return 0;
+   h->gain = (short)shortval;
+
+   if (!read_chars(&p, &ch, 1))
+      return 0;
+   h->channel_mapping = ch;
+
+   if (h->channel_mapping != 0)
+   {
+      if (!read_chars(&p, &ch, 1))
+         return 0;
+
+      if (ch<1)
+         return 0;
+      h->nb_streams = ch;
+
+      if (!read_chars(&p, &ch, 1))
+         return 0;
+
+      if (ch>h->nb_streams || (ch+h->nb_streams)>255)
+         return 0;
+      h->nb_coupled = ch;
+
+      /* Multi-stream support */
+      for (i=0;i<h->channels;i++)
+      {
+         if (!read_chars(&p, &h->stream_map[i], 1))
+            return 0;
+         if (h->stream_map[i]>(h->nb_streams+h->nb_coupled) && h->stream_map[i]!=255)
+            return 0;
+      }
+   } else {
+      if(h->channels>2)
+         return 0;
+      h->nb_streams = 1;
+      h->nb_coupled = h->channels>1;
+      h->stream_map[0]=0;
+      h->stream_map[1]=1;
+   }
+   /*For version 0/1 we know there won't be any more data
+     so reject any that have data past the end.*/
+   if ((h->version==0 || h->version==1) && p.pos != len)
+      return 0;
+   return 1;
+}
+
+int opus_header_to_packet(const OpusHeader *h, unsigned char *packet, int len)
+{
+   int i;
+   Packet p;
+   unsigned char ch;
+
+   p.data = packet;
+   p.maxlen = len;
+   p.pos = 0;
+   if (len<19)return 0;
+   if (!write_chars(&p, (const unsigned char*)"OpusHead", 8))
+      return 0;
+   /* Version is 1 */
+   ch = 1;
+   if (!write_chars(&p, &ch, 1))
+      return 0;
+
+   ch = h->channels;
+   if (!write_chars(&p, &ch, 1))
+      return 0;
+
+   if (!write_uint16(&p, h->preskip))
+      return 0;
+
+   if (!write_uint32(&p, h->input_sample_rate))
+      return 0;
+
+   if (!write_uint16(&p, h->gain))
+      return 0;
+
+   ch = h->channel_mapping;
+   if (!write_chars(&p, &ch, 1))
+      return 0;
+
+   if (h->channel_mapping != 0)
+   {
+      ch = h->nb_streams;
+      if (!write_chars(&p, &ch, 1))
+         return 0;
+
+      ch = h->nb_coupled;
+      if (!write_chars(&p, &ch, 1))
+         return 0;
+
+      /* Multi-stream support */
+      for (i=0;i<h->channels;i++)
+      {
+         if (!write_chars(&p, &h->stream_map[i], 1))
+            return 0;
+      }
+   }
+
+   return p.pos;
+}
+
+/* This is just here because it's a convenient file linked by both opusenc and
+   opusdec (to guarantee this maps stays in sync). */
+const int wav_permute_matrix[8][8] =
+{
+  {0},              /* 1.0 mono   */
+  {0,1},            /* 2.0 stereo */
+  {0,2,1},          /* 3.0 channel ('wide') stereo */
+  {0,1,2,3},        /* 4.0 discrete quadraphonic */
+  {0,2,1,3,4},      /* 5.0 surround */
+  {0,2,1,4,5,3},    /* 5.1 surround */
+  {0,2,1,5,6,4,3},  /* 6.1 surround */
+  {0,2,1,6,7,4,5,3} /* 7.1 surround (classic theater 8-track) */
+};
diff --git a/lib/rbcodec/codecs/libopus/opus_header.h b/lib/rbcodec/codecs/libopus/opus_header.h
new file mode 100644
index 0000000000..7bfacfe48c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_header.h
@@ -0,0 +1,51 @@
+/* Copyright (C)2012 Xiph.Org Foundation
+   File: opus_header.h
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef OPUS_HEADER_H
+#define OPUS_HEADER_H
+
+#include "ogg/ogg.h"
+
+typedef struct {
+   int version;
+   int channels; /* Number of channels: 1..255 */
+   int preskip;
+   ogg_uint32_t input_sample_rate;
+   int gain; /* in dB S7.8 should be zero whenever possible */
+   int channel_mapping;
+   /* The rest is only used if channel_mapping != 0 */
+   int nb_streams;
+   int nb_coupled;
+   unsigned char stream_map[255];
+} OpusHeader;
+
+int opus_header_parse(const unsigned char *header, int len, OpusHeader *h);
+int opus_header_to_packet(const OpusHeader *h, unsigned char *packet, int len);
+
+extern const int wav_permute_matrix[8][8];
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/opus_private.h b/lib/rbcodec/codecs/libopus/opus_private.h
new file mode 100644
index 0000000000..52482bc18c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_private.h
@@ -0,0 +1,85 @@
+/* Copyright (c) 2012 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef OPUS_PRIVATE_H
+#define OPUS_PRIVATE_H
+
+#include "arch.h"
+#include "opus.h"
+
+struct OpusRepacketizer {
+   unsigned char toc;
+   int nb_frames;
+   const unsigned char *frames[48];
+   short len[48];
+   int framesize;
+};
+
+
+#define MODE_SILK_ONLY          1000
+#define MODE_HYBRID             1001
+#define MODE_CELT_ONLY          1002
+
+#define OPUS_SET_VOICE_RATIO_REQUEST         11018
+#define OPUS_GET_VOICE_RATIO_REQUEST         11019
+
+/** Configures the encoder's expected percentage of voice
+  * opposed to music or other signals.
+  *
+  * @note This interface is currently more aspiration than actuality. It's
+  * ultimately expected to bias an automatic signal classifier, but it currently
+  * just shifts the static bitrate to mode mapping around a little bit.
+  *
+  * @param[in] x <tt>int</tt>:   Voice percentage in the range 0-100, inclusive.
+  * @hideinitializer */
+#define OPUS_SET_VOICE_RATIO(x) OPUS_SET_VOICE_RATIO_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured voice ratio value, @see OPUS_SET_VOICE_RATIO
+  *
+  * @param[out] x <tt>int*</tt>:  Voice percentage in the range 0-100, inclusive.
+  * @hideinitializer */
+#define OPUS_GET_VOICE_RATIO(x) OPUS_GET_VOICE_RATIO_REQUEST, __opus_check_int_ptr(x)
+
+
+#define OPUS_SET_FORCE_MODE_REQUEST    11002
+#define OPUS_SET_FORCE_MODE(x) OPUS_SET_FORCE_MODE_REQUEST, __opus_check_int(x)
+
+
+int encode_size(int size, unsigned char *data);
+
+int opus_decode_native(OpusDecoder *st, const unsigned char *data, opus_int32 len,
+      opus_val16 *pcm, int frame_size, int decode_fec, int self_delimited, int *packet_offset);
+
+/* Make sure everything's aligned to sizeof(void *) bytes */
+static inline int align(int i)
+{
+    return (i+sizeof(void *)-1)&-((int)sizeof(void *));
+}
+
+opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen, int self_delimited);
+
+#endif /* OPUS_PRIVATE_H */
diff --git a/lib/rbcodec/codecs/libopus/opus_types.h b/lib/rbcodec/codecs/libopus/opus_types.h
new file mode 100644
index 0000000000..b28e03aea2
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/opus_types.h
@@ -0,0 +1,159 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+#endif  /* OPUS_TYPES_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/API.h b/lib/rbcodec/codecs/libopus/silk/API.h
new file mode 100644
index 0000000000..4b8ca12ac6
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/API.h
@@ -0,0 +1,132 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_API_H
+#define SILK_API_H
+
+#include "control.h"
+#include "typedef.h"
+#include "errors.h"
+#include "entenc.h"
+#include "entdec.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+#define SILK_MAX_FRAMES_PER_PACKET  3
+
+/* Struct for TOC (Table of Contents) */
+typedef struct {
+    opus_int    VADFlag;                                /* Voice activity for packet                            */
+    opus_int    VADFlags[ SILK_MAX_FRAMES_PER_PACKET ]; /* Voice activity for each frame in packet              */
+    opus_int    inbandFECFlag;                          /* Flag indicating if packet contains in-band FEC       */
+} silk_TOC_struct;
+
+/****************************************/
+/* Encoder functions                    */
+/****************************************/
+
+/***********************************************/
+/* Get size in bytes of the Silk encoder state */
+/***********************************************/
+opus_int silk_Get_Encoder_Size(                         /* O    Returns error code                              */
+    opus_int                        *encSizeBytes       /* O    Number of bytes in SILK encoder state           */
+);
+
+/*************************/
+/* Init or reset encoder */
+/*************************/
+opus_int silk_InitEncoder(                              /* O    Returns error code                              */
+    void                            *encState,          /* I/O  State                                           */
+    silk_EncControlStruct           *encStatus          /* O    Encoder Status                                  */
+);
+
+/**************************/
+/* Encode frame with Silk */
+/**************************/
+/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what                     */
+/* encControl->payloadSize_ms is set to                                                                         */
+opus_int silk_Encode(                                   /* O    Returns error code                              */
+    void                            *encState,          /* I/O  State                                           */
+    silk_EncControlStruct           *encControl,        /* I    Control status                                  */
+    const opus_int16                *samplesIn,         /* I    Speech sample input vector                      */
+    opus_int                        nSamplesIn,         /* I    Number of samples in input vector               */
+    ec_enc                          *psRangeEnc,        /* I/O  Compressor data structure                       */
+    opus_int32                      *nBytesOut,         /* I/O  Number of bytes in payload (input: Max bytes)   */
+    const opus_int                  prefillFlag         /* I    Flag to indicate prefilling buffers no coding   */
+);
+
+/****************************************/
+/* Decoder functions                    */
+/****************************************/
+
+/***********************************************/
+/* Get size in bytes of the Silk decoder state */
+/***********************************************/
+opus_int silk_Get_Decoder_Size(                         /* O    Returns error code                              */
+    opus_int                        *decSizeBytes       /* O    Number of bytes in SILK decoder state           */
+);
+
+/*************************/
+/* Init or Reset decoder */
+/*************************/
+opus_int silk_InitDecoder(                              /* O    Returns error code                              */
+    void                            *decState           /* I/O  State                                           */
+);
+
+/******************/
+/* Decode a frame */
+/******************/
+opus_int silk_Decode(                                   /* O    Returns error code                              */
+    void*                           decState,           /* I/O  State                                           */
+    silk_DecControlStruct*          decControl,         /* I/O  Control Structure                               */
+    opus_int                        lostFlag,           /* I    0: no loss, 1 loss, 2 decode fec                */
+    opus_int                        newPacketFlag,      /* I    Indicates first decoder call for this packet    */
+    ec_dec                          *psRangeDec,        /* I/O  Compressor data structure                       */
+    opus_int16                      *samplesOut,        /* O    Decoded output speech vector                    */
+    opus_int32                      *nSamplesOut        /* O    Number of samples decoded                       */
+);
+
+#if 0
+/**************************************/
+/* Get table of contents for a packet */
+/**************************************/
+opus_int silk_get_TOC(
+    const opus_uint8                *payload,           /* I    Payload data                                */
+    const opus_int                  nBytesIn,           /* I    Number of input bytes                       */
+    const opus_int                  nFramesPerPayload,  /* I    Number of SILK frames per payload           */
+    silk_TOC_struct                 *Silk_TOC           /* O    Type of content                             */
+);
+#endif
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/CNG.c b/lib/rbcodec/codecs/libopus/silk/CNG.c
new file mode 100644
index 0000000000..fc19391057
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/CNG.c
@@ -0,0 +1,167 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Generates excitation for CNG LPC synthesis */
+static inline void silk_CNG_exc(
+    opus_int32                       residual_Q10[],     /* O    CNG residual signal Q10                     */
+    opus_int32                       exc_buf_Q14[],      /* I    Random samples buffer Q10                   */
+    opus_int32                       Gain_Q16,           /* I    Gain to apply                               */
+    opus_int                         length,             /* I    Length                                      */
+    opus_int32                       *rand_seed          /* I/O  Seed to random index generator              */
+)
+{
+    opus_int32 seed;
+    opus_int   i, idx, exc_mask;
+
+    exc_mask = CNG_BUF_MASK_MAX;
+    while( exc_mask > length ) {
+        exc_mask = silk_RSHIFT( exc_mask, 1 );
+    }
+
+    seed = *rand_seed;
+    for( i = 0; i < length; i++ ) {
+        seed = silk_RAND( seed );
+        idx = (opus_int)( silk_RSHIFT( seed, 24 ) & exc_mask );
+        silk_assert( idx >= 0 );
+        silk_assert( idx <= CNG_BUF_MASK_MAX );
+        residual_Q10[ i ] = (opus_int16)silk_SAT16( silk_SMULWW( exc_buf_Q14[ idx ], Gain_Q16 >> 4 ) );
+    }
+    *rand_seed = seed;
+}
+
+void silk_CNG_Reset(
+    silk_decoder_state          *psDec                          /* I/O  Decoder state                               */
+)
+{
+    opus_int i, NLSF_step_Q15, NLSF_acc_Q15;
+
+    NLSF_step_Q15 = silk_DIV32_16( silk_int16_MAX, psDec->LPC_order + 1 );
+    NLSF_acc_Q15 = 0;
+    for( i = 0; i < psDec->LPC_order; i++ ) {
+        NLSF_acc_Q15 += NLSF_step_Q15;
+        psDec->sCNG.CNG_smth_NLSF_Q15[ i ] = NLSF_acc_Q15;
+    }
+    psDec->sCNG.CNG_smth_Gain_Q16 = 0;
+    psDec->sCNG.rand_seed = 3176576;
+}
+
+/* Updates CNG estimate, and applies the CNG when packet was lost   */
+void silk_CNG(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state                               */
+    silk_decoder_control        *psDecCtrl,                     /* I/O  Decoder control                             */
+    opus_int16                  frame[],                        /* I/O  Signal                                      */
+    opus_int                    length                          /* I    Length of residual                          */
+)
+{
+    opus_int   i, subfr;
+    opus_int32 sum_Q6, max_Gain_Q16;
+    opus_int16 A_Q12[ MAX_LPC_ORDER ];
+    opus_int32 CNG_sig_Q10[ MAX_FRAME_LENGTH + MAX_LPC_ORDER ];
+    silk_CNG_struct *psCNG = &psDec->sCNG;
+
+    if( psDec->fs_kHz != psCNG->fs_kHz ) {
+        /* Reset state */
+        silk_CNG_Reset( psDec );
+
+        psCNG->fs_kHz = psDec->fs_kHz;
+    }
+    if( psDec->lossCnt == 0 && psDec->prevSignalType == TYPE_NO_VOICE_ACTIVITY ) {
+        /* Update CNG parameters */
+
+        /* Smoothing of LSF's  */
+        for( i = 0; i < psDec->LPC_order; i++ ) {
+            psCNG->CNG_smth_NLSF_Q15[ i ] += silk_SMULWB( (opus_int32)psDec->prevNLSF_Q15[ i ] - (opus_int32)psCNG->CNG_smth_NLSF_Q15[ i ], CNG_NLSF_SMTH_Q16 );
+        }
+        /* Find the subframe with the highest gain */
+        max_Gain_Q16 = 0;
+        subfr        = 0;
+        for( i = 0; i < psDec->nb_subfr; i++ ) {
+            if( psDecCtrl->Gains_Q16[ i ] > max_Gain_Q16 ) {
+                max_Gain_Q16 = psDecCtrl->Gains_Q16[ i ];
+                subfr        = i;
+            }
+        }
+        /* Update CNG excitation buffer with excitation from this subframe */
+        silk_memmove( &psCNG->CNG_exc_buf_Q14[ psDec->subfr_length ], psCNG->CNG_exc_buf_Q14, ( psDec->nb_subfr - 1 ) * psDec->subfr_length * sizeof( opus_int32 ) );
+        silk_memcpy(   psCNG->CNG_exc_buf_Q14, &psDec->exc_Q14[ subfr * psDec->subfr_length ], psDec->subfr_length * sizeof( opus_int32 ) );
+
+        /* Smooth gains */
+        for( i = 0; i < psDec->nb_subfr; i++ ) {
+            psCNG->CNG_smth_Gain_Q16 += silk_SMULWB( psDecCtrl->Gains_Q16[ i ] - psCNG->CNG_smth_Gain_Q16, CNG_GAIN_SMTH_Q16 );
+        }
+    }
+
+    /* Add CNG when packet is lost or during DTX */
+    if( psDec->lossCnt ) {
+
+        /* Generate CNG excitation */
+        silk_CNG_exc( CNG_sig_Q10 + MAX_LPC_ORDER, psCNG->CNG_exc_buf_Q14, psCNG->CNG_smth_Gain_Q16, length, &psCNG->rand_seed );
+
+        /* Convert CNG NLSF to filter representation */
+        silk_NLSF2A( A_Q12, psCNG->CNG_smth_NLSF_Q15, psDec->LPC_order );
+
+        /* Generate CNG signal, by synthesis filtering */
+        silk_memcpy( CNG_sig_Q10, psCNG->CNG_synth_state, MAX_LPC_ORDER * sizeof( opus_int32 ) );
+        for( i = 0; i < length; i++ ) {
+            silk_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 );
+            /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */
+            sum_Q6 = silk_RSHIFT( psDec->LPC_order, 1 );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  1 ], A_Q12[ 0 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  2 ], A_Q12[ 1 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  3 ], A_Q12[ 2 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  4 ], A_Q12[ 3 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  5 ], A_Q12[ 4 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  6 ], A_Q12[ 5 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  7 ], A_Q12[ 6 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  8 ], A_Q12[ 7 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i -  9 ], A_Q12[ 8 ] );
+            sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] );
+            if( psDec->LPC_order == 16 ) {
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 11 ], A_Q12[ 10 ] );
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 12 ], A_Q12[ 11 ] );
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 13 ], A_Q12[ 12 ] );
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 14 ], A_Q12[ 13 ] );
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 15 ], A_Q12[ 14 ] );
+                sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 16 ], A_Q12[ 15 ] );
+            }
+
+            /* Update states */
+            CNG_sig_Q10[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT( CNG_sig_Q10[ MAX_LPC_ORDER + i ], sum_Q6, 4 );
+
+            frame[ i ] = silk_ADD_SAT16( frame[ i ], silk_RSHIFT_ROUND( sum_Q6, 6 ) );
+        }
+        silk_memcpy( psCNG->CNG_synth_state, &CNG_sig_Q10[ length ], MAX_LPC_ORDER * sizeof( opus_int32 ) );
+    } else {
+        silk_memset( psCNG->CNG_synth_state, 0, psDec->LPC_order *  sizeof( opus_int32 ) );
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/Inlines.h b/lib/rbcodec/codecs/libopus/silk/Inlines.h
new file mode 100644
index 0000000000..87ac2e20d6
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/Inlines.h
@@ -0,0 +1,188 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+/*! \file silk_Inlines.h
+ *  \brief silk_Inlines.h defines inline signal processing functions.
+ */
+
+#ifndef SILK_FIX_INLINES_H
+#define SILK_FIX_INLINES_H
+
+#ifdef  __cplusplus
+extern "C"
+{
+#endif
+
+/* count leading zeros of opus_int64 */
+static inline opus_int32 silk_CLZ64( opus_int64 in )
+{
+    opus_int32 in_upper;
+
+    in_upper = (opus_int32)silk_RSHIFT64(in, 32);
+    if (in_upper == 0) {
+        /* Search in the lower 32 bits */
+        return 32 + silk_CLZ32( (opus_int32) in );
+    } else {
+        /* Search in the upper 32 bits */
+        return silk_CLZ32( in_upper );
+    }
+}
+
+/* get number of leading zeros and fractional part (the bits right after the leading one */
+static inline void silk_CLZ_FRAC(
+    opus_int32 in,            /* I  input                               */
+    opus_int32 *lz,           /* O  number of leading zeros             */
+    opus_int32 *frac_Q7       /* O  the 7 bits right after the leading one */
+)
+{
+    opus_int32 lzeros = silk_CLZ32(in);
+
+    * lz = lzeros;
+    * frac_Q7 = silk_ROR32(in, 24 - lzeros) & 0x7f;
+}
+
+/* Approximation of square root                                          */
+/* Accuracy: < +/- 10%  for output values > 15                           */
+/*           < +/- 2.5% for output values > 120                          */
+static inline opus_int32 silk_SQRT_APPROX( opus_int32 x )
+{
+    opus_int32 y, lz, frac_Q7;
+
+    if( x <= 0 ) {
+        return 0;
+    }
+
+    silk_CLZ_FRAC(x, &lz, &frac_Q7);
+
+    if( lz & 1 ) {
+        y = 32768;
+    } else {
+        y = 46214;        /* 46214 = sqrt(2) * 32768 */
+    }
+
+    /* get scaling right */
+    y >>= silk_RSHIFT(lz, 1);
+
+    /* increment using fractional part of input */
+    y = silk_SMLAWB(y, y, silk_SMULBB(213, frac_Q7));
+
+    return y;
+}
+
+/* Divide two int32 values and return result as int32 in a given Q-domain */
+static inline opus_int32 silk_DIV32_varQ(   /* O    returns a good approximation of "(a32 << Qres) / b32" */
+    const opus_int32     a32,               /* I    numerator (Q0)                  */
+    const opus_int32     b32,               /* I    denominator (Q0)                */
+    const opus_int       Qres               /* I    Q-domain of result (>= 0)       */
+)
+{
+    opus_int   a_headrm, b_headrm, lshift;
+    opus_int32 b32_inv, a32_nrm, b32_nrm, result;
+
+    silk_assert( b32 != 0 );
+    silk_assert( Qres >= 0 );
+
+    /* Compute number of bits head room and normalize inputs */
+    a_headrm = silk_CLZ32( silk_abs(a32) ) - 1;
+    a32_nrm = silk_LSHIFT(a32, a_headrm);                                       /* Q: a_headrm                  */
+    b_headrm = silk_CLZ32( silk_abs(b32) ) - 1;
+    b32_nrm = silk_LSHIFT(b32, b_headrm);                                       /* Q: b_headrm                  */
+
+    /* Inverse of b32, with 14 bits of precision */
+    b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) );   /* Q: 29 + 16 - b_headrm        */
+
+    /* First approximation */
+    result = silk_SMULWB(a32_nrm, b32_inv);                                     /* Q: 29 + a_headrm - b_headrm  */
+
+    /* Compute residual by subtracting product of denominator and first approximation */
+    /* It's OK to overflow because the final value of a32_nrm should always be small */
+    a32_nrm = silk_SUB32_ovflw(a32_nrm, silk_LSHIFT_ovflw( silk_SMMUL(b32_nrm, result), 3 ));  /* Q: a_headrm   */
+
+    /* Refinement */
+    result = silk_SMLAWB(result, a32_nrm, b32_inv);                             /* Q: 29 + a_headrm - b_headrm  */
+
+    /* Convert to Qres domain */
+    lshift = 29 + a_headrm - b_headrm - Qres;
+    if( lshift < 0 ) {
+        return silk_LSHIFT_SAT32(result, -lshift);
+    } else {
+        if( lshift < 32){
+            return silk_RSHIFT(result, lshift);
+        } else {
+            /* Avoid undefined result */
+            return 0;
+        }
+    }
+}
+
+/* Invert int32 value and return result as int32 in a given Q-domain */
+static inline opus_int32 silk_INVERSE32_varQ(   /* O    returns a good approximation of "(1 << Qres) / b32" */
+    const opus_int32     b32,                   /* I    denominator (Q0)                */
+    const opus_int       Qres                   /* I    Q-domain of result (> 0)        */
+)
+{
+    opus_int   b_headrm, lshift;
+    opus_int32 b32_inv, b32_nrm, err_Q32, result;
+
+    silk_assert( b32 != 0 );
+    silk_assert( Qres > 0 );
+
+    /* Compute number of bits head room and normalize input */
+    b_headrm = silk_CLZ32( silk_abs(b32) ) - 1;
+    b32_nrm = silk_LSHIFT(b32, b_headrm);                                       /* Q: b_headrm                */
+
+    /* Inverse of b32, with 14 bits of precision */
+    b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) );   /* Q: 29 + 16 - b_headrm    */
+
+    /* First approximation */
+    result = silk_LSHIFT(b32_inv, 16);                                          /* Q: 61 - b_headrm            */
+
+    /* Compute residual by subtracting product of denominator and first approximation from one */
+    err_Q32 = silk_LSHIFT( ((opus_int32)1<<29) - silk_SMULWB(b32_nrm, b32_inv), 3 );        /* Q32                        */
+
+    /* Refinement */
+    result = silk_SMLAWW(result, err_Q32, b32_inv);                             /* Q: 61 - b_headrm            */
+
+    /* Convert to Qres domain */
+    lshift = 61 - b_headrm - Qres;
+    if( lshift <= 0 ) {
+        return silk_LSHIFT_SAT32(result, -lshift);
+    } else {
+        if( lshift < 32){
+            return silk_RSHIFT(result, lshift);
+        }else{
+            /* Avoid undefined result */
+            return 0;
+        }
+    }
+}
+
+#ifdef  __cplusplus
+}
+#endif
+
+#endif /* SILK_FIX_INLINES_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/LPC_analysis_filter.c b/lib/rbcodec/codecs/libopus/silk/LPC_analysis_filter.c
new file mode 100644
index 0000000000..7d4458a637
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/LPC_analysis_filter.c
@@ -0,0 +1,85 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+/*******************************************/
+/* LPC analysis filter                     */
+/* NB! State is kept internally and the    */
+/* filter always starts with zero state    */
+/* first d output samples are set to zero  */
+/*******************************************/
+
+void silk_LPC_analysis_filter(
+    opus_int16                  *out,               /* O    Output signal                                               */
+    const opus_int16            *in,                /* I    Input signal                                                */
+    const opus_int16            *B,                 /* I    MA prediction coefficients, Q12 [order]                     */
+    const opus_int32            len,                /* I    Signal length                                               */
+    const opus_int32            d                   /* I    Filter order                                                */
+)
+{
+    opus_int         ix, j;
+    opus_int32       out32_Q12, out32;
+    const opus_int16 *in_ptr;
+
+    silk_assert( d >= 6 );
+    silk_assert( (d & 1) == 0 );
+    silk_assert( d <= len );
+
+    for( ix = d; ix < len; ix++ ) {
+        in_ptr = &in[ ix - 1 ];
+
+        out32_Q12 = silk_SMULBB( in_ptr[  0 ], B[ 0 ] );
+        /* Allowing wrap around so that two wraps can cancel each other. The rare
+           cases where the result wraps around can only be triggered by invalid streams*/
+        out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -1 ], B[ 1 ] );
+        out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -2 ], B[ 2 ] );
+        out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -3 ], B[ 3 ] );
+        out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -4 ], B[ 4 ] );
+        out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -5 ], B[ 5 ] );
+        for( j = 6; j < d; j += 2 ) {
+            out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j     ], B[ j     ] );
+            out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j - 1 ], B[ j + 1 ] );
+        }
+
+        /* Subtract prediction */
+        out32_Q12 = silk_SUB32_ovflw( silk_LSHIFT( (opus_int32)in_ptr[ 1 ], 12 ), out32_Q12 );
+
+        /* Scale to Q0 */
+        out32 = silk_RSHIFT_ROUND( out32_Q12, 12 );
+
+        /* Saturate output */
+        out[ ix ] = (opus_int16)silk_SAT16( out32 );
+    }
+
+    /* Set first d output samples to zero */
+    silk_memset( out, 0, d * sizeof( opus_int16 ) );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/LPC_inv_pred_gain.c b/lib/rbcodec/codecs/libopus/silk/LPC_inv_pred_gain.c
new file mode 100644
index 0000000000..afbb48ae1d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/LPC_inv_pred_gain.c
@@ -0,0 +1,154 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+#define QA                          24
+#define A_LIMIT                     SILK_FIX_CONST( 0.99975, QA )
+
+#define MUL32_FRAC_Q(a32, b32, Q)   ((opus_int32)(silk_RSHIFT_ROUND64(silk_SMULL(a32, b32), Q)))
+
+/* Compute inverse of LPC prediction gain, and                          */
+/* test if LPC coefficients are stable (all poles within unit circle)   */
+static opus_int32 LPC_inverse_pred_gain_QA(                 /* O   Returns inverse prediction gain in energy domain, Q30    */
+    opus_int32           A_QA[ 2 ][ SILK_MAX_ORDER_LPC ],   /* I   Prediction coefficients                                  */
+    const opus_int       order                              /* I   Prediction order                                         */
+)
+{
+    opus_int   k, n, mult2Q;
+    opus_int32 invGain_Q30, rc_Q31, rc_mult1_Q30, rc_mult2, tmp_QA;
+    opus_int32 *Aold_QA, *Anew_QA;
+
+    Anew_QA = A_QA[ order & 1 ];
+
+    invGain_Q30 = (opus_int32)1 << 30;
+    for( k = order - 1; k > 0; k-- ) {
+        /* Check for stability */
+        if( ( Anew_QA[ k ] > A_LIMIT ) || ( Anew_QA[ k ] < -A_LIMIT ) ) {
+            return 0;
+        }
+
+        /* Set RC equal to negated AR coef */
+        rc_Q31 = -silk_LSHIFT( Anew_QA[ k ], 31 - QA );
+
+        /* rc_mult1_Q30 range: [ 1 : 2^30 ] */
+        rc_mult1_Q30 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 );
+        silk_assert( rc_mult1_Q30 > ( 1 << 15 ) );                   /* reduce A_LIMIT if fails */
+        silk_assert( rc_mult1_Q30 <= ( 1 << 30 ) );
+
+        /* rc_mult2 range: [ 2^30 : silk_int32_MAX ] */
+        mult2Q = 32 - silk_CLZ32( silk_abs( rc_mult1_Q30 ) );
+        rc_mult2 = silk_INVERSE32_varQ( rc_mult1_Q30, mult2Q + 30 );
+
+        /* Update inverse gain */
+        /* invGain_Q30 range: [ 0 : 2^30 ] */
+        invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 );
+        silk_assert( invGain_Q30 >= 0           );
+        silk_assert( invGain_Q30 <= ( 1 << 30 ) );
+
+        /* Swap pointers */
+        Aold_QA = Anew_QA;
+        Anew_QA = A_QA[ k & 1 ];
+
+        /* Update AR coefficient */
+        for( n = 0; n < k; n++ ) {
+            tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 );
+            Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q );
+        }
+    }
+
+    /* Check for stability */
+    if( ( Anew_QA[ 0 ] > A_LIMIT ) || ( Anew_QA[ 0 ] < -A_LIMIT ) ) {
+        return 0;
+    }
+
+    /* Set RC equal to negated AR coef */
+    rc_Q31 = -silk_LSHIFT( Anew_QA[ 0 ], 31 - QA );
+
+    /* Range: [ 1 : 2^30 ] */
+    rc_mult1_Q30 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 );
+
+    /* Update inverse gain */
+    /* Range: [ 0 : 2^30 ] */
+    invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 );
+    silk_assert( invGain_Q30 >= 0     );
+    silk_assert( invGain_Q30 <= 1<<30 );
+
+    return invGain_Q30;
+}
+
+/* For input in Q12 domain */
+opus_int32 silk_LPC_inverse_pred_gain(              /* O   Returns inverse prediction gain in energy domain, Q30        */
+    const opus_int16            *A_Q12,             /* I   Prediction coefficients, Q12 [order]                         */
+    const opus_int              order               /* I   Prediction order                                             */
+)
+{
+    opus_int   k;
+    opus_int32 Atmp_QA[ 2 ][ SILK_MAX_ORDER_LPC ];
+    opus_int32 *Anew_QA;
+    opus_int32 DC_resp = 0;
+
+    Anew_QA = Atmp_QA[ order & 1 ];
+
+    /* Increase Q domain of the AR coefficients */
+    for( k = 0; k < order; k++ ) {
+        DC_resp += (opus_int32)A_Q12[ k ];
+        Anew_QA[ k ] = silk_LSHIFT32( (opus_int32)A_Q12[ k ], QA - 12 );
+    }
+    /* If the DC is unstable, we don't even need to do the full calculations */
+    if( DC_resp >= 4096 ) {
+        return 0;
+    }
+    return LPC_inverse_pred_gain_QA( Atmp_QA, order );
+}
+
+#ifdef FIXED_POINT
+
+/* For input in Q24 domain */
+opus_int32 silk_LPC_inverse_pred_gain_Q24(          /* O    Returns inverse prediction gain in energy domain, Q30       */
+    const opus_int32            *A_Q24,             /* I    Prediction coefficients [order]                             */
+    const opus_int              order               /* I    Prediction order                                            */
+)
+{
+    opus_int   k;
+    opus_int32 Atmp_QA[ 2 ][ SILK_MAX_ORDER_LPC ];
+    opus_int32 *Anew_QA;
+
+    Anew_QA = Atmp_QA[ order & 1 ];
+
+    /* Increase Q domain of the AR coefficients */
+    for( k = 0; k < order; k++ ) {
+        Anew_QA[ k ] = silk_RSHIFT32( A_Q24[ k ], 24 - QA );
+    }
+
+    return LPC_inverse_pred_gain_QA( Atmp_QA, order );
+}
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/MacroCount.h b/lib/rbcodec/codecs/libopus/silk/MacroCount.h
new file mode 100644
index 0000000000..2829e8ccb4
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/MacroCount.h
@@ -0,0 +1,718 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SIGPROCFIX_API_MACROCOUNT_H
+#define SIGPROCFIX_API_MACROCOUNT_H
+#include <stdio.h>
+
+#ifdef    silk_MACRO_COUNT
+#define varDefine opus_int64 ops_count = 0;
+
+extern opus_int64 ops_count;
+
+static inline opus_int64 silk_SaveCount(){
+    return(ops_count);
+}
+
+static inline opus_int64 silk_SaveResetCount(){
+    opus_int64 ret;
+
+    ret = ops_count;
+    ops_count = 0;
+    return(ret);
+}
+
+static inline silk_PrintCount(){
+    printf("ops_count = %d \n ", (opus_int32)ops_count);
+}
+
+#undef silk_MUL
+static inline opus_int32 silk_MUL(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    ops_count += 4;
+    ret = a32 * b32;
+    return ret;
+}
+
+#undef silk_MUL_uint
+static inline opus_uint32 silk_MUL_uint(opus_uint32 a32, opus_uint32 b32){
+    opus_uint32 ret;
+    ops_count += 4;
+    ret = a32 * b32;
+    return ret;
+}
+#undef silk_MLA
+static inline opus_int32 silk_MLA(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 4;
+    ret = a32 + b32 * c32;
+    return ret;
+}
+
+#undef silk_MLA_uint
+static inline opus_int32 silk_MLA_uint(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32){
+    opus_uint32 ret;
+    ops_count += 4;
+    ret = a32 + b32 * c32;
+    return ret;
+}
+
+#undef silk_SMULWB
+static inline opus_int32 silk_SMULWB(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    ops_count += 5;
+    ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16);
+    return ret;
+}
+#undef    silk_SMLAWB
+static inline opus_int32 silk_SMLAWB(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 5;
+    ret = ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16)));
+    return ret;
+}
+
+#undef silk_SMULWT
+static inline opus_int32 silk_SMULWT(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    ops_count += 4;
+    ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16);
+    return ret;
+}
+#undef silk_SMLAWT
+static inline opus_int32 silk_SMLAWT(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 4;
+    ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16));
+    return ret;
+}
+
+#undef silk_SMULBB
+static inline opus_int32 silk_SMULBB(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = (opus_int32)((opus_int16)a32) * (opus_int32)((opus_int16)b32);
+    return ret;
+}
+#undef silk_SMLABB
+static inline opus_int32 silk_SMLABB(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32);
+    return ret;
+}
+
+#undef silk_SMULBT
+static inline opus_int32 silk_SMULBT(opus_int32 a32, opus_int32 b32 ){
+    opus_int32 ret;
+    ops_count += 4;
+    ret = ((opus_int32)((opus_int16)a32)) * (b32 >> 16);
+    return ret;
+}
+
+#undef silk_SMLABT
+static inline opus_int32 silk_SMLABT(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16);
+    return ret;
+}
+
+#undef silk_SMULTT
+static inline opus_int32 silk_SMULTT(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = (a32 >> 16) * (b32 >> 16);
+    return ret;
+}
+
+#undef    silk_SMLATT
+static inline opus_int32 silk_SMLATT(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a32 + (b32 >> 16) * (c32 >> 16);
+    return ret;
+}
+
+
+/* multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode)*/
+#undef    silk_MLA_ovflw
+#define silk_MLA_ovflw silk_MLA
+
+#undef silk_SMLABB_ovflw
+#define silk_SMLABB_ovflw silk_SMLABB
+
+#undef silk_SMLABT_ovflw
+#define silk_SMLABT_ovflw silk_SMLABT
+
+#undef silk_SMLATT_ovflw
+#define silk_SMLATT_ovflw silk_SMLATT
+
+#undef silk_SMLAWB_ovflw
+#define silk_SMLAWB_ovflw silk_SMLAWB
+
+#undef silk_SMLAWT_ovflw
+#define silk_SMLAWT_ovflw silk_SMLAWT
+
+#undef silk_SMULL
+static inline opus_int64 silk_SMULL(opus_int32 a32, opus_int32 b32){
+    opus_int64 ret;
+    ops_count += 8;
+    ret = ((opus_int64)(a32) * /*(opus_int64)*/(b32));
+    return ret;
+}
+
+#undef    silk_SMLAL
+static inline opus_int64 silk_SMLAL(opus_int64 a64, opus_int32 b32, opus_int32 c32){
+    opus_int64 ret;
+    ops_count += 8;
+    ret = a64 + ((opus_int64)(b32) * /*(opus_int64)*/(c32));
+    return ret;
+}
+#undef    silk_SMLALBB
+static inline opus_int64 silk_SMLALBB(opus_int64 a64, opus_int16 b16, opus_int16 c16){
+    opus_int64 ret;
+    ops_count += 4;
+    ret = a64 + ((opus_int64)(b16) * /*(opus_int64)*/(c16));
+    return ret;
+}
+
+#undef    SigProcFIX_CLZ16
+static inline opus_int32 SigProcFIX_CLZ16(opus_int16 in16)
+{
+    opus_int32 out32 = 0;
+    ops_count += 10;
+    if( in16 == 0 ) {
+        return 16;
+    }
+    /* test nibbles */
+    if( in16 & 0xFF00 ) {
+        if( in16 & 0xF000 ) {
+            in16 >>= 12;
+        } else {
+            out32 += 4;
+            in16 >>= 8;
+        }
+    } else {
+        if( in16 & 0xFFF0 ) {
+            out32 += 8;
+            in16 >>= 4;
+        } else {
+            out32 += 12;
+        }
+    }
+    /* test bits and return */
+    if( in16 & 0xC ) {
+        if( in16 & 0x8 )
+            return out32 + 0;
+        else
+            return out32 + 1;
+    } else {
+        if( in16 & 0xE )
+            return out32 + 2;
+        else
+            return out32 + 3;
+    }
+}
+
+#undef SigProcFIX_CLZ32
+static inline opus_int32 SigProcFIX_CLZ32(opus_int32 in32)
+{
+    /* test highest 16 bits and convert to opus_int16 */
+    ops_count += 2;
+    if( in32 & 0xFFFF0000 ) {
+        return SigProcFIX_CLZ16((opus_int16)(in32 >> 16));
+    } else {
+        return SigProcFIX_CLZ16((opus_int16)in32) + 16;
+    }
+}
+
+#undef silk_DIV32
+static inline opus_int32 silk_DIV32(opus_int32 a32, opus_int32 b32){
+    ops_count += 64;
+    return a32 / b32;
+}
+
+#undef silk_DIV32_16
+static inline opus_int32 silk_DIV32_16(opus_int32 a32, opus_int32 b32){
+    ops_count += 32;
+    return a32 / b32;
+}
+
+#undef silk_SAT8
+static inline opus_int8 silk_SAT8(opus_int64 a){
+    opus_int8 tmp;
+    ops_count += 1;
+    tmp = (opus_int8)((a) > silk_int8_MAX ? silk_int8_MAX  : \
+                    ((a) < silk_int8_MIN ? silk_int8_MIN  : (a)));
+    return(tmp);
+}
+
+#undef silk_SAT16
+static inline opus_int16 silk_SAT16(opus_int64 a){
+    opus_int16 tmp;
+    ops_count += 1;
+    tmp = (opus_int16)((a) > silk_int16_MAX ? silk_int16_MAX  : \
+                     ((a) < silk_int16_MIN ? silk_int16_MIN  : (a)));
+    return(tmp);
+}
+#undef silk_SAT32
+static inline opus_int32 silk_SAT32(opus_int64 a){
+    opus_int32 tmp;
+    ops_count += 1;
+    tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX  : \
+                     ((a) < silk_int32_MIN ? silk_int32_MIN  : (a)));
+    return(tmp);
+}
+#undef silk_POS_SAT32
+static inline opus_int32 silk_POS_SAT32(opus_int64 a){
+    opus_int32 tmp;
+    ops_count += 1;
+    tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX : (a));
+    return(tmp);
+}
+
+#undef silk_ADD_POS_SAT8
+static inline opus_int8 silk_ADD_POS_SAT8(opus_int64 a, opus_int64 b){
+    opus_int8 tmp;
+    ops_count += 1;
+    tmp = (opus_int8)((((a)+(b)) & 0x80) ? silk_int8_MAX  : ((a)+(b)));
+    return(tmp);
+}
+#undef silk_ADD_POS_SAT16
+static inline opus_int16 silk_ADD_POS_SAT16(opus_int64 a, opus_int64 b){
+    opus_int16 tmp;
+    ops_count += 1;
+    tmp = (opus_int16)((((a)+(b)) & 0x8000) ? silk_int16_MAX : ((a)+(b)));
+    return(tmp);
+}
+
+#undef silk_ADD_POS_SAT32
+static inline opus_int32 silk_ADD_POS_SAT32(opus_int64 a, opus_int64 b){
+    opus_int32 tmp;
+    ops_count += 1;
+    tmp = (opus_int32)((((a)+(b)) & 0x80000000) ? silk_int32_MAX : ((a)+(b)));
+    return(tmp);
+}
+
+#undef silk_ADD_POS_SAT64
+static inline opus_int64 silk_ADD_POS_SAT64(opus_int64 a, opus_int64 b){
+    opus_int64 tmp;
+    ops_count += 1;
+    tmp = ((((a)+(b)) & 0x8000000000000000LL) ? silk_int64_MAX : ((a)+(b)));
+    return(tmp);
+}
+
+#undef    silk_LSHIFT8
+static inline opus_int8 silk_LSHIFT8(opus_int8 a, opus_int32 shift){
+    opus_int8 ret;
+    ops_count += 1;
+    ret = a << shift;
+    return ret;
+}
+#undef    silk_LSHIFT16
+static inline opus_int16 silk_LSHIFT16(opus_int16 a, opus_int32 shift){
+    opus_int16 ret;
+    ops_count += 1;
+    ret = a << shift;
+    return ret;
+}
+#undef    silk_LSHIFT32
+static inline opus_int32 silk_LSHIFT32(opus_int32 a, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a << shift;
+    return ret;
+}
+#undef    silk_LSHIFT64
+static inline opus_int64 silk_LSHIFT64(opus_int64 a, opus_int shift){
+    ops_count += 1;
+    return a << shift;
+}
+
+#undef    silk_LSHIFT_ovflw
+static inline opus_int32 silk_LSHIFT_ovflw(opus_int32 a, opus_int32 shift){
+    ops_count += 1;
+    return a << shift;
+}
+
+#undef    silk_LSHIFT_uint
+static inline opus_uint32 silk_LSHIFT_uint(opus_uint32 a, opus_int32 shift){
+    opus_uint32 ret;
+    ops_count += 1;
+    ret = a << shift;
+    return ret;
+}
+
+#undef    silk_RSHIFT8
+static inline opus_int8 silk_RSHIFT8(opus_int8 a, opus_int32 shift){
+    ops_count += 1;
+    return a >> shift;
+}
+#undef    silk_RSHIFT16
+static inline opus_int16 silk_RSHIFT16(opus_int16 a, opus_int32 shift){
+    ops_count += 1;
+    return a >> shift;
+}
+#undef    silk_RSHIFT32
+static inline opus_int32 silk_RSHIFT32(opus_int32 a, opus_int32 shift){
+    ops_count += 1;
+    return a >> shift;
+}
+#undef    silk_RSHIFT64
+static inline opus_int64 silk_RSHIFT64(opus_int64 a, opus_int64 shift){
+    ops_count += 1;
+    return a >> shift;
+}
+
+#undef    silk_RSHIFT_uint
+static inline opus_uint32 silk_RSHIFT_uint(opus_uint32 a, opus_int32 shift){
+    ops_count += 1;
+    return a >> shift;
+}
+
+#undef    silk_ADD_LSHIFT
+static inline opus_int32 silk_ADD_LSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a + (b << shift);
+    return ret;                /* shift >= 0*/
+}
+#undef    silk_ADD_LSHIFT32
+static inline opus_int32 silk_ADD_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a + (b << shift);
+    return ret;                /* shift >= 0*/
+}
+#undef    silk_ADD_LSHIFT_uint
+static inline opus_uint32 silk_ADD_LSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){
+    opus_uint32 ret;
+    ops_count += 1;
+    ret = a + (b << shift);
+    return ret;                /* shift >= 0*/
+}
+#undef    silk_ADD_RSHIFT
+static inline opus_int32 silk_ADD_RSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a + (b >> shift);
+    return ret;                /* shift  > 0*/
+}
+#undef    silk_ADD_RSHIFT32
+static inline opus_int32 silk_ADD_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a + (b >> shift);
+    return ret;                /* shift  > 0*/
+}
+#undef    silk_ADD_RSHIFT_uint
+static inline opus_uint32 silk_ADD_RSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){
+    opus_uint32 ret;
+    ops_count += 1;
+    ret = a + (b >> shift);
+    return ret;                /* shift  > 0*/
+}
+#undef    silk_SUB_LSHIFT32
+static inline opus_int32 silk_SUB_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a - (b << shift);
+    return ret;                /* shift >= 0*/
+}
+#undef    silk_SUB_RSHIFT32
+static inline opus_int32 silk_SUB_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a - (b >> shift);
+    return ret;                /* shift  > 0*/
+}
+
+#undef    silk_RSHIFT_ROUND
+static inline opus_int32 silk_RSHIFT_ROUND(opus_int32 a, opus_int32 shift){
+    opus_int32 ret;
+    ops_count += 3;
+    ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1;
+    return ret;
+}
+
+#undef    silk_RSHIFT_ROUND64
+static inline opus_int64 silk_RSHIFT_ROUND64(opus_int64 a, opus_int32 shift){
+    opus_int64 ret;
+    ops_count += 6;
+    ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1;
+    return ret;
+}
+
+#undef    silk_abs_int64
+static inline opus_int64 silk_abs_int64(opus_int64 a){
+    ops_count += 1;
+    return (((a) >  0)  ? (a) : -(a));            /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN*/
+}
+
+#undef    silk_abs_int32
+static inline opus_int32 silk_abs_int32(opus_int32 a){
+    ops_count += 1;
+    return silk_abs(a);
+}
+
+
+#undef silk_min
+static silk_min(a, b){
+    ops_count += 1;
+    return (((a) < (b)) ? (a) :  (b));
+}
+#undef silk_max
+static silk_max(a, b){
+    ops_count += 1;
+    return (((a) > (b)) ? (a) :  (b));
+}
+#undef silk_sign
+static silk_sign(a){
+    ops_count += 1;
+    return ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 ));
+}
+
+#undef    silk_ADD16
+static inline opus_int16 silk_ADD16(opus_int16 a, opus_int16 b){
+    opus_int16 ret;
+    ops_count += 1;
+    ret = a + b;
+    return ret;
+}
+
+#undef    silk_ADD32
+static inline opus_int32 silk_ADD32(opus_int32 a, opus_int32 b){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a + b;
+    return ret;
+}
+
+#undef    silk_ADD64
+static inline opus_int64 silk_ADD64(opus_int64 a, opus_int64 b){
+    opus_int64 ret;
+    ops_count += 2;
+    ret = a + b;
+    return ret;
+}
+
+#undef    silk_SUB16
+static inline opus_int16 silk_SUB16(opus_int16 a, opus_int16 b){
+    opus_int16 ret;
+    ops_count += 1;
+    ret = a - b;
+    return ret;
+}
+
+#undef    silk_SUB32
+static inline opus_int32 silk_SUB32(opus_int32 a, opus_int32 b){
+    opus_int32 ret;
+    ops_count += 1;
+    ret = a - b;
+    return ret;
+}
+
+#undef    silk_SUB64
+static inline opus_int64 silk_SUB64(opus_int64 a, opus_int64 b){
+    opus_int64 ret;
+    ops_count += 2;
+    ret = a - b;
+    return ret;
+}
+
+#undef silk_ADD_SAT16
+static inline opus_int16 silk_ADD_SAT16( opus_int16 a16, opus_int16 b16 ) {
+    opus_int16 res;
+    /* Nb will be counted in AKP_add32 and silk_SAT16*/
+    res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) );
+    return res;
+}
+
+#undef silk_ADD_SAT32
+static inline opus_int32 silk_ADD_SAT32(opus_int32 a32, opus_int32 b32){
+    opus_int32 res;
+    ops_count += 1;
+    res =    ((((a32) + (b32)) & 0x80000000) == 0 ?                                    \
+            ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) :    \
+            ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) );
+    return res;
+}
+
+#undef silk_ADD_SAT64
+static inline opus_int64 silk_ADD_SAT64( opus_int64 a64, opus_int64 b64 ) {
+    opus_int64 res;
+    ops_count += 1;
+    res =    ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ?                                \
+            ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) :    \
+            ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) );
+    return res;
+}
+
+#undef silk_SUB_SAT16
+static inline opus_int16 silk_SUB_SAT16( opus_int16 a16, opus_int16 b16 ) {
+    opus_int16 res;
+    silk_assert(0);
+    /* Nb will be counted in sub-macros*/
+    res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) );
+    return res;
+}
+
+#undef silk_SUB_SAT32
+static inline opus_int32 silk_SUB_SAT32( opus_int32 a32, opus_int32 b32 ) {
+    opus_int32 res;
+    ops_count += 1;
+    res =     ((((a32)-(b32)) & 0x80000000) == 0 ?                                            \
+            (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) :    \
+            ((((a32)^0x80000000) & (b32)  & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) );
+    return res;
+}
+
+#undef silk_SUB_SAT64
+static inline opus_int64 silk_SUB_SAT64( opus_int64 a64, opus_int64 b64 ) {
+    opus_int64 res;
+    ops_count += 1;
+    res =    ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ?                                                        \
+            (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) :    \
+            ((((a64)^0x8000000000000000LL) & (b64)  & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) );
+
+    return res;
+}
+
+#undef    silk_SMULWW
+static inline opus_int32 silk_SMULWW(opus_int32 a32, opus_int32 b32){
+    opus_int32 ret;
+    /* Nb will be counted in sub-macros*/
+    ret = silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16));
+    return ret;
+}
+
+#undef    silk_SMLAWW
+static inline opus_int32 silk_SMLAWW(opus_int32 a32, opus_int32 b32, opus_int32 c32){
+    opus_int32 ret;
+    /* Nb will be counted in sub-macros*/
+    ret = silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16));
+    return ret;
+}
+
+#undef    silk_min_int
+static inline opus_int silk_min_int(opus_int a, opus_int b)
+{
+    ops_count += 1;
+    return (((a) < (b)) ? (a) : (b));
+}
+
+#undef    silk_min_16
+static inline opus_int16 silk_min_16(opus_int16 a, opus_int16 b)
+{
+    ops_count += 1;
+    return (((a) < (b)) ? (a) : (b));
+}
+#undef    silk_min_32
+static inline opus_int32 silk_min_32(opus_int32 a, opus_int32 b)
+{
+    ops_count += 1;
+    return (((a) < (b)) ? (a) : (b));
+}
+#undef    silk_min_64
+static inline opus_int64 silk_min_64(opus_int64 a, opus_int64 b)
+{
+    ops_count += 1;
+    return (((a) < (b)) ? (a) : (b));
+}
+
+/* silk_min() versions with typecast in the function call */
+#undef    silk_max_int
+static inline opus_int silk_max_int(opus_int a, opus_int b)
+{
+    ops_count += 1;
+    return (((a) > (b)) ? (a) : (b));
+}
+#undef    silk_max_16
+static inline opus_int16 silk_max_16(opus_int16 a, opus_int16 b)
+{
+    ops_count += 1;
+    return (((a) > (b)) ? (a) : (b));
+}
+#undef    silk_max_32
+static inline opus_int32 silk_max_32(opus_int32 a, opus_int32 b)
+{
+    ops_count += 1;
+    return (((a) > (b)) ? (a) : (b));
+}
+
+#undef    silk_max_64
+static inline opus_int64 silk_max_64(opus_int64 a, opus_int64 b)
+{
+    ops_count += 1;
+    return (((a) > (b)) ? (a) : (b));
+}
+
+
+#undef silk_LIMIT_int
+static inline opus_int silk_LIMIT_int(opus_int a, opus_int limit1, opus_int limit2)
+{
+    opus_int ret;
+    ops_count += 6;
+
+    ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \
+        : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a))));
+
+    return(ret);
+}
+
+#undef silk_LIMIT_16
+static inline opus_int16 silk_LIMIT_16(opus_int16 a, opus_int16 limit1, opus_int16 limit2)
+{
+    opus_int16 ret;
+    ops_count += 6;
+
+    ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \
+        : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a))));
+
+return(ret);
+}
+
+
+#undef silk_LIMIT_32
+static inline opus_int silk_LIMIT_32(opus_int32 a, opus_int32 limit1, opus_int32 limit2)
+{
+    opus_int32 ret;
+    ops_count += 6;
+
+    ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \
+        : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a))));
+    return(ret);
+}
+
+#else
+#define varDefine
+#define silk_SaveCount()
+
+#endif
+#endif
+
diff --git a/lib/rbcodec/codecs/libopus/silk/MacroDebug.h b/lib/rbcodec/codecs/libopus/silk/MacroDebug.h
new file mode 100644
index 0000000000..ecd90bc4de
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/MacroDebug.h
@@ -0,0 +1,952 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Copyright (C) 2012 Xiph.Org Foundation
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef MACRO_DEBUG_H
+#define MACRO_DEBUG_H
+
+/* Redefine macro functions with extensive assertion in DEBUG mode.
+   As functions can't be undefined, this file can't work with SigProcFIX_MacroCount.h */
+
+#if ( defined (FIXED_DEBUG) || ( 0 && defined (_DEBUG) ) ) && !defined (silk_MACRO_COUNT)
+
+#undef silk_ADD16
+#define silk_ADD16(a,b) silk_ADD16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_ADD16_(opus_int16 a, opus_int16 b, char *file, int line){
+    opus_int16 ret;
+
+    ret = a + b;
+    if ( ret != silk_ADD_SAT16( a, b ) )
+    {
+        fprintf (stderr, "silk_ADD16(%d, %d) in %s: line %d\n", a, b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_ADD32
+#define silk_ADD32(a,b) silk_ADD32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_ADD32_(opus_int32 a, opus_int32 b, char *file, int line){
+    opus_int32 ret;
+
+    ret = a + b;
+    if ( ret != silk_ADD_SAT32( a, b ) )
+    {
+        fprintf (stderr, "silk_ADD32(%d, %d) in %s: line %d\n", a, b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_ADD64
+#define silk_ADD64(a,b) silk_ADD64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_ADD64_(opus_int64 a, opus_int64 b, char *file, int line){
+    opus_int64 ret;
+
+    ret = a + b;
+    if ( ret != silk_ADD_SAT64( a, b ) )
+    {
+        fprintf (stderr, "silk_ADD64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SUB16
+#define silk_SUB16(a,b) silk_SUB16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_SUB16_(opus_int16 a, opus_int16 b, char *file, int line){
+    opus_int16 ret;
+
+    ret = a - b;
+    if ( ret != silk_SUB_SAT16( a, b ) )
+    {
+        fprintf (stderr, "silk_SUB16(%d, %d) in %s: line %d\n", a, b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SUB32
+#define silk_SUB32(a,b) silk_SUB32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_SUB32_(opus_int32 a, opus_int32 b, char *file, int line){
+    opus_int32 ret;
+
+    ret = a - b;
+    if ( ret != silk_SUB_SAT32( a, b ) )
+    {
+        fprintf (stderr, "silk_SUB32(%d, %d) in %s: line %d\n", a, b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SUB64
+#define silk_SUB64(a,b) silk_SUB64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_SUB64_(opus_int64 a, opus_int64 b, char *file, int line){
+    opus_int64 ret;
+
+    ret = a - b;
+    if ( ret != silk_SUB_SAT64( a, b ) )
+    {
+        fprintf (stderr, "silk_SUB64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_ADD_SAT16
+#define silk_ADD_SAT16(a,b) silk_ADD_SAT16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_ADD_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line) {
+    opus_int16 res;
+    res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) );
+    if ( res != silk_SAT16( (opus_int32)a16 + (opus_int32)b16 ) )
+    {
+        fprintf (stderr, "silk_ADD_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_ADD_SAT32
+#define silk_ADD_SAT32(a,b) silk_ADD_SAT32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_ADD_SAT32_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    opus_int32 res;
+    res =   ((((opus_uint32)(a32) + (opus_uint32)(b32)) & 0x80000000) == 0 ?       \
+            ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) : \
+            ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) );
+    if ( res != silk_SAT32( (opus_int64)a32 + (opus_int64)b32 ) )
+    {
+        fprintf (stderr, "silk_ADD_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_ADD_SAT64
+#define silk_ADD_SAT64(a,b) silk_ADD_SAT64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_ADD_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line) {
+    opus_int64 res;
+    int        fail = 0;
+    res =   ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ?                                 \
+            ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) : \
+            ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) );
+    if( res != a64 + b64 ) {
+        /* Check that we saturated to the correct extreme value */
+        if ( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) ||
+               ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) ) )
+        {
+            fail = 1;
+        }
+    } else {
+        /* Saturation not necessary */
+        fail = res != a64 + b64;
+    }
+    if ( fail )
+    {
+        fprintf (stderr, "silk_ADD_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_SUB_SAT16
+#define silk_SUB_SAT16(a,b) silk_SUB_SAT16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_SUB_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line ) {
+    opus_int16 res;
+    res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) );
+    if ( res != silk_SAT16( (opus_int32)a16 - (opus_int32)b16 ) )
+    {
+        fprintf (stderr, "silk_SUB_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_SUB_SAT32
+#define silk_SUB_SAT32(a,b) silk_SUB_SAT32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_SUB_SAT32_( opus_int32 a32, opus_int32 b32, char *file, int line ) {
+    opus_int32 res;
+    res =   ((((opus_uint32)(a32)-(opus_uint32)(b32)) & 0x80000000) == 0 ?                \
+            (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) : \
+            ((((a32)^0x80000000) & (b32)  & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) );
+    if ( res != silk_SAT32( (opus_int64)a32 - (opus_int64)b32 ) )
+    {
+        fprintf (stderr, "silk_SUB_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_SUB_SAT64
+#define silk_SUB_SAT64(a,b) silk_SUB_SAT64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_SUB_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line ) {
+    opus_int64 res;
+    int        fail = 0;
+    res =   ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ?                                                    \
+            (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) : \
+            ((((a64)^0x8000000000000000LL) & (b64)  & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) );
+    if( res != a64 - b64 ) {
+        /* Check that we saturated to the correct extreme value */
+        if( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) ||
+              ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) ))
+        {
+            fail = 1;
+        }
+    } else {
+        /* Saturation not necessary */
+        fail = res != a64 - b64;
+    }
+    if ( fail )
+    {
+        fprintf (stderr, "silk_SUB_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return res;
+}
+
+#undef silk_MUL
+#define silk_MUL(a,b) silk_MUL_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_MUL_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    opus_int32 ret;
+    opus_int64 ret64;
+    ret = a32 * b32;
+    ret64 = (opus_int64)a32 * (opus_int64)b32;
+    if ( (opus_int64)ret != ret64 )
+    {
+        fprintf (stderr, "silk_MUL(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_MUL_uint
+#define silk_MUL_uint(a,b) silk_MUL_uint_((a), (b), __FILE__, __LINE__)
+static inline opus_uint32 silk_MUL_uint_(opus_uint32 a32, opus_uint32 b32, char *file, int line){
+    opus_uint32 ret;
+    ret = a32 * b32;
+    if ( (opus_uint64)ret != (opus_uint64)a32 * (opus_uint64)b32 )
+    {
+        fprintf (stderr, "silk_MUL_uint(%u, %u) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_MLA
+#define silk_MLA(a,b,c) silk_MLA_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_MLA_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = a32 + b32 * c32;
+    if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 )
+    {
+        fprintf (stderr, "silk_MLA(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_MLA_uint
+#define silk_MLA_uint(a,b,c) silk_MLA_uint_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_MLA_uint_(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32, char *file, int line){
+    opus_uint32 ret;
+    ret = a32 + b32 * c32;
+    if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 )
+    {
+        fprintf (stderr, "silk_MLA_uint(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMULWB
+#define silk_SMULWB(a,b) silk_SMULWB_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_SMULWB_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    opus_int32 ret;
+    ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16);
+    if ( (opus_int64)ret != ((opus_int64)a32 * (opus_int16)b32) >> 16 )
+    {
+        fprintf (stderr, "silk_SMULWB(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMLAWB
+#define silk_SMLAWB(a,b,c) silk_SMLAWB_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLAWB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = silk_ADD32( a32, silk_SMULWB( b32, c32 ) );
+    if ( silk_ADD32( a32, silk_SMULWB( b32, c32 ) ) != silk_ADD_SAT32( a32, silk_SMULWB( b32, c32 ) ) )
+    {
+        fprintf (stderr, "silk_SMLAWB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMULWT
+#define silk_SMULWT(a,b) silk_SMULWT_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_SMULWT_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    opus_int32 ret;
+    ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16);
+    if ( (opus_int64)ret != ((opus_int64)a32 * (b32 >> 16)) >> 16 )
+    {
+        fprintf (stderr, "silk_SMULWT(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMLAWT
+#define silk_SMLAWT(a,b,c) silk_SMLAWT_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLAWT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16));
+    if ( (opus_int64)ret != (opus_int64)a32 + (((opus_int64)b32 * (c32 >> 16)) >> 16) )
+    {
+        fprintf (stderr, "silk_SMLAWT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMULL
+#define silk_SMULL(a,b) silk_SMULL_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_SMULL_(opus_int64 a64, opus_int64 b64, char *file, int line){
+    opus_int64 ret64;
+    int        fail = 0;
+    ret64 = a64 * b64;
+    if( b64 != 0 ) {
+        fail = a64 != (ret64 / b64);
+    } else if( a64 != 0 ) {
+        fail = b64 != (ret64 / a64);
+    }
+    if ( fail )
+    {
+        fprintf (stderr, "silk_SMULL(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret64;
+}
+
+/* no checking needed for silk_SMULBB */
+#undef silk_SMLABB
+#define silk_SMLABB(a,b,c) silk_SMLABB_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLABB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32);
+    if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int16)c32 )
+    {
+        fprintf (stderr, "silk_SMLABB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+/* no checking needed for silk_SMULBT */
+#undef silk_SMLABT
+#define silk_SMLABT(a,b,c) silk_SMLABT_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLABT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16);
+    if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (c32 >> 16) )
+    {
+        fprintf (stderr, "silk_SMLABT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+/* no checking needed for silk_SMULTT */
+#undef silk_SMLATT
+#define silk_SMLATT(a,b,c) silk_SMLATT_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLATT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret;
+    ret = a32 + (b32 >> 16) * (c32 >> 16);
+    if ( (opus_int64)ret != (opus_int64)a32 + (b32 >> 16) * (c32 >> 16) )
+    {
+        fprintf (stderr, "silk_SMLATT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_SMULWW
+#define silk_SMULWW(a,b) silk_SMULWW_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_SMULWW_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    opus_int32 ret, tmp1, tmp2;
+    opus_int64 ret64;
+    int        fail = 0;
+
+    ret  = silk_SMULWB( a32, b32 );
+    tmp1 = silk_RSHIFT_ROUND( b32, 16 );
+    tmp2 = silk_MUL( a32, tmp1 );
+
+    fail |= (opus_int64)tmp2 != (opus_int64) a32 * (opus_int64) tmp1;
+
+    tmp1 = ret;
+    ret  = silk_ADD32( tmp1, tmp2 );
+    fail |= silk_ADD32( tmp1, tmp2 ) != silk_ADD_SAT32( tmp1, tmp2 );
+
+    ret64 = silk_RSHIFT64( silk_SMULL( a32, b32 ), 16 );
+    fail |= (opus_int64)ret != ret64;
+
+    if ( fail )
+    {
+        fprintf (stderr, "silk_SMULWT(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+
+    return ret;
+}
+
+#undef silk_SMLAWW
+#define silk_SMLAWW(a,b,c) silk_SMLAWW_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SMLAWW_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){
+    opus_int32 ret, tmp;
+
+    tmp = silk_SMULWW( b32, c32 );
+    ret = silk_ADD32( a32, tmp );
+    if ( ret != silk_ADD_SAT32( a32, tmp ) )
+    {
+        fprintf (stderr, "silk_SMLAWW(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+/* Multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode) */
+#undef  silk_MLA_ovflw
+#define silk_MLA_ovflw(a32, b32, c32)    ((a32) + ((b32) * (c32)))
+#undef  silk_SMLABB_ovflw
+#define silk_SMLABB_ovflw(a32, b32, c32)    ((a32) + ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32)))
+
+/* no checking needed for silk_SMULL
+   no checking needed for silk_SMLAL
+   no checking needed for silk_SMLALBB
+   no checking needed for SigProcFIX_CLZ16
+   no checking needed for SigProcFIX_CLZ32*/
+
+#undef silk_DIV32
+#define silk_DIV32(a,b) silk_DIV32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_DIV32_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    if ( b32 == 0 )
+    {
+        fprintf (stderr, "silk_DIV32(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a32 / b32;
+}
+
+#undef silk_DIV32_16
+#define silk_DIV32_16(a,b) silk_DIV32_16_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_DIV32_16_(opus_int32 a32, opus_int32 b32, char *file, int line){
+    int fail = 0;
+    fail |= b32 == 0;
+    fail |= b32 > silk_int16_MAX;
+    fail |= b32 < silk_int16_MIN;
+    if ( fail )
+    {
+        fprintf (stderr, "silk_DIV32_16(%d, %d) in %s: line %d\n", a32, b32, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a32 / b32;
+}
+
+/* no checking needed for silk_SAT8
+   no checking needed for silk_SAT16
+   no checking needed for silk_SAT32
+   no checking needed for silk_POS_SAT32
+   no checking needed for silk_ADD_POS_SAT8
+   no checking needed for silk_ADD_POS_SAT16
+   no checking needed for silk_ADD_POS_SAT32
+   no checking needed for silk_ADD_POS_SAT64 */
+
+#undef silk_LSHIFT8
+#define silk_LSHIFT8(a,b) silk_LSHIFT8_((a), (b), __FILE__, __LINE__)
+static inline opus_int8 silk_LSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){
+    opus_int8 ret;
+    int       fail = 0;
+    ret = a << shift;
+    fail |= shift < 0;
+    fail |= shift >= 8;
+    fail |= (opus_int64)ret != ((opus_int64)a) << shift;
+    if ( fail )
+    {
+        fprintf (stderr, "silk_LSHIFT8(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_LSHIFT16
+#define silk_LSHIFT16(a,b) silk_LSHIFT16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_LSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){
+    opus_int16 ret;
+    int        fail = 0;
+    ret = a << shift;
+    fail |= shift < 0;
+    fail |= shift >= 16;
+    fail |= (opus_int64)ret != ((opus_int64)a) << shift;
+    if ( fail )
+    {
+        fprintf (stderr, "silk_LSHIFT16(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_LSHIFT32
+#define silk_LSHIFT32(a,b) silk_LSHIFT32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_LSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    int        fail = 0;
+    ret = a << shift;
+    fail |= shift < 0;
+    fail |= shift >= 32;
+    fail |= (opus_int64)ret != ((opus_int64)a) << shift;
+    if ( fail )
+    {
+        fprintf (stderr, "silk_LSHIFT32(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_LSHIFT64
+#define silk_LSHIFT64(a,b) silk_LSHIFT64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_LSHIFT64_(opus_int64 a, opus_int shift, char *file, int line){
+    opus_int64 ret;
+    int        fail = 0;
+    ret = a << shift;
+    fail |= shift < 0;
+    fail |= shift >= 64;
+    fail |= (ret>>shift) != ((opus_int64)a);
+    if ( fail )
+    {
+        fprintf (stderr, "silk_LSHIFT64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_LSHIFT_ovflw
+#define silk_LSHIFT_ovflw(a,b) silk_LSHIFT_ovflw_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_LSHIFT_ovflw_(opus_int32 a, opus_int32 shift, char *file, int line){
+    if ( (shift < 0) || (shift >= 32) ) /* no check for overflow */
+    {
+        fprintf (stderr, "silk_LSHIFT_ovflw(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a << shift;
+}
+
+#undef silk_LSHIFT_uint
+#define silk_LSHIFT_uint(a,b) silk_LSHIFT_uint_((a), (b), __FILE__, __LINE__)
+static inline opus_uint32 silk_LSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){
+    opus_uint32 ret;
+    ret = a << shift;
+    if ( (shift < 0) || ((opus_int64)ret != ((opus_int64)a) << shift))
+    {
+        fprintf (stderr, "silk_LSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_RSHIFT8
+#define silk_RSHITF8(a,b) silk_RSHIFT8_((a), (b), __FILE__, __LINE__)
+static inline opus_int8 silk_RSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){
+    if ( (shift < 0) || (shift>=8) )
+    {
+        fprintf (stderr, "silk_RSHITF8(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a >> shift;
+}
+
+#undef silk_RSHIFT16
+#define silk_RSHITF16(a,b) silk_RSHIFT16_((a), (b), __FILE__, __LINE__)
+static inline opus_int16 silk_RSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){
+    if ( (shift < 0) || (shift>=16) )
+    {
+        fprintf (stderr, "silk_RSHITF16(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a >> shift;
+}
+
+#undef silk_RSHIFT32
+#define silk_RSHIFT32(a,b) silk_RSHIFT32_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_RSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){
+    if ( (shift < 0) || (shift>=32) )
+    {
+        fprintf (stderr, "silk_RSHITF32(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a >> shift;
+}
+
+#undef silk_RSHIFT64
+#define silk_RSHIFT64(a,b) silk_RSHIFT64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_RSHIFT64_(opus_int64 a, opus_int64 shift, char *file, int line){
+    if ( (shift < 0) || (shift>=64) )
+    {
+        fprintf (stderr, "silk_RSHITF64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a >> shift;
+}
+
+#undef silk_RSHIFT_uint
+#define silk_RSHIFT_uint(a,b) silk_RSHIFT_uint_((a), (b), __FILE__, __LINE__)
+static inline opus_uint32 silk_RSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){
+    if ( (shift < 0) || (shift>32) )
+    {
+        fprintf (stderr, "silk_RSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return a >> shift;
+}
+
+#undef silk_ADD_LSHIFT
+#define silk_ADD_LSHIFT(a,b,c) silk_ADD_LSHIFT_((a), (b), (c), __FILE__, __LINE__)
+static inline int silk_ADD_LSHIFT_(int a, int b, int shift, char *file, int line){
+    opus_int16 ret;
+    ret = a + (b << shift);
+    if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) )
+    {
+        fprintf (stderr, "silk_ADD_LSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift >= 0 */
+}
+
+#undef silk_ADD_LSHIFT32
+#define silk_ADD_LSHIFT32(a,b,c) silk_ADD_LSHIFT32_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_ADD_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    ret = a + (b << shift);
+    if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) )
+    {
+        fprintf (stderr, "silk_ADD_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift >= 0 */
+}
+
+#undef silk_ADD_LSHIFT_uint
+#define silk_ADD_LSHIFT_uint(a,b,c) silk_ADD_LSHIFT_uint_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_uint32 silk_ADD_LSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){
+    opus_uint32 ret;
+    ret = a + (b << shift);
+    if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) )
+    {
+        fprintf (stderr, "silk_ADD_LSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift >= 0 */
+}
+
+#undef silk_ADD_RSHIFT
+#define silk_ADD_RSHIFT(a,b,c) silk_ADD_RSHIFT_((a), (b), (c), __FILE__, __LINE__)
+static inline int silk_ADD_RSHIFT_(int a, int b, int shift, char *file, int line){
+    opus_int16 ret;
+    ret = a + (b >> shift);
+    if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) )
+    {
+        fprintf (stderr, "silk_ADD_RSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift  > 0 */
+}
+
+#undef silk_ADD_RSHIFT32
+#define silk_ADD_RSHIFT32(a,b,c) silk_ADD_RSHIFT32_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_ADD_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    ret = a + (b >> shift);
+    if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) )
+    {
+        fprintf (stderr, "silk_ADD_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift  > 0 */
+}
+
+#undef silk_ADD_RSHIFT_uint
+#define silk_ADD_RSHIFT_uint(a,b,c) silk_ADD_RSHIFT_uint_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_uint32 silk_ADD_RSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){
+    opus_uint32 ret;
+    ret = a + (b >> shift);
+    if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) )
+    {
+        fprintf (stderr, "silk_ADD_RSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift  > 0 */
+}
+
+#undef silk_SUB_LSHIFT32
+#define silk_SUB_LSHIFT32(a,b,c) silk_SUB_LSHIFT32_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SUB_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    ret = a - (b << shift);
+    if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (((opus_int64)b) << shift)) )
+    {
+        fprintf (stderr, "silk_SUB_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift >= 0 */
+}
+
+#undef silk_SUB_RSHIFT32
+#define silk_SUB_RSHIFT32(a,b,c) silk_SUB_RSHIFT32_((a), (b), (c), __FILE__, __LINE__)
+static inline opus_int32 silk_SUB_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    ret = a - (b >> shift);
+    if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (((opus_int64)b) >> shift)) )
+    {
+        fprintf (stderr, "silk_SUB_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;                /* shift  > 0 */
+}
+
+#undef silk_RSHIFT_ROUND
+#define silk_RSHIFT_ROUND(a,b) silk_RSHIFT_ROUND_((a), (b), __FILE__, __LINE__)
+static inline opus_int32 silk_RSHIFT_ROUND_(opus_int32 a, opus_int32 shift, char *file, int line){
+    opus_int32 ret;
+    ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1;
+    /* the marco definition can't handle a shift of zero */
+    if ( (shift <= 0) || (shift>31) || ((opus_int64)ret != ((opus_int64)a + ((opus_int64)1 << (shift - 1))) >> shift) )
+    {
+        fprintf (stderr, "silk_RSHIFT_ROUND(%d, %d) in %s: line %d\n", a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return ret;
+}
+
+#undef silk_RSHIFT_ROUND64
+#define silk_RSHIFT_ROUND64(a,b) silk_RSHIFT_ROUND64_((a), (b), __FILE__, __LINE__)
+static inline opus_int64 silk_RSHIFT_ROUND64_(opus_int64 a, opus_int32 shift, char *file, int line){
+    opus_int64 ret;
+    /* the marco definition can't handle a shift of zero */
+    if ( (shift <= 0) || (shift>=64) )
+    {
+        fprintf (stderr, "silk_RSHIFT_ROUND64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1;
+    return ret;
+}
+
+/* silk_abs is used on floats also, so doesn't work... */
+/*#undef silk_abs
+static inline opus_int32 silk_abs(opus_int32 a){
+    silk_assert(a != 0x80000000);
+    return (((a) >  0)  ? (a) : -(a));            // Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN
+}*/
+
+#undef silk_abs_int64
+#define silk_abs_int64(a) silk_abs_int64_((a), __FILE__, __LINE__)
+static inline opus_int64 silk_abs_int64_(opus_int64 a, char *file, int line){
+    if ( a == silk_int64_MIN )
+    {
+        fprintf (stderr, "silk_abs_int64(%lld) in %s: line %d\n", (long long)a, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return (((a) >  0)  ? (a) : -(a));            /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */
+}
+
+#undef silk_abs_int32
+#define silk_abs_int32(a) silk_abs_int32_((a), __FILE__, __LINE__)
+static inline opus_int32 silk_abs_int32_(opus_int32 a, char *file, int line){
+    if ( a == silk_int32_MIN )
+    {
+        fprintf (stderr, "silk_abs_int32(%d) in %s: line %d\n", a, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return silk_abs(a);
+}
+
+#undef silk_CHECK_FIT8
+#define silk_CHECK_FIT8(a) silk_CHECK_FIT8_((a), __FILE__, __LINE__)
+static inline opus_int8 silk_CHECK_FIT8_( opus_int64 a, char *file, int line ){
+    opus_int8 ret;
+    ret = (opus_int8)a;
+    if ( (opus_int64)ret != a )
+    {
+        fprintf (stderr, "silk_CHECK_FIT8(%lld) in %s: line %d\n", (long long)a, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return( ret );
+}
+
+#undef silk_CHECK_FIT16
+#define silk_CHECK_FIT16(a) silk_CHECK_FIT16_((a), __FILE__, __LINE__)
+static inline opus_int16 silk_CHECK_FIT16_( opus_int64 a, char *file, int line ){
+    opus_int16 ret;
+    ret = (opus_int16)a;
+    if ( (opus_int64)ret != a )
+    {
+        fprintf (stderr, "silk_CHECK_FIT16(%lld) in %s: line %d\n", (long long)a, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return( ret );
+}
+
+#undef silk_CHECK_FIT32
+#define silk_CHECK_FIT32(a) silk_CHECK_FIT32_((a), __FILE__, __LINE__)
+static inline opus_int32 silk_CHECK_FIT32_( opus_int64 a, char *file, int line ){
+    opus_int32 ret;
+    ret = (opus_int32)a;
+    if ( (opus_int64)ret != a )
+    {
+        fprintf (stderr, "silk_CHECK_FIT32(%lld) in %s: line %d\n", (long long)a, file, line);
+#ifdef FIXED_DEBUG_ASSERT
+        silk_assert( 0 );
+#endif
+    }
+    return( ret );
+}
+
+/* no checking for silk_NSHIFT_MUL_32_32
+   no checking for silk_NSHIFT_MUL_16_16
+   no checking needed for silk_min
+   no checking needed for silk_max
+   no checking needed for silk_sign
+*/
+
+#endif
+#endif /* MACRO_DEBUG_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/NLSF2A.c b/lib/rbcodec/codecs/libopus/silk/NLSF2A.c
new file mode 100644
index 0000000000..ffc2a96939
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/NLSF2A.c
@@ -0,0 +1,178 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/* conversion between prediction filter coefficients and LSFs   */
+/* order should be even                                         */
+/* a piecewise linear approximation maps LSF <-> cos(LSF)       */
+/* therefore the result is not accurate LSFs, but the two       */
+/* functions are accurate inverses of each other                */
+
+#include "SigProc_FIX.h"
+#include "tables.h"
+
+#define QA      16
+
+/* helper function for NLSF2A(..) */
+static inline void silk_NLSF2A_find_poly(
+    opus_int32          *out,      /* O    intermediate polynomial, QA [dd+1]        */
+    const opus_int32    *cLSF,     /* I    vector of interleaved 2*cos(LSFs), QA [d] */
+    opus_int            dd         /* I    polynomial order (= 1/2 * filter order)   */
+)
+{
+    opus_int   k, n;
+    opus_int32 ftmp;
+
+    out[0] = silk_LSHIFT( 1, QA );
+    out[1] = -cLSF[0];
+    for( k = 1; k < dd; k++ ) {
+        ftmp = cLSF[2*k];            /* QA*/
+        out[k+1] = silk_LSHIFT( out[k-1], 1 ) - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[k] ), QA );
+        for( n = k; n > 1; n-- ) {
+            out[n] += out[n-2] - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[n-1] ), QA );
+        }
+        out[1] -= ftmp;
+    }
+}
+
+/* compute whitening filter coefficients from normalized line spectral frequencies */
+void silk_NLSF2A(
+    opus_int16                  *a_Q12,             /* O    monic whitening filter coefficients in Q12,  [ d ]          */
+    const opus_int16            *NLSF,              /* I    normalized line spectral frequencies in Q15, [ d ]          */
+    const opus_int              d                   /* I    filter order (should be even)                               */
+)
+{
+    /* This ordering was found to maximize quality. It improves numerical accuracy of
+       silk_NLSF2A_find_poly() compared to "standard" ordering. */
+    static const unsigned char ordering16[16] = {
+      0, 15, 8, 7, 4, 11, 12, 3, 2, 13, 10, 5, 6, 9, 14, 1
+    };
+    static const unsigned char ordering10[10] = {
+      0, 9, 6, 3, 4, 5, 8, 1, 2, 7
+    };
+    const unsigned char *ordering;
+    opus_int   k, i, dd;
+    opus_int32 cos_LSF_QA[ SILK_MAX_ORDER_LPC ];
+    opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ], Q[ SILK_MAX_ORDER_LPC / 2 + 1 ];
+    opus_int32 Ptmp, Qtmp, f_int, f_frac, cos_val, delta;
+    opus_int32 a32_QA1[ SILK_MAX_ORDER_LPC ];
+    opus_int32 maxabs, absval, idx=0, sc_Q16;
+
+    silk_assert( LSF_COS_TAB_SZ_FIX == 128 );
+    silk_assert( d==10||d==16 );
+
+    /* convert LSFs to 2*cos(LSF), using piecewise linear curve from table */
+    ordering = d == 16 ? ordering16 : ordering10;
+    for( k = 0; k < d; k++ ) {
+        silk_assert(NLSF[k] >= 0 );
+
+        /* f_int on a scale 0-127 (rounded down) */
+        f_int = silk_RSHIFT( NLSF[k], 15 - 7 );
+
+        /* f_frac, range: 0..255 */
+        f_frac = NLSF[k] - silk_LSHIFT( f_int, 15 - 7 );
+
+        silk_assert(f_int >= 0);
+        silk_assert(f_int < LSF_COS_TAB_SZ_FIX );
+
+        /* Read start and end value from table */
+        cos_val = silk_LSFCosTab_FIX_Q12[ f_int ];                /* Q12 */
+        delta   = silk_LSFCosTab_FIX_Q12[ f_int + 1 ] - cos_val;  /* Q12, with a range of 0..200 */
+
+        /* Linear interpolation */
+        cos_LSF_QA[ordering[k]] = silk_RSHIFT_ROUND( silk_LSHIFT( cos_val, 8 ) + silk_MUL( delta, f_frac ), 20 - QA ); /* QA */
+    }
+
+    dd = silk_RSHIFT( d, 1 );
+
+    /* generate even and odd polynomials using convolution */
+    silk_NLSF2A_find_poly( P, &cos_LSF_QA[ 0 ], dd );
+    silk_NLSF2A_find_poly( Q, &cos_LSF_QA[ 1 ], dd );
+
+    /* convert even and odd polynomials to opus_int32 Q12 filter coefs */
+    for( k = 0; k < dd; k++ ) {
+        Ptmp = P[ k+1 ] + P[ k ];
+        Qtmp = Q[ k+1 ] - Q[ k ];
+
+        /* the Ptmp and Qtmp values at this stage need to fit in int32 */
+        a32_QA1[ k ]     = -Qtmp - Ptmp;        /* QA+1 */
+        a32_QA1[ d-k-1 ] =  Qtmp - Ptmp;        /* QA+1 */
+    }
+
+    /* Limit the maximum absolute value of the prediction coefficients, so that they'll fit in int16 */
+    for( i = 0; i < 10; i++ ) {
+        /* Find maximum absolute value and its index */
+        maxabs = 0;
+        for( k = 0; k < d; k++ ) {
+            absval = silk_abs( a32_QA1[k] );
+            if( absval > maxabs ) {
+                maxabs = absval;
+                idx    = k;
+            }
+        }
+        maxabs = silk_RSHIFT_ROUND( maxabs, QA + 1 - 12 );                                          /* QA+1 -> Q12 */
+
+        if( maxabs > silk_int16_MAX ) {
+            /* Reduce magnitude of prediction coefficients */
+            maxabs = silk_min( maxabs, 163838 );  /* ( silk_int32_MAX >> 14 ) + silk_int16_MAX = 163838 */
+            sc_Q16 = SILK_FIX_CONST( 0.999, 16 ) - silk_DIV32( silk_LSHIFT( maxabs - silk_int16_MAX, 14 ),
+                                        silk_RSHIFT32( silk_MUL( maxabs, idx + 1), 2 ) );
+            silk_bwexpander_32( a32_QA1, d, sc_Q16 );
+        } else {
+            break;
+        }
+    }
+
+    if( i == 10 ) {
+        /* Reached the last iteration, clip the coefficients */
+        for( k = 0; k < d; k++ ) {
+            a_Q12[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ) );  /* QA+1 -> Q12 */
+            a32_QA1[ k ] = silk_LSHIFT( (opus_int32)a_Q12[ k ], QA + 1 - 12 );
+        }
+    } else {
+        for( k = 0; k < d; k++ ) {
+            a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 );                /* QA+1 -> Q12 */
+        }
+    }
+
+    for( i = 0; i < MAX_LPC_STABILIZE_ITERATIONS; i++ ) {
+        if( silk_LPC_inverse_pred_gain( a_Q12, d ) < SILK_FIX_CONST( 1.0 / MAX_PREDICTION_POWER_GAIN, 30 ) ) {
+            /* Prediction coefficients are (too close to) unstable; apply bandwidth expansion   */
+            /* on the unscaled coefficients, convert to Q12 and measure again                   */
+            silk_bwexpander_32( a32_QA1, d, 65536 - silk_LSHIFT( 2, i ) );
+            for( k = 0; k < d; k++ ) {
+                a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 );            /* QA+1 -> Q12 */
+            }
+        } else {
+            break;
+        }
+    }
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/NLSF_VQ_weights_laroia.c b/lib/rbcodec/codecs/libopus/silk/NLSF_VQ_weights_laroia.c
new file mode 100644
index 0000000000..a89d6405ac
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/NLSF_VQ_weights_laroia.c
@@ -0,0 +1,80 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "define.h"
+#include "SigProc_FIX.h"
+
+/*
+R. Laroia, N. Phamdo and N. Farvardin, "Robust and Efficient Quantization of Speech LSP
+Parameters Using Structured Vector Quantization", Proc. IEEE Int. Conf. Acoust., Speech,
+Signal Processing, pp. 641-644, 1991.
+*/
+
+/* Laroia low complexity NLSF weights */
+void silk_NLSF_VQ_weights_laroia(
+    opus_int16                  *pNLSFW_Q_OUT,      /* O     Pointer to input vector weights [D]                        */
+    const opus_int16            *pNLSF_Q15,         /* I     Pointer to input vector         [D]                        */
+    const opus_int              D                   /* I     Input vector dimension (even)                              */
+)
+{
+    opus_int   k;
+    opus_int32 tmp1_int, tmp2_int;
+
+    silk_assert( D > 0 );
+    silk_assert( ( D & 1 ) == 0 );
+
+    /* First value */
+    tmp1_int = silk_max_int( pNLSF_Q15[ 0 ], 1 );
+    tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int );
+    tmp2_int = silk_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], 1 );
+    tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int );
+    pNLSFW_Q_OUT[ 0 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
+    silk_assert( pNLSFW_Q_OUT[ 0 ] > 0 );
+
+    /* Main loop */
+    for( k = 1; k < D - 1; k += 2 ) {
+        tmp1_int = silk_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], 1 );
+        tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int );
+        pNLSFW_Q_OUT[ k ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
+        silk_assert( pNLSFW_Q_OUT[ k ] > 0 );
+
+        tmp2_int = silk_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], 1 );
+        tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int );
+        pNLSFW_Q_OUT[ k + 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
+        silk_assert( pNLSFW_Q_OUT[ k + 1 ] > 0 );
+    }
+
+    /* Last value */
+    tmp1_int = silk_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], 1 );
+    tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int );
+    pNLSFW_Q_OUT[ D - 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
+    silk_assert( pNLSFW_Q_OUT[ D - 1 ] > 0 );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/NLSF_decode.c b/lib/rbcodec/codecs/libopus/silk/NLSF_decode.c
new file mode 100644
index 0000000000..6c2db4fd9d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/NLSF_decode.c
@@ -0,0 +1,101 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Predictive dequantizer for NLSF residuals */
+static inline void silk_NLSF_residual_dequant(               /* O    Returns RD value in Q30                     */
+          opus_int16         x_Q10[],                        /* O    Output [ order ]                            */
+    const opus_int8          indices[],                      /* I    Quantization indices [ order ]              */
+    const opus_uint8         pred_coef_Q8[],                 /* I    Backward predictor coefs [ order ]          */
+    const opus_int           quant_step_size_Q16,            /* I    Quantization step size                      */
+    const opus_int16         order                           /* I    Number of input values                      */
+)
+{
+    opus_int     i, out_Q10, pred_Q10;
+
+    out_Q10 = 0;
+    for( i = order-1; i >= 0; i-- ) {
+        pred_Q10 = silk_RSHIFT( silk_SMULBB( out_Q10, (opus_int16)pred_coef_Q8[ i ] ), 8 );
+        out_Q10  = silk_LSHIFT( indices[ i ], 10 );
+        if( out_Q10 > 0 ) {
+            out_Q10 = silk_SUB16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) );
+        } else if( out_Q10 < 0 ) {
+            out_Q10 = silk_ADD16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) );
+        }
+        out_Q10  = silk_SMLAWB( pred_Q10, (opus_int32)out_Q10, quant_step_size_Q16 );
+        x_Q10[ i ] = out_Q10;
+    }
+}
+
+
+/***********************/
+/* NLSF vector decoder */
+/***********************/
+void silk_NLSF_decode(
+          opus_int16            *pNLSF_Q15,                     /* O    Quantized NLSF vector [ LPC_ORDER ]         */
+          opus_int8             *NLSFIndices,                   /* I    Codebook path vector [ LPC_ORDER + 1 ]      */
+    const silk_NLSF_CB_struct   *psNLSF_CB                      /* I    Codebook object                             */
+)
+{
+    opus_int         i;
+    opus_uint8       pred_Q8[  MAX_LPC_ORDER ];
+    opus_int16       ec_ix[    MAX_LPC_ORDER ];
+    opus_int16       res_Q10[  MAX_LPC_ORDER ];
+    opus_int16       W_tmp_QW[ MAX_LPC_ORDER ];
+    opus_int32       W_tmp_Q9, NLSF_Q15_tmp;
+    const opus_uint8 *pCB_element;
+
+    /* Decode first stage */
+    pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ NLSFIndices[ 0 ] * psNLSF_CB->order ];
+    for( i = 0; i < psNLSF_CB->order; i++ ) {
+        pNLSF_Q15[ i ] = silk_LSHIFT( (opus_int16)pCB_element[ i ], 7 );
+    }
+
+    /* Unpack entropy table indices and predictor for current CB1 index */
+    silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, NLSFIndices[ 0 ] );
+
+    /* Predictive residual dequantizer */
+    silk_NLSF_residual_dequant( res_Q10, &NLSFIndices[ 1 ], pred_Q8, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->order );
+
+    /* Weights from codebook vector */
+    silk_NLSF_VQ_weights_laroia( W_tmp_QW, pNLSF_Q15, psNLSF_CB->order );
+
+    /* Apply inverse square-rooted weights and add to output */
+    for( i = 0; i < psNLSF_CB->order; i++ ) {
+        W_tmp_Q9 = silk_SQRT_APPROX( silk_LSHIFT( (opus_int32)W_tmp_QW[ i ], 18 - NLSF_W_Q ) );
+        NLSF_Q15_tmp = silk_ADD32( pNLSF_Q15[ i ], silk_DIV32_16( silk_LSHIFT( (opus_int32)res_Q10[ i ], 14 ), W_tmp_Q9 ) );
+        pNLSF_Q15[ i ] = (opus_int16)silk_LIMIT( NLSF_Q15_tmp, 0, 32767 );
+    }
+
+    /* NLSF stabilization */
+    silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/NLSF_stabilize.c b/lib/rbcodec/codecs/libopus/silk/NLSF_stabilize.c
new file mode 100644
index 0000000000..25ec49f4c1
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/NLSF_stabilize.c
@@ -0,0 +1,142 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/* NLSF stabilizer:                                         */
+/*                                                          */
+/* - Moves NLSFs futher apart if they are too close         */
+/* - Moves NLSFs away from borders if they are too close    */
+/* - High effort to achieve a modification with minimum     */
+/*     Euclidean distance to input vector                   */
+/* - Output are sorted NLSF coefficients                    */
+/*                                                          */
+
+#include "SigProc_FIX.h"
+
+/* Constant Definitions */
+#define MAX_LOOPS        20
+
+/* NLSF stabilizer, for a single input data vector */
+void silk_NLSF_stabilize(
+          opus_int16            *NLSF_Q15,          /* I/O   Unstable/stabilized normalized LSF vector in Q15 [L]       */
+    const opus_int16            *NDeltaMin_Q15,     /* I     Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1]   */
+    const opus_int              L                   /* I     Number of NLSF parameters in the input vector              */
+)
+{
+    opus_int   i, I=0, k, loops;
+    opus_int16 center_freq_Q15;
+    opus_int32 diff_Q15, min_diff_Q15, min_center_Q15, max_center_Q15;
+
+    /* This is necessary to ensure an output within range of a opus_int16 */
+    silk_assert( NDeltaMin_Q15[L] >= 1 );
+
+    for( loops = 0; loops < MAX_LOOPS; loops++ ) {
+        /**************************/
+        /* Find smallest distance */
+        /**************************/
+        /* First element */
+        min_diff_Q15 = NLSF_Q15[0] - NDeltaMin_Q15[0];
+        I = 0;
+        /* Middle elements */
+        for( i = 1; i <= L-1; i++ ) {
+            diff_Q15 = NLSF_Q15[i] - ( NLSF_Q15[i-1] + NDeltaMin_Q15[i] );
+            if( diff_Q15 < min_diff_Q15 ) {
+                min_diff_Q15 = diff_Q15;
+                I = i;
+            }
+        }
+        /* Last element */
+        diff_Q15 = ( 1 << 15 ) - ( NLSF_Q15[L-1] + NDeltaMin_Q15[L] );
+        if( diff_Q15 < min_diff_Q15 ) {
+            min_diff_Q15 = diff_Q15;
+            I = L;
+        }
+
+        /***************************************************/
+        /* Now check if the smallest distance non-negative */
+        /***************************************************/
+        if( min_diff_Q15 >= 0 ) {
+            return;
+        }
+
+        if( I == 0 ) {
+            /* Move away from lower limit */
+            NLSF_Q15[0] = NDeltaMin_Q15[0];
+
+        } else if( I == L) {
+            /* Move away from higher limit */
+            NLSF_Q15[L-1] = ( 1 << 15 ) - NDeltaMin_Q15[L];
+
+        } else {
+            /* Find the lower extreme for the location of the current center frequency */
+            min_center_Q15 = 0;
+            for( k = 0; k < I; k++ ) {
+                min_center_Q15 += NDeltaMin_Q15[k];
+            }
+            min_center_Q15 += silk_RSHIFT( NDeltaMin_Q15[I], 1 );
+
+            /* Find the upper extreme for the location of the current center frequency */
+            max_center_Q15 = 1 << 15;
+            for( k = L; k > I; k-- ) {
+                max_center_Q15 -= NDeltaMin_Q15[k];
+            }
+            max_center_Q15 -= silk_RSHIFT( NDeltaMin_Q15[I], 1 );
+
+            /* Move apart, sorted by value, keeping the same center frequency */
+            center_freq_Q15 = (opus_int16)silk_LIMIT_32( silk_RSHIFT_ROUND( (opus_int32)NLSF_Q15[I-1] + (opus_int32)NLSF_Q15[I], 1 ),
+                min_center_Q15, max_center_Q15 );
+            NLSF_Q15[I-1] = center_freq_Q15 - silk_RSHIFT( NDeltaMin_Q15[I], 1 );
+            NLSF_Q15[I] = NLSF_Q15[I-1] + NDeltaMin_Q15[I];
+        }
+    }
+
+    /* Safe and simple fall back method, which is less ideal than the above */
+    if( loops == MAX_LOOPS )
+    {
+        /* Insertion sort (fast for already almost sorted arrays):   */
+        /* Best case:  O(n)   for an already sorted array            */
+        /* Worst case: O(n^2) for an inversely sorted array          */
+        silk_insertion_sort_increasing_all_values_int16( &NLSF_Q15[0], L );
+
+        /* First NLSF should be no less than NDeltaMin[0] */
+        NLSF_Q15[0] = silk_max_int( NLSF_Q15[0], NDeltaMin_Q15[0] );
+
+        /* Keep delta_min distance between the NLSFs */
+        for( i = 1; i < L; i++ )
+            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], NLSF_Q15[i-1] + NDeltaMin_Q15[i] );
+
+        /* Last NLSF should be no higher than 1 - NDeltaMin[L] */
+        NLSF_Q15[L-1] = silk_min_int( NLSF_Q15[L-1], (1<<15) - NDeltaMin_Q15[L] );
+
+        /* Keep NDeltaMin distance between the NLSFs */
+        for( i = L-2; i >= 0; i-- )
+            NLSF_Q15[i] = silk_min_int( NLSF_Q15[i], NLSF_Q15[i+1] - NDeltaMin_Q15[i+1] );
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/NLSF_unpack.c b/lib/rbcodec/codecs/libopus/silk/NLSF_unpack.c
new file mode 100644
index 0000000000..5e059f2615
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/NLSF_unpack.c
@@ -0,0 +1,55 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Unpack predictor values and indices for entropy coding tables */
+void silk_NLSF_unpack(
+          opus_int16            ec_ix[],                        /* O    Indices to entropy tables [ LPC_ORDER ]     */
+          opus_uint8            pred_Q8[],                      /* O    LSF predictor [ LPC_ORDER ]                 */
+    const silk_NLSF_CB_struct   *psNLSF_CB,                     /* I    Codebook object                             */
+    const opus_int              CB1_index                       /* I    Index of vector in first LSF codebook       */
+)
+{
+    opus_int   i;
+    opus_uint8 entry;
+    const opus_uint8 *ec_sel_ptr;
+
+    ec_sel_ptr = &psNLSF_CB->ec_sel[ CB1_index * psNLSF_CB->order / 2 ];
+    for( i = 0; i < psNLSF_CB->order; i += 2 ) {
+        entry = *ec_sel_ptr++;
+        ec_ix  [ i     ] = silk_SMULBB( silk_RSHIFT( entry, 1 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 );
+        pred_Q8[ i     ] = psNLSF_CB->pred_Q8[ i + ( entry & 1 ) * ( psNLSF_CB->order - 1 ) ];
+        ec_ix  [ i + 1 ] = silk_SMULBB( silk_RSHIFT( entry, 5 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 );
+        pred_Q8[ i + 1 ] = psNLSF_CB->pred_Q8[ i + ( silk_RSHIFT( entry, 4 ) & 1 ) * ( psNLSF_CB->order - 1 ) + 1 ];
+    }
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/PLC.c b/lib/rbcodec/codecs/libopus/silk/PLC.c
new file mode 100644
index 0000000000..08ae5b7617
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/PLC.c
@@ -0,0 +1,423 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+#include "stack_alloc.h"
+#include "PLC.h"
+
+#define NB_ATT 2
+static const opus_int16 HARM_ATT_Q15[NB_ATT]              = { 32440, 31130 }; /* 0.99, 0.95 */
+static const opus_int16 PLC_RAND_ATTENUATE_V_Q15[NB_ATT]  = { 31130, 26214 }; /* 0.95, 0.8 */
+static const opus_int16 PLC_RAND_ATTENUATE_UV_Q15[NB_ATT] = { 32440, 29491 }; /* 0.99, 0.9 */
+
+static inline void silk_PLC_update(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl          /* I/O Decoder control      */
+);
+
+static inline void silk_PLC_conceal(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl,         /* I/O Decoder control      */
+    opus_int16                          frame[]             /* O LPC residual signal    */
+);
+
+
+void silk_PLC_Reset(
+    silk_decoder_state                  *psDec              /* I/O Decoder state        */
+)
+{
+    psDec->sPLC.pitchL_Q8 = silk_LSHIFT( psDec->frame_length, 8 - 1 );
+    psDec->sPLC.prevGain_Q16[ 0 ] = SILK_FIX_CONST( 1, 16 );
+    psDec->sPLC.prevGain_Q16[ 1 ] = SILK_FIX_CONST( 1, 16 );
+    psDec->sPLC.subfr_length = 20;
+    psDec->sPLC.nb_subfr = 2;
+}
+
+void silk_PLC(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl,         /* I/O Decoder control      */
+    opus_int16                          frame[],            /* I/O  signal              */
+    opus_int                            lost                /* I Loss flag              */
+)
+{
+    /* PLC control function */
+    if( psDec->fs_kHz != psDec->sPLC.fs_kHz ) {
+        silk_PLC_Reset( psDec );
+        psDec->sPLC.fs_kHz = psDec->fs_kHz;
+    }
+
+    if( lost ) {
+        /****************************/
+        /* Generate Signal          */
+        /****************************/
+        silk_PLC_conceal( psDec, psDecCtrl, frame );
+
+        psDec->lossCnt++;
+    } else {
+        /****************************/
+        /* Update state             */
+        /****************************/
+        silk_PLC_update( psDec, psDecCtrl );
+    }
+}
+
+/**************************************************/
+/* Update state of PLC                            */
+/**************************************************/
+static inline void silk_PLC_update(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl          /* I/O Decoder control      */
+)
+{
+    opus_int32 LTP_Gain_Q14, temp_LTP_Gain_Q14;
+    opus_int   i, j;
+    silk_PLC_struct *psPLC;
+
+    psPLC = &psDec->sPLC;
+
+    /* Update parameters used in case of packet loss */
+    psDec->prevSignalType = psDec->indices.signalType;
+    LTP_Gain_Q14 = 0;
+    if( psDec->indices.signalType == TYPE_VOICED ) {
+        /* Find the parameters for the last subframe which contains a pitch pulse */
+        for( j = 0; j * psDec->subfr_length < psDecCtrl->pitchL[ psDec->nb_subfr - 1 ]; j++ ) {
+            if( j == psDec->nb_subfr ) {
+                break;
+            }
+            temp_LTP_Gain_Q14 = 0;
+            for( i = 0; i < LTP_ORDER; i++ ) {
+                temp_LTP_Gain_Q14 += psDecCtrl->LTPCoef_Q14[ ( psDec->nb_subfr - 1 - j ) * LTP_ORDER  + i ];
+            }
+            if( temp_LTP_Gain_Q14 > LTP_Gain_Q14 ) {
+                LTP_Gain_Q14 = temp_LTP_Gain_Q14;
+                silk_memcpy( psPLC->LTPCoef_Q14,
+                    &psDecCtrl->LTPCoef_Q14[ silk_SMULBB( psDec->nb_subfr - 1 - j, LTP_ORDER ) ],
+                    LTP_ORDER * sizeof( opus_int16 ) );
+
+                psPLC->pitchL_Q8 = silk_LSHIFT( psDecCtrl->pitchL[ psDec->nb_subfr - 1 - j ], 8 );
+            }
+        }
+
+        silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) );
+        psPLC->LTPCoef_Q14[ LTP_ORDER / 2 ] = LTP_Gain_Q14;
+
+        /* Limit LT coefs */
+        if( LTP_Gain_Q14 < V_PITCH_GAIN_START_MIN_Q14 ) {
+            opus_int   scale_Q10;
+            opus_int32 tmp;
+
+            tmp = silk_LSHIFT( V_PITCH_GAIN_START_MIN_Q14, 10 );
+            scale_Q10 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) );
+            for( i = 0; i < LTP_ORDER; i++ ) {
+                psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q10 ), 10 );
+            }
+        } else if( LTP_Gain_Q14 > V_PITCH_GAIN_START_MAX_Q14 ) {
+            opus_int   scale_Q14;
+            opus_int32 tmp;
+
+            tmp = silk_LSHIFT( V_PITCH_GAIN_START_MAX_Q14, 14 );
+            scale_Q14 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) );
+            for( i = 0; i < LTP_ORDER; i++ ) {
+                psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q14 ), 14 );
+            }
+        }
+    } else {
+        psPLC->pitchL_Q8 = silk_LSHIFT( silk_SMULBB( psDec->fs_kHz, 18 ), 8 );
+        silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 ));
+    }
+
+    /* Save LPC coeficients */
+    silk_memcpy( psPLC->prevLPC_Q12, psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) );
+    psPLC->prevLTP_scale_Q14 = psDecCtrl->LTP_scale_Q14;
+
+    /* Save last two gains */
+    silk_memcpy( psPLC->prevGain_Q16, &psDecCtrl->Gains_Q16[ psDec->nb_subfr - 2 ], 2 * sizeof( opus_int32 ) );
+
+    psPLC->subfr_length = psDec->subfr_length;
+    psPLC->nb_subfr = psDec->nb_subfr;
+}
+
+static inline void silk_PLC_conceal(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl,         /* I/O Decoder control      */
+    opus_int16                          frame[]             /* O LPC residual signal    */
+)
+{
+    opus_int   i, j, k;
+    opus_int   lag, idx, sLTP_buf_idx, shift1, shift2;
+    opus_int32 rand_seed, harm_Gain_Q15, rand_Gain_Q15, inv_gain_Q30;
+    opus_int32 energy1, energy2, *rand_ptr, *pred_lag_ptr;
+    opus_int32 LPC_pred_Q10, LTP_pred_Q12;
+    opus_int16 rand_scale_Q14;
+    opus_int16 *B_Q14, *exc_buf_ptr;
+    opus_int32 *sLPC_Q14_ptr;
+    VARDECL( opus_int16, exc_buf );
+    opus_int16 A_Q12[ MAX_LPC_ORDER ];
+    VARDECL( opus_int16, sLTP );
+    VARDECL( opus_int32, sLTP_Q14 );
+    silk_PLC_struct *psPLC = &psDec->sPLC;
+    opus_int32 prevGain_Q10[2];
+    SAVE_STACK;
+
+    ALLOC( exc_buf, 2*psPLC->subfr_length, opus_int16 );
+    ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 );
+    ALLOC( sLTP_Q14, psDec->ltp_mem_length + psDec->frame_length, opus_int32 );
+
+    prevGain_Q10[0] = silk_RSHIFT( psPLC->prevGain_Q16[ 0 ], 6);
+    prevGain_Q10[1] = silk_RSHIFT( psPLC->prevGain_Q16[ 1 ], 6);
+
+    if( psDec->first_frame_after_reset ) {
+       silk_memset( psPLC->prevLPC_Q12, 0, sizeof( psPLC->prevLPC_Q12 ) );
+    }
+
+    /* Find random noise component */
+    /* Scale previous excitation signal */
+    exc_buf_ptr = exc_buf;
+    for( k = 0; k < 2; k++ ) {
+        for( i = 0; i < psPLC->subfr_length; i++ ) {
+            exc_buf_ptr[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT(
+                silk_SMULWW( psDec->exc_Q14[ i + ( k + psPLC->nb_subfr - 2 ) * psPLC->subfr_length ], prevGain_Q10[ k ] ), 8 ) );
+        }
+        exc_buf_ptr += psPLC->subfr_length;
+    }
+    /* Find the subframe with lowest energy of the last two and use that as random noise generator */
+    silk_sum_sqr_shift( &energy1, &shift1, exc_buf,                         psPLC->subfr_length );
+    silk_sum_sqr_shift( &energy2, &shift2, &exc_buf[ psPLC->subfr_length ], psPLC->subfr_length );
+
+    if( silk_RSHIFT( energy1, shift2 ) < silk_RSHIFT( energy2, shift1 ) ) {
+        /* First sub-frame has lowest energy */
+        rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, ( psPLC->nb_subfr - 1 ) * psPLC->subfr_length - RAND_BUF_SIZE ) ];
+    } else {
+        /* Second sub-frame has lowest energy */
+        rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, psPLC->nb_subfr * psPLC->subfr_length - RAND_BUF_SIZE ) ];
+    }
+
+    /* Set up Gain to random noise component */
+    B_Q14          = psPLC->LTPCoef_Q14;
+    rand_scale_Q14 = psPLC->randScale_Q14;
+
+    /* Set up attenuation gains */
+    harm_Gain_Q15 = HARM_ATT_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
+    if( psDec->prevSignalType == TYPE_VOICED ) {
+        rand_Gain_Q15 = PLC_RAND_ATTENUATE_V_Q15[  silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
+    } else {
+        rand_Gain_Q15 = PLC_RAND_ATTENUATE_UV_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
+    }
+
+    /* LPC concealment. Apply BWE to previous LPC */
+    silk_bwexpander( psPLC->prevLPC_Q12, psDec->LPC_order, SILK_FIX_CONST( BWE_COEF, 16 ) );
+
+    /* Preload LPC coeficients to array on stack. Gives small performance gain */
+    silk_memcpy( A_Q12, psPLC->prevLPC_Q12, psDec->LPC_order * sizeof( opus_int16 ) );
+
+    /* First Lost frame */
+    if( psDec->lossCnt == 0 ) {
+        rand_scale_Q14 = 1 << 14;
+
+        /* Reduce random noise Gain for voiced frames */
+        if( psDec->prevSignalType == TYPE_VOICED ) {
+            for( i = 0; i < LTP_ORDER; i++ ) {
+                rand_scale_Q14 -= B_Q14[ i ];
+            }
+            rand_scale_Q14 = silk_max_16( 3277, rand_scale_Q14 ); /* 0.2 */
+            rand_scale_Q14 = (opus_int16)silk_RSHIFT( silk_SMULBB( rand_scale_Q14, psPLC->prevLTP_scale_Q14 ), 14 );
+        } else {
+            /* Reduce random noise for unvoiced frames with high LPC gain */
+            opus_int32 invGain_Q30, down_scale_Q30;
+
+            invGain_Q30 = silk_LPC_inverse_pred_gain( psPLC->prevLPC_Q12, psDec->LPC_order );
+
+            down_scale_Q30 = silk_min_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_HIGH_THRES ), invGain_Q30 );
+            down_scale_Q30 = silk_max_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_LOW_THRES ), down_scale_Q30 );
+            down_scale_Q30 = silk_LSHIFT( down_scale_Q30, LOG2_INV_LPC_GAIN_HIGH_THRES );
+
+            rand_Gain_Q15 = silk_RSHIFT( silk_SMULWB( down_scale_Q30, rand_Gain_Q15 ), 14 );
+        }
+    }
+
+    rand_seed    = psPLC->rand_seed;
+    lag          = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 );
+    sLTP_buf_idx = psDec->ltp_mem_length;
+
+    /* Rewhiten LTP state */
+    idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2;
+    silk_assert( idx > 0 );
+    silk_LPC_analysis_filter( &sLTP[ idx ], &psDec->outBuf[ idx ], A_Q12, psDec->ltp_mem_length - idx, psDec->LPC_order );
+    /* Scale LTP state */
+    inv_gain_Q30 = silk_INVERSE32_varQ( psPLC->prevGain_Q16[ 1 ], 46 );
+    inv_gain_Q30 = silk_min( inv_gain_Q30, silk_int32_MAX >> 1 );
+    for( i = idx + psDec->LPC_order; i < psDec->ltp_mem_length; i++ ) {
+        sLTP_Q14[ i ] = silk_SMULWB( inv_gain_Q30, sLTP[ i ] );
+    }
+
+    /***************************/
+    /* LTP synthesis filtering */
+    /***************************/
+    for( k = 0; k < psDec->nb_subfr; k++ ) {
+        /* Set up pointer */
+        pred_lag_ptr = &sLTP_Q14[ sLTP_buf_idx - lag + LTP_ORDER / 2 ];
+        for( i = 0; i < psDec->subfr_length; i++ ) {
+            /* Unrolled loop */
+            /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */
+            LTP_pred_Q12 = 2;
+            LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[  0 ], B_Q14[ 0 ] );
+            LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -1 ], B_Q14[ 1 ] );
+            LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -2 ], B_Q14[ 2 ] );
+            LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -3 ], B_Q14[ 3 ] );
+            LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -4 ], B_Q14[ 4 ] );
+            pred_lag_ptr++;
+
+            /* Generate LPC excitation */
+            rand_seed = silk_RAND( rand_seed );
+            idx = silk_RSHIFT( rand_seed, 25 ) & RAND_BUF_MASK;
+            sLTP_Q14[ sLTP_buf_idx ] = silk_LSHIFT32( silk_SMLAWB( LTP_pred_Q12, rand_ptr[ idx ], rand_scale_Q14 ), 2 );
+            sLTP_buf_idx++;
+        }
+
+        /* Gradually reduce LTP gain */
+        for( j = 0; j < LTP_ORDER; j++ ) {
+            B_Q14[ j ] = silk_RSHIFT( silk_SMULBB( harm_Gain_Q15, B_Q14[ j ] ), 15 );
+        }
+        /* Gradually reduce excitation gain */
+        rand_scale_Q14 = silk_RSHIFT( silk_SMULBB( rand_scale_Q14, rand_Gain_Q15 ), 15 );
+
+        /* Slowly increase pitch lag */
+        psPLC->pitchL_Q8 = silk_SMLAWB( psPLC->pitchL_Q8, psPLC->pitchL_Q8, PITCH_DRIFT_FAC_Q16 );
+        psPLC->pitchL_Q8 = silk_min_32( psPLC->pitchL_Q8, silk_LSHIFT( silk_SMULBB( MAX_PITCH_LAG_MS, psDec->fs_kHz ), 8 ) );
+        lag = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 );
+    }
+
+    /***************************/
+    /* LPC synthesis filtering */
+    /***************************/
+    sLPC_Q14_ptr = &sLTP_Q14[ psDec->ltp_mem_length - MAX_LPC_ORDER ];
+
+    /* Copy LPC state */
+    silk_memcpy( sLPC_Q14_ptr, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) );
+
+    silk_assert( psDec->LPC_order >= 10 ); /* check that unrolling works */
+    for( i = 0; i < psDec->frame_length; i++ ) {
+        /* partly unrolled */
+        /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */
+        LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  1 ], A_Q12[ 0 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  2 ], A_Q12[ 1 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  3 ], A_Q12[ 2 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  4 ], A_Q12[ 3 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  5 ], A_Q12[ 4 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  6 ], A_Q12[ 5 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  7 ], A_Q12[ 6 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  8 ], A_Q12[ 7 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i -  9 ], A_Q12[ 8 ] );
+        LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] );
+        for( j = 10; j < psDec->LPC_order; j++ ) {
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - j - 1 ], A_Q12[ j ] );
+        }
+
+        /* Add prediction to LPC excitation */
+        sLPC_Q14_ptr[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT32( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], LPC_pred_Q10, 4 );
+
+        /* Scale with Gain */
+        frame[ i ] = (opus_int16)silk_SAT16( silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], prevGain_Q10[ 1 ] ), 8 ) ) );
+    }
+
+    /* Save LPC state */
+    silk_memcpy( psDec->sLPC_Q14_buf, &sLPC_Q14_ptr[ psDec->frame_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) );
+
+    /**************************************/
+    /* Update states                      */
+    /**************************************/
+    psPLC->rand_seed     = rand_seed;
+    psPLC->randScale_Q14 = rand_scale_Q14;
+    for( i = 0; i < MAX_NB_SUBFR; i++ ) {
+        psDecCtrl->pitchL[ i ] = lag;
+    }
+    RESTORE_STACK;
+}
+
+/* Glues concealed frames with new good recieved frames */
+void silk_PLC_glue_frames(
+    silk_decoder_state                  *psDec,             /* I/O decoder state        */
+    opus_int16                          frame[],            /* I/O signal               */
+    opus_int                            length              /* I length of signal       */
+)
+{
+    opus_int   i, energy_shift;
+    opus_int32 energy;
+    silk_PLC_struct *psPLC;
+    psPLC = &psDec->sPLC;
+
+    if( psDec->lossCnt ) {
+        /* Calculate energy in concealed residual */
+        silk_sum_sqr_shift( &psPLC->conc_energy, &psPLC->conc_energy_shift, frame, length );
+
+        psPLC->last_frame_lost = 1;
+    } else {
+        if( psDec->sPLC.last_frame_lost ) {
+            /* Calculate residual in decoded signal if last frame was lost */
+            silk_sum_sqr_shift( &energy, &energy_shift, frame, length );
+
+            /* Normalize energies */
+            if( energy_shift > psPLC->conc_energy_shift ) {
+                psPLC->conc_energy = silk_RSHIFT( psPLC->conc_energy, energy_shift - psPLC->conc_energy_shift );
+            } else if( energy_shift < psPLC->conc_energy_shift ) {
+                energy = silk_RSHIFT( energy, psPLC->conc_energy_shift - energy_shift );
+            }
+
+            /* Fade in the energy difference */
+            if( energy > psPLC->conc_energy ) {
+                opus_int32 frac_Q24, LZ;
+                opus_int32 gain_Q16, slope_Q16;
+
+                LZ = silk_CLZ32( psPLC->conc_energy );
+                LZ = LZ - 1;
+                psPLC->conc_energy = silk_LSHIFT( psPLC->conc_energy, LZ );
+                energy = silk_RSHIFT( energy, silk_max_32( 24 - LZ, 0 ) );
+
+                frac_Q24 = silk_DIV32( psPLC->conc_energy, silk_max( energy, 1 ) );
+
+                gain_Q16 = silk_LSHIFT( silk_SQRT_APPROX( frac_Q24 ), 4 );
+                slope_Q16 = silk_DIV32_16( ( (opus_int32)1 << 16 ) - gain_Q16, length );
+                /* Make slope 4x steeper to avoid missing onsets after DTX */
+                slope_Q16 = silk_LSHIFT( slope_Q16, 2 );
+
+                for( i = 0; i < length; i++ ) {
+                    frame[ i ] = silk_SMULWB( gain_Q16, frame[ i ] );
+                    gain_Q16 += slope_Q16;
+                    if( gain_Q16 > (opus_int32)1 << 16 ) {
+                        break;
+                    }
+                }
+            }
+        }
+        psPLC->last_frame_lost = 0;
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/PLC.h b/lib/rbcodec/codecs/libopus/silk/PLC.h
new file mode 100644
index 0000000000..1d2d9061d9
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/PLC.h
@@ -0,0 +1,61 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_PLC_H
+#define SILK_PLC_H
+
+#include "main.h"
+
+#define BWE_COEF                        0.99
+#define V_PITCH_GAIN_START_MIN_Q14      11469               /* 0.7 in Q14               */
+#define V_PITCH_GAIN_START_MAX_Q14      15565               /* 0.95 in Q14              */
+#define MAX_PITCH_LAG_MS                18
+#define RAND_BUF_SIZE                   128
+#define RAND_BUF_MASK                   ( RAND_BUF_SIZE - 1 )
+#define LOG2_INV_LPC_GAIN_HIGH_THRES    3                   /* 2^3 = 8 dB LPC gain      */
+#define LOG2_INV_LPC_GAIN_LOW_THRES     8                   /* 2^8 = 24 dB LPC gain     */
+#define PITCH_DRIFT_FAC_Q16             655                 /* 0.01 in Q16              */
+
+void silk_PLC_Reset(
+    silk_decoder_state                  *psDec              /* I/O Decoder state        */
+);
+
+void silk_PLC(
+    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
+    silk_decoder_control                *psDecCtrl,         /* I/O Decoder control      */
+    opus_int16                          frame[],            /* I/O  signal              */
+    opus_int                            lost                /* I Loss flag              */
+);
+
+void silk_PLC_glue_frames(
+    silk_decoder_state                  *psDec,             /* I/O decoder state        */
+    opus_int16                          frame[],            /* I/O signal               */
+    opus_int                            length              /* I length of signal       */
+);
+
+#endif
+
diff --git a/lib/rbcodec/codecs/libopus/silk/SigProc_FIX.h b/lib/rbcodec/codecs/libopus/silk/SigProc_FIX.h
new file mode 100644
index 0000000000..72ec26a67f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/SigProc_FIX.h
@@ -0,0 +1,589 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_SIGPROC_FIX_H
+#define SILK_SIGPROC_FIX_H
+
+#ifdef  __cplusplus
+extern "C"
+{
+#endif
+
+/*#define silk_MACRO_COUNT */          /* Used to enable WMOPS counting */
+
+#define SILK_MAX_ORDER_LPC            16            /* max order of the LPC analysis in schur() and k2a() */
+
+#include <string.h>                                 /* for memset(), memcpy(), memmove() */
+#include "typedef.h"
+#include "resampler_structs.h"
+#include "macros.h"
+
+
+/********************************************************************/
+/*                    SIGNAL PROCESSING FUNCTIONS                   */
+/********************************************************************/
+
+/*!
+ * Initialize/reset the resampler state for a given pair of input/output sampling rates
+*/
+opus_int silk_resampler_init(
+    silk_resampler_state_struct *S,                 /* I/O  Resampler state                                             */
+    opus_int32                  Fs_Hz_in,           /* I    Input sampling rate (Hz)                                    */
+    opus_int32                  Fs_Hz_out,          /* I    Output sampling rate (Hz)                                   */
+    opus_int                    forEnc              /* I    If 1: encoder; if 0: decoder                                */
+);
+
+/*!
+ * Resampler: convert from one sampling rate to another
+ */
+opus_int silk_resampler(
+    silk_resampler_state_struct *S,                 /* I/O  Resampler state                                             */
+    opus_int16                  out[],              /* O    Output signal                                               */
+    const opus_int16            in[],               /* I    Input signal                                                */
+    opus_int32                  inLen               /* I    Number of input samples                                     */
+);
+
+/*!
+* Downsample 2x, mediocre quality
+*/
+void silk_resampler_down2(
+    opus_int32                  *S,                 /* I/O  State vector [ 2 ]                                          */
+    opus_int16                  *out,               /* O    Output signal [ len ]                                       */
+    const opus_int16            *in,                /* I    Input signal [ floor(len/2) ]                               */
+    opus_int32                  inLen               /* I    Number of input samples                                     */
+);
+
+/*!
+ * Downsample by a factor 2/3, low quality
+*/
+void silk_resampler_down2_3(
+    opus_int32                  *S,                 /* I/O  State vector [ 6 ]                                          */
+    opus_int16                  *out,               /* O    Output signal [ floor(2*inLen/3) ]                          */
+    const opus_int16            *in,                /* I    Input signal [ inLen ]                                      */
+    opus_int32                  inLen               /* I    Number of input samples                                     */
+);
+
+/*!
+ * second order ARMA filter;
+ * slower than biquad() but uses more precise coefficients
+ * can handle (slowly) varying coefficients
+ */
+void silk_biquad_alt(
+    const opus_int16            *in,                /* I     input signal                                               */
+    const opus_int32            *B_Q28,             /* I     MA coefficients [3]                                        */
+    const opus_int32            *A_Q28,             /* I     AR coefficients [2]                                        */
+    opus_int32                  *S,                 /* I/O   State vector [2]                                           */
+    opus_int16                  *out,               /* O     output signal                                              */
+    const opus_int32            len,                /* I     signal length (must be even)                               */
+    opus_int                    stride              /* I     Operate on interleaved signal if > 1                       */
+);
+
+/* Variable order MA prediction error filter. */
+void silk_LPC_analysis_filter(
+    opus_int16                  *out,               /* O    Output signal                                               */
+    const opus_int16            *in,                /* I    Input signal                                                */
+    const opus_int16            *B,                 /* I    MA prediction coefficients, Q12 [order]                     */
+    const opus_int32            len,                /* I    Signal length                                               */
+    const opus_int32            d                   /* I    Filter order                                                */
+);
+
+/* Chirp (bandwidth expand) LP AR filter */
+void silk_bwexpander(
+    opus_int16                  *ar,                /* I/O  AR filter to be expanded (without leading 1)                */
+    const opus_int              d,                  /* I    Length of ar                                                */
+    opus_int32                  chirp_Q16           /* I    Chirp factor (typically in the range 0 to 1)                */
+);
+
+/* Chirp (bandwidth expand) LP AR filter */
+void silk_bwexpander_32(
+    opus_int32                  *ar,                /* I/O  AR filter to be expanded (without leading 1)                */
+    const opus_int              d,                  /* I    Length of ar                                                */
+    opus_int32                  chirp_Q16           /* I    Chirp factor in Q16                                         */
+);
+
+/* Compute inverse of LPC prediction gain, and                           */
+/* test if LPC coefficients are stable (all poles within unit circle)    */
+opus_int32 silk_LPC_inverse_pred_gain(              /* O   Returns inverse prediction gain in energy domain, Q30        */
+    const opus_int16            *A_Q12,             /* I   Prediction coefficients, Q12 [order]                         */
+    const opus_int              order               /* I   Prediction order                                             */
+);
+
+/* For input in Q24 domain */
+opus_int32 silk_LPC_inverse_pred_gain_Q24(          /* O    Returns inverse prediction gain in energy domain, Q30       */
+    const opus_int32            *A_Q24,             /* I    Prediction coefficients [order]                             */
+    const opus_int              order               /* I    Prediction order                                            */
+);
+
+/* Split signal in two decimated bands using first-order allpass filters */
+void silk_ana_filt_bank_1(
+    const opus_int16            *in,                /* I    Input signal [N]                                            */
+    opus_int32                  *S,                 /* I/O  State vector [2]                                            */
+    opus_int16                  *outL,              /* O    Low band [N/2]                                              */
+    opus_int16                  *outH,              /* O    High band [N/2]                                             */
+    const opus_int32            N                   /* I    Number of input samples                                     */
+);
+
+/********************************************************************/
+/*                        SCALAR FUNCTIONS                          */
+/********************************************************************/
+
+/* Approximation of 128 * log2() (exact inverse of approx 2^() below) */
+/* Convert input to a log scale    */
+opus_int32 silk_lin2log(
+    const opus_int32            inLin               /* I  input in linear scale                                         */
+);
+
+/* Approximation of a sigmoid function */
+opus_int silk_sigm_Q15(
+    opus_int                    in_Q5               /* I                                                                */
+);
+
+/* Approximation of 2^() (exact inverse of approx log2() above) */
+/* Convert input to a linear scale */
+opus_int32 silk_log2lin(
+    const opus_int32            inLog_Q7            /* I  input on log scale                                            */
+);
+
+/* Function that returns the maximum absolut value of the input vector */
+opus_int16 silk_int16_array_maxabs(                 /* O   Maximum absolute value, max: 2^15-1                          */
+    const opus_int16            *vec,               /* I   Input vector  [len]                                          */
+    const opus_int32            len                 /* I   Length of input vector                                       */
+);
+
+/* Compute number of bits to right shift the sum of squares of a vector    */
+/* of int16s to make it fit in an int32                                    */
+void silk_sum_sqr_shift(
+    opus_int32                  *energy,            /* O   Energy of x, after shifting to the right                     */
+    opus_int                    *shift,             /* O   Number of bits right shift applied to energy                 */
+    const opus_int16            *x,                 /* I   Input vector                                                 */
+    opus_int                    len                 /* I   Length of input vector                                       */
+);
+
+/* Calculates the reflection coefficients from the correlation sequence    */
+/* Faster than schur64(), but much less accurate.                          */
+/* uses SMLAWB(), requiring armv5E and higher.                             */
+opus_int32 silk_schur(                              /* O    Returns residual energy                                     */
+    opus_int16                  *rc_Q15,            /* O    reflection coefficients [order] Q15                         */
+    const opus_int32            *c,                 /* I    correlations [order+1]                                      */
+    const opus_int32            order               /* I    prediction order                                            */
+);
+
+/* Calculates the reflection coefficients from the correlation sequence    */
+/* Slower than schur(), but more accurate.                                 */
+/* Uses SMULL(), available on armv4                                        */
+opus_int32 silk_schur64(                            /* O    returns residual energy                                     */
+    opus_int32                  rc_Q16[],           /* O    Reflection coefficients [order] Q16                         */
+    const opus_int32            c[],                /* I    Correlations [order+1]                                      */
+    opus_int32                  order               /* I    Prediction order                                            */
+);
+
+/* Step up function, converts reflection coefficients to prediction coefficients */
+void silk_k2a(
+    opus_int32                  *A_Q24,             /* O    Prediction coefficients [order] Q24                         */
+    const opus_int16            *rc_Q15,            /* I    Reflection coefficients [order] Q15                         */
+    const opus_int32            order               /* I    Prediction order                                            */
+);
+
+/* Step up function, converts reflection coefficients to prediction coefficients */
+void silk_k2a_Q16(
+    opus_int32                  *A_Q24,             /* O    Prediction coefficients [order] Q24                         */
+    const opus_int32            *rc_Q16,            /* I    Reflection coefficients [order] Q16                         */
+    const opus_int32            order               /* I    Prediction order                                            */
+);
+
+/* Apply sine window to signal vector.                              */
+/* Window types:                                                    */
+/*    1 -> sine window from 0 to pi/2                               */
+/*    2 -> sine window from pi/2 to pi                              */
+/* every other sample of window is linearly interpolated, for speed */
+void silk_apply_sine_window(
+    opus_int16                  px_win[],           /* O    Pointer to windowed signal                                  */
+    const opus_int16            px[],               /* I    Pointer to input signal                                     */
+    const opus_int              win_type,           /* I    Selects a window type                                       */
+    const opus_int              length              /* I    Window length, multiple of 4                                */
+);
+
+/* Compute autocorrelation */
+void silk_autocorr(
+    opus_int32                  *results,           /* O    Result (length correlationCount)                            */
+    opus_int                    *scale,             /* O    Scaling of the correlation vector                           */
+    const opus_int16            *inputData,         /* I    Input data to correlate                                     */
+    const opus_int              inputDataSize,      /* I    Length of input                                             */
+    const opus_int              correlationCount    /* I    Number of correlation taps to compute                       */
+);
+
+void silk_decode_pitch(
+    opus_int16                  lagIndex,           /* I                                                                */
+    opus_int8                   contourIndex,       /* O                                                                */
+    opus_int                    pitch_lags[],       /* O    4 pitch values                                              */
+    const opus_int              Fs_kHz,             /* I    sampling frequency (kHz)                                    */
+    const opus_int              nb_subfr            /* I    number of sub frames                                        */
+);
+
+opus_int silk_pitch_analysis_core(                  /* O    Voicing estimate: 0 voiced, 1 unvoiced                      */
+    const opus_int16            *frame,             /* I    Signal of length PE_FRAME_LENGTH_MS*Fs_kHz                  */
+    opus_int                    *pitch_out,         /* O    4 pitch lag values                                          */
+    opus_int16                  *lagIndex,          /* O    Lag Index                                                   */
+    opus_int8                   *contourIndex,      /* O    Pitch contour Index                                         */
+    opus_int                    *LTPCorr_Q15,       /* I/O  Normalized correlation; input: value from previous frame    */
+    opus_int                    prevLag,            /* I    Last lag of previous frame; set to zero is unvoiced         */
+    const opus_int32            search_thres1_Q16,  /* I    First stage threshold for lag candidates 0 - 1              */
+    const opus_int              search_thres2_Q15,  /* I    Final threshold for lag candidates 0 - 1                    */
+    const opus_int              Fs_kHz,             /* I    Sample frequency (kHz)                                      */
+    const opus_int              complexity,         /* I    Complexity setting, 0-2, where 2 is highest                 */
+    const opus_int              nb_subfr            /* I    number of 5 ms subframes                                    */
+);
+
+/* Compute Normalized Line Spectral Frequencies (NLSFs) from whitening filter coefficients      */
+/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */
+void silk_A2NLSF(
+    opus_int16                  *NLSF,              /* O    Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */
+    opus_int32                  *a_Q16,             /* I/O  Monic whitening filter coefficients in Q16 [d]              */
+    const opus_int              d                   /* I    Filter order (must be even)                                 */
+);
+
+/* compute whitening filter coefficients from normalized line spectral frequencies */
+void silk_NLSF2A(
+    opus_int16                  *a_Q12,             /* O    monic whitening filter coefficients in Q12,  [ d ]          */
+    const opus_int16            *NLSF,              /* I    normalized line spectral frequencies in Q15, [ d ]          */
+    const opus_int              d                   /* I    filter order (should be even)                               */
+);
+
+void silk_insertion_sort_increasing(
+    opus_int32                  *a,                 /* I/O   Unsorted / Sorted vector                                   */
+    opus_int                    *idx,               /* O     Index vector for the sorted elements                       */
+    const opus_int              L,                  /* I     Vector length                                              */
+    const opus_int              K                   /* I     Number of correctly sorted positions                       */
+);
+
+void silk_insertion_sort_decreasing_int16(
+    opus_int16                  *a,                 /* I/O   Unsorted / Sorted vector                                   */
+    opus_int                    *idx,               /* O     Index vector for the sorted elements                       */
+    const opus_int              L,                  /* I     Vector length                                              */
+    const opus_int              K                   /* I     Number of correctly sorted positions                       */
+);
+
+void silk_insertion_sort_increasing_all_values_int16(
+     opus_int16                 *a,                 /* I/O   Unsorted / Sorted vector                                   */
+     const opus_int             L                   /* I     Vector length                                              */
+);
+
+/* NLSF stabilizer, for a single input data vector */
+void silk_NLSF_stabilize(
+          opus_int16            *NLSF_Q15,          /* I/O   Unstable/stabilized normalized LSF vector in Q15 [L]       */
+    const opus_int16            *NDeltaMin_Q15,     /* I     Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1]   */
+    const opus_int              L                   /* I     Number of NLSF parameters in the input vector              */
+);
+
+/* Laroia low complexity NLSF weights */
+void silk_NLSF_VQ_weights_laroia(
+    opus_int16                  *pNLSFW_Q_OUT,      /* O     Pointer to input vector weights [D]                        */
+    const opus_int16            *pNLSF_Q15,         /* I     Pointer to input vector         [D]                        */
+    const opus_int              D                   /* I     Input vector dimension (even)                              */
+);
+
+/* Compute reflection coefficients from input signal */
+void silk_burg_modified(
+    opus_int32                  *res_nrg,           /* O    Residual energy                                             */
+    opus_int                    *res_nrg_Q,         /* O    Residual energy Q value                                     */
+    opus_int32                  A_Q16[],            /* O    Prediction coefficients (length order)                      */
+    const opus_int16            x[],                /* I    Input signal, length: nb_subfr * ( D + subfr_length )       */
+    const opus_int32            minInvGain_Q30,     /* I    Inverse of max prediction gain                              */
+    const opus_int              subfr_length,       /* I    Input signal subframe length (incl. D preceeding samples)   */
+    const opus_int              nb_subfr,           /* I    Number of subframes stacked in x                            */
+    const opus_int              D                   /* I    Order                                                       */
+);
+
+/* Copy and multiply a vector by a constant */
+void silk_scale_copy_vector16(
+    opus_int16                  *data_out,
+    const opus_int16            *data_in,
+    opus_int32                  gain_Q16,           /* I    Gain in Q16                                                 */
+    const opus_int              dataSize            /* I    Length                                                      */
+);
+
+/* Some for the LTP related function requires Q26 to work.*/
+void silk_scale_vector32_Q26_lshift_18(
+    opus_int32                  *data1,             /* I/O  Q0/Q18                                                      */
+    opus_int32                  gain_Q26,           /* I    Q26                                                         */
+    opus_int                    dataSize            /* I    length                                                      */
+);
+
+/********************************************************************/
+/*                        INLINE ARM MATH                           */
+/********************************************************************/
+
+/*    return sum( inVec1[i] * inVec2[i] ) */
+opus_int32 silk_inner_prod_aligned(
+    const opus_int16 *const     inVec1,             /*    I input vector 1                                              */
+    const opus_int16 *const     inVec2,             /*    I input vector 2                                              */
+    const opus_int              len                 /*    I vector lengths                                              */
+);
+
+opus_int32 silk_inner_prod_aligned_scale(
+    const opus_int16 *const     inVec1,             /*    I input vector 1                                              */
+    const opus_int16 *const     inVec2,             /*    I input vector 2                                              */
+    const opus_int              scale,              /*    I number of bits to shift                                     */
+    const opus_int              len                 /*    I vector lengths                                              */
+);
+
+opus_int64 silk_inner_prod16_aligned_64(
+    const opus_int16            *inVec1,            /*    I input vector 1                                              */
+    const opus_int16            *inVec2,            /*    I input vector 2                                              */
+    const opus_int              len                 /*    I vector lengths                                              */
+);
+
+/********************************************************************/
+/*                                MACROS                            */
+/********************************************************************/
+
+/* Rotate a32 right by 'rot' bits. Negative rot values result in rotating
+   left. Output is 32bit int.
+   Note: contemporary compilers recognize the C expression below and
+   compile it into a 'ror' instruction if available. No need for inline ASM! */
+static inline opus_int32 silk_ROR32( opus_int32 a32, opus_int rot )
+{
+    opus_uint32 x = (opus_uint32) a32;
+    opus_uint32 r = (opus_uint32) rot;
+    opus_uint32 m = (opus_uint32) -rot;
+    if( rot == 0 ) {
+        return a32;
+    } else if( rot < 0 ) {
+        return (opus_int32) ((x << m) | (x >> (32 - m)));
+    } else {
+        return (opus_int32) ((x << (32 - r)) | (x >> r));
+    }
+}
+
+/* Allocate opus_int16 alligned to 4-byte memory address */
+#if EMBEDDED_ARM
+#define silk_DWORD_ALIGN __attribute__((aligned(4)))
+#else
+#define silk_DWORD_ALIGN
+#endif
+
+/* Useful Macros that can be adjusted to other platforms */
+#define silk_memcpy(dest, src, size)        memcpy((dest), (src), (size))
+#define silk_memset(dest, src, size)        memset((dest), (src), (size))
+#define silk_memmove(dest, src, size)       memmove((dest), (src), (size))
+
+/* Fixed point macros */
+
+/* (a32 * b32) output have to be 32bit int */
+#define silk_MUL(a32, b32)                  ((a32) * (b32))
+
+/* (a32 * b32) output have to be 32bit uint */
+#define silk_MUL_uint(a32, b32)             silk_MUL(a32, b32)
+
+/* a32 + (b32 * c32) output have to be 32bit int */
+#define silk_MLA(a32, b32, c32)             silk_ADD32((a32),((b32) * (c32)))
+
+/* a32 + (b32 * c32) output have to be 32bit uint */
+#define silk_MLA_uint(a32, b32, c32)        silk_MLA(a32, b32, c32)
+
+/* ((a32 >> 16)  * (b32 >> 16)) output have to be 32bit int */
+#define silk_SMULTT(a32, b32)               (((a32) >> 16) * ((b32) >> 16))
+
+/* a32 + ((a32 >> 16)  * (b32 >> 16)) output have to be 32bit int */
+#define silk_SMLATT(a32, b32, c32)          silk_ADD32((a32),((b32) >> 16) * ((c32) >> 16))
+
+#define silk_SMLALBB(a64, b16, c16)         silk_ADD64((a64),(opus_int64)((opus_int32)(b16) * (opus_int32)(c16)))
+
+/* (a32 * b32) */
+#define silk_SMULL(a32, b32)                ((opus_int64)(a32) * /*(opus_int64)*/(b32))
+
+/* Adds two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour
+   (just standard two's complement implementation-specific behaviour) */
+#define silk_ADD32_ovflw(a, b)              ((opus_int32)((opus_uint32)(a) + (opus_uint32)(b)))
+/* Subtractss two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour
+   (just standard two's complement implementation-specific behaviour) */
+#define silk_SUB32_ovflw(a, b)              ((opus_int32)((opus_uint32)(a) - (opus_uint32)(b)))
+
+/* Multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode) */
+#define silk_MLA_ovflw(a32, b32, c32)       silk_ADD32_ovflw((a32), (opus_uint32)(b32) * (opus_uint32)(c32))
+#define silk_SMLABB_ovflw(a32, b32, c32)    (silk_ADD32_ovflw((a32) , ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32))))
+
+#define silk_DIV32_16(a32, b16)             ((opus_int32)((a32) / (b16)))
+#define silk_DIV32(a32, b32)                ((opus_int32)((a32) / (b32)))
+
+/* These macros enables checking for overflow in silk_API_Debug.h*/
+#define silk_ADD16(a, b)                    ((a) + (b))
+#define silk_ADD32(a, b)                    ((a) + (b))
+#define silk_ADD64(a, b)                    ((a) + (b))
+
+#define silk_SUB16(a, b)                    ((a) - (b))
+#define silk_SUB32(a, b)                    ((a) - (b))
+#define silk_SUB64(a, b)                    ((a) - (b))
+
+#define silk_SAT8(a)                        ((a) > silk_int8_MAX ? silk_int8_MAX  :       \
+                                            ((a) < silk_int8_MIN ? silk_int8_MIN  : (a)))
+#define silk_SAT16(a)                       ((a) > silk_int16_MAX ? silk_int16_MAX :      \
+                                            ((a) < silk_int16_MIN ? silk_int16_MIN : (a)))
+#define silk_SAT32(a)                       ((a) > silk_int32_MAX ? silk_int32_MAX :      \
+                                            ((a) < silk_int32_MIN ? silk_int32_MIN : (a)))
+
+#define silk_CHECK_FIT8(a)                  (a)
+#define silk_CHECK_FIT16(a)                 (a)
+#define silk_CHECK_FIT32(a)                 (a)
+
+#define silk_ADD_SAT16(a, b)                (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a), (b) ) )
+#define silk_ADD_SAT64(a, b)                ((((a) + (b)) & 0x8000000000000000LL) == 0 ?                            \
+                                            ((((a) & (b)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a)+(b)) : \
+                                            ((((a) | (b)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a)+(b)) )
+
+#define silk_SUB_SAT16(a, b)                (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a), (b) ) )
+#define silk_SUB_SAT64(a, b)                ((((a)-(b)) & 0x8000000000000000LL) == 0 ?                                               \
+                                            (( (a) & ((b)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a)-(b)) : \
+                                            ((((a)^0x8000000000000000LL) & (b)  & 0x8000000000000000LL) ? silk_int64_MAX : (a)-(b)) )
+
+/* Saturation for positive input values */
+#define silk_POS_SAT32(a)                   ((a) > silk_int32_MAX ? silk_int32_MAX : (a))
+
+/* Add with saturation for positive input values */
+#define silk_ADD_POS_SAT8(a, b)             ((((a)+(b)) & 0x80)                 ? silk_int8_MAX  : ((a)+(b)))
+#define silk_ADD_POS_SAT16(a, b)            ((((a)+(b)) & 0x8000)               ? silk_int16_MAX : ((a)+(b)))
+#define silk_ADD_POS_SAT32(a, b)            ((((a)+(b)) & 0x80000000)           ? silk_int32_MAX : ((a)+(b)))
+#define silk_ADD_POS_SAT64(a, b)            ((((a)+(b)) & 0x8000000000000000LL) ? silk_int64_MAX : ((a)+(b)))
+
+#define silk_LSHIFT8(a, shift)              ((opus_int8)((opus_uint8)(a)<<(shift)))         /* shift >= 0, shift < 8  */
+#define silk_LSHIFT16(a, shift)             ((opus_int16)((opus_uint16)(a)<<(shift)))       /* shift >= 0, shift < 16 */
+#define silk_LSHIFT32(a, shift)             ((opus_int32)((opus_uint32)(a)<<(shift)))       /* shift >= 0, shift < 32 */
+#define silk_LSHIFT64(a, shift)             ((opus_int64)((opus_uint64)(a)<<(shift)))       /* shift >= 0, shift < 64 */
+#define silk_LSHIFT(a, shift)               silk_LSHIFT32(a, shift)                         /* shift >= 0, shift < 32 */
+
+#define silk_RSHIFT8(a, shift)              ((a)>>(shift))                                  /* shift >= 0, shift < 8  */
+#define silk_RSHIFT16(a, shift)             ((a)>>(shift))                                  /* shift >= 0, shift < 16 */
+#define silk_RSHIFT32(a, shift)             ((a)>>(shift))                                  /* shift >= 0, shift < 32 */
+#define silk_RSHIFT64(a, shift)             ((a)>>(shift))                                  /* shift >= 0, shift < 64 */
+#define silk_RSHIFT(a, shift)               silk_RSHIFT32(a, shift)                         /* shift >= 0, shift < 32 */
+
+/* saturates before shifting */
+#define silk_LSHIFT_SAT32(a, shift)         (silk_LSHIFT32( silk_LIMIT( (a), silk_RSHIFT32( silk_int32_MIN, (shift) ), \
+                                                    silk_RSHIFT32( silk_int32_MAX, (shift) ) ), (shift) ))
+
+#define silk_LSHIFT_ovflw(a, shift)         ((opus_int32)((opus_uint32)(a) << (shift)))     /* shift >= 0, allowed to overflow */
+#define silk_LSHIFT_uint(a, shift)          ((a) << (shift))                                /* shift >= 0 */
+#define silk_RSHIFT_uint(a, shift)          ((a) >> (shift))                                /* shift >= 0 */
+
+#define silk_ADD_LSHIFT(a, b, shift)        ((a) + silk_LSHIFT((b), (shift)))               /* shift >= 0 */
+#define silk_ADD_LSHIFT32(a, b, shift)      silk_ADD32((a), silk_LSHIFT32((b), (shift)))    /* shift >= 0 */
+#define silk_ADD_LSHIFT_uint(a, b, shift)   ((a) + silk_LSHIFT_uint((b), (shift)))          /* shift >= 0 */
+#define silk_ADD_RSHIFT(a, b, shift)        ((a) + silk_RSHIFT((b), (shift)))               /* shift >= 0 */
+#define silk_ADD_RSHIFT32(a, b, shift)      silk_ADD32((a), silk_RSHIFT32((b), (shift)))    /* shift >= 0 */
+#define silk_ADD_RSHIFT_uint(a, b, shift)   ((a) + silk_RSHIFT_uint((b), (shift)))          /* shift >= 0 */
+#define silk_SUB_LSHIFT32(a, b, shift)      silk_SUB32((a), silk_LSHIFT32((b), (shift)))    /* shift >= 0 */
+#define silk_SUB_RSHIFT32(a, b, shift)      silk_SUB32((a), silk_RSHIFT32((b), (shift)))    /* shift >= 0 */
+
+/* Requires that shift > 0 */
+#define silk_RSHIFT_ROUND(a, shift)         ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1)
+#define silk_RSHIFT_ROUND64(a, shift)       ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1)
+
+/* Number of rightshift required to fit the multiplication */
+#define silk_NSHIFT_MUL_32_32(a, b)         ( -(31- (32-silk_CLZ32(silk_abs(a)) + (32-silk_CLZ32(silk_abs(b))))) )
+#define silk_NSHIFT_MUL_16_16(a, b)         ( -(15- (16-silk_CLZ16(silk_abs(a)) + (16-silk_CLZ16(silk_abs(b))))) )
+
+
+#define silk_min(a, b)                      (((a) < (b)) ? (a) : (b))
+#define silk_max(a, b)                      (((a) > (b)) ? (a) : (b))
+
+/* Macro to convert floating-point constants to fixed-point */
+#define SILK_FIX_CONST( C, Q )              ((opus_int32)((C) * ((opus_int64)1 << (Q)) + 0.5))
+
+/* silk_min() versions with typecast in the function call */
+static inline opus_int silk_min_int(opus_int a, opus_int b)
+{
+    return (((a) < (b)) ? (a) : (b));
+}
+static inline opus_int16 silk_min_16(opus_int16 a, opus_int16 b)
+{
+    return (((a) < (b)) ? (a) : (b));
+}
+static inline opus_int32 silk_min_32(opus_int32 a, opus_int32 b)
+{
+    return (((a) < (b)) ? (a) : (b));
+}
+static inline opus_int64 silk_min_64(opus_int64 a, opus_int64 b)
+{
+    return (((a) < (b)) ? (a) : (b));
+}
+
+/* silk_min() versions with typecast in the function call */
+static inline opus_int silk_max_int(opus_int a, opus_int b)
+{
+    return (((a) > (b)) ? (a) : (b));
+}
+static inline opus_int16 silk_max_16(opus_int16 a, opus_int16 b)
+{
+    return (((a) > (b)) ? (a) : (b));
+}
+static inline opus_int32 silk_max_32(opus_int32 a, opus_int32 b)
+{
+    return (((a) > (b)) ? (a) : (b));
+}
+static inline opus_int64 silk_max_64(opus_int64 a, opus_int64 b)
+{
+    return (((a) > (b)) ? (a) : (b));
+}
+
+#define silk_LIMIT( a, limit1, limit2)      ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \
+                                                                 : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a))))
+
+#define silk_LIMIT_int                      silk_LIMIT
+#define silk_LIMIT_16                       silk_LIMIT
+#define silk_LIMIT_32                       silk_LIMIT
+
+#define silk_abs(a)                         (((a) >  0)  ? (a) : -(a))            /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */
+#define silk_abs_int(a)                     (((a) ^ ((a) >> (8 * sizeof(a) - 1))) - ((a) >> (8 * sizeof(a) - 1)))
+#define silk_abs_int32(a)                   (((a) ^ ((a) >> 31)) - ((a) >> 31))
+#define silk_abs_int64(a)                   (((a) >  0)  ? (a) : -(a))
+
+#define silk_sign(a)                        ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 ))
+
+/* PSEUDO-RANDOM GENERATOR                                                          */
+/* Make sure to store the result as the seed for the next call (also in between     */
+/* frames), otherwise result won't be random at all. When only using some of the    */
+/* bits, take the most significant bits by right-shifting.                          */
+#define silk_RAND(seed)                     (silk_MLA_ovflw(907633515, (seed), 196314165))
+
+/*  Add some multiplication functions that can be easily mapped to ARM. */
+
+/*    silk_SMMUL: Signed top word multiply.
+          ARMv6        2 instruction cycles.
+          ARMv3M+      3 instruction cycles. use SMULL and ignore LSB registers.(except xM)*/
+/*#define silk_SMMUL(a32, b32)                (opus_int32)silk_RSHIFT(silk_SMLAL(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)), 16)*/
+/* the following seems faster on x86 */
+#define silk_SMMUL(a32, b32)                (opus_int32)silk_RSHIFT64(silk_SMULL((a32), (b32)), 32)
+
+#include "Inlines.h"
+#include "MacroCount.h"
+#include "MacroDebug.h"
+
+#ifdef  __cplusplus
+}
+#endif
+
+#endif /* SILK_SIGPROC_FIX_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/bwexpander.c b/lib/rbcodec/codecs/libopus/silk/bwexpander.c
new file mode 100644
index 0000000000..9bb5f62810
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/bwexpander.c
@@ -0,0 +1,51 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+/* Chirp (bandwidth expand) LP AR filter */
+void silk_bwexpander(
+    opus_int16                  *ar,                /* I/O  AR filter to be expanded (without leading 1)                */
+    const opus_int              d,                  /* I    Length of ar                                                */
+    opus_int32                  chirp_Q16           /* I    Chirp factor (typically in the range 0 to 1)                */
+)
+{
+    opus_int   i;
+    opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536;
+
+    /* NB: Dont use silk_SMULWB, instead of silk_RSHIFT_ROUND( silk_MUL(), 16 ), below.  */
+    /* Bias in silk_SMULWB can lead to unstable filters                                */
+    for( i = 0; i < d - 1; i++ ) {
+        ar[ i ]    = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ i ]             ), 16 );
+        chirp_Q16 +=            silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 );
+    }
+    ar[ d - 1 ] = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ d - 1 ] ), 16 );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/bwexpander_32.c b/lib/rbcodec/codecs/libopus/silk/bwexpander_32.c
new file mode 100644
index 0000000000..fe3cc4c9d1
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/bwexpander_32.c
@@ -0,0 +1,50 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+/* Chirp (bandwidth expand) LP AR filter */
+void silk_bwexpander_32(
+    opus_int32                  *ar,                /* I/O  AR filter to be expanded (without leading 1)                */
+    const opus_int              d,                  /* I    Length of ar                                                */
+    opus_int32                  chirp_Q16           /* I    Chirp factor in Q16                                         */
+)
+{
+    opus_int   i;
+    opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536;
+
+    for( i = 0; i < d - 1; i++ ) {
+        ar[ i ]    = silk_SMULWW( chirp_Q16, ar[ i ] );
+        chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 );
+    }
+    ar[ d - 1 ] = silk_SMULWW( chirp_Q16, ar[ d - 1 ] );
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/code_signs.c b/lib/rbcodec/codecs/libopus/silk/code_signs.c
new file mode 100644
index 0000000000..3903eb1f16
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/code_signs.c
@@ -0,0 +1,115 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/*#define silk_enc_map(a)                ((a) > 0 ? 1 : 0)*/
+/*#define silk_dec_map(a)                ((a) > 0 ? 1 : -1)*/
+/* shifting avoids if-statement */
+#define silk_enc_map(a)                  ( silk_RSHIFT( (a), 15 ) + 1 )
+#define silk_dec_map(a)                  ( silk_LSHIFT( (a),  1 ) - 1 )
+
+/* Encodes signs of excitation */
+void silk_encode_signs(
+    ec_enc                      *psRangeEnc,                        /* I/O  Compressor data structure                   */
+    const opus_int8             pulses[],                           /* I    pulse signal                                */
+    opus_int                    length,                             /* I    length of input                             */
+    const opus_int              signalType,                         /* I    Signal type                                 */
+    const opus_int              quantOffsetType,                    /* I    Quantization offset type                    */
+    const opus_int              sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
+)
+{
+    opus_int         i, j, p;
+    opus_uint8       icdf[ 2 ];
+    const opus_int8  *q_ptr;
+    const opus_uint8 *icdf_ptr;
+
+    icdf[ 1 ] = 0;
+    q_ptr = pulses;
+    i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
+    icdf_ptr = &silk_sign_iCDF[ i ];
+    length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
+    for( i = 0; i < length; i++ ) {
+        p = sum_pulses[ i ];
+        if( p > 0 ) {
+            icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ];
+            for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
+                if( q_ptr[ j ] != 0 ) {
+                    ec_enc_icdf( psRangeEnc, silk_enc_map( q_ptr[ j ]), icdf, 8 );
+                }
+            }
+        }
+        q_ptr += SHELL_CODEC_FRAME_LENGTH;
+    }
+}
+
+/* Decodes signs of excitation */
+void silk_decode_signs(
+    ec_dec                      *psRangeDec,                        /* I/O  Compressor data structure                   */
+    opus_int                    pulses[],                           /* I/O  pulse signal                                */
+    opus_int                    length,                             /* I    length of input                             */
+    const opus_int              signalType,                         /* I    Signal type                                 */
+    const opus_int              quantOffsetType,                    /* I    Quantization offset type                    */
+    const opus_int              sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
+)
+{
+    opus_int         i, j, p;
+    opus_uint8       icdf[ 2 ];
+    opus_int         *q_ptr;
+    const opus_uint8 *icdf_ptr;
+
+    icdf[ 1 ] = 0;
+    q_ptr = pulses;
+    i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
+    icdf_ptr = &silk_sign_iCDF[ i ];
+    length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
+    for( i = 0; i < length; i++ ) {
+        p = sum_pulses[ i ];
+        if( p > 0 ) {
+            icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ];
+            for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
+                if( q_ptr[ j ] > 0 ) {
+                    /* attach sign */
+#if 0
+                    /* conditional implementation */
+                    if( ec_dec_icdf( psRangeDec, icdf, 8 ) == 0 ) {
+                        q_ptr[ j ] = -q_ptr[ j ];
+                    }
+#else
+                    /* implementation with shift, subtraction, multiplication */
+                    q_ptr[ j ] *= silk_dec_map( ec_dec_icdf( psRangeDec, icdf, 8 ) );
+#endif
+                }
+            }
+        }
+        q_ptr += SHELL_CODEC_FRAME_LENGTH;
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/control.h b/lib/rbcodec/codecs/libopus/silk/control.h
new file mode 100644
index 0000000000..c52ec3fe38
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/control.h
@@ -0,0 +1,139 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_CONTROL_H
+#define SILK_CONTROL_H
+
+#include "typedef.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/* Decoder API flags */
+#define FLAG_DECODE_NORMAL                      0
+#define FLAG_PACKET_LOST                        1
+#define FLAG_DECODE_LBRR                        2
+
+/***********************************************/
+/* Structure for controlling encoder operation */
+/***********************************************/
+typedef struct {
+    /* I:   Number of channels; 1/2                                                         */
+    opus_int32 nChannelsAPI;
+
+    /* I:   Number of channels; 1/2                                                         */
+    opus_int32 nChannelsInternal;
+
+    /* I:   Input signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000   */
+    opus_int32 API_sampleRate;
+
+    /* I:   Maximum internal sampling rate in Hertz; 8000/12000/16000                       */
+    opus_int32 maxInternalSampleRate;
+
+    /* I:   Minimum internal sampling rate in Hertz; 8000/12000/16000                       */
+    opus_int32 minInternalSampleRate;
+
+    /* I:   Soft request for internal sampling rate in Hertz; 8000/12000/16000              */
+    opus_int32 desiredInternalSampleRate;
+
+    /* I:   Number of samples per packet in milliseconds; 10/20/40/60                       */
+    opus_int payloadSize_ms;
+
+    /* I:   Bitrate during active speech in bits/second; internally limited                 */
+    opus_int32 bitRate;
+
+    /* I:   Uplink packet loss in percent (0-100)                                           */
+    opus_int packetLossPercentage;
+
+    /* I:   Complexity mode; 0 is lowest, 10 is highest complexity                          */
+    opus_int complexity;
+
+    /* I:   Flag to enable in-band Forward Error Correction (FEC); 0/1                      */
+    opus_int useInBandFEC;
+
+    /* I:   Flag to enable discontinuous transmission (DTX); 0/1                            */
+    opus_int useDTX;
+
+    /* I:   Flag to use constant bitrate                                                    */
+    opus_int useCBR;
+
+    /* I:   Maximum number of bits allowed for the frame                                    */
+    opus_int maxBits;
+
+    /* I:   Causes a smooth downmix to mono                                                 */
+    opus_int toMono;
+
+    /* I:   Opus encoder is allowing us to switch bandwidth                                 */
+    opus_int opusCanSwitch;
+
+    /* O:   Internal sampling rate used, in Hertz; 8000/12000/16000                         */
+    opus_int32 internalSampleRate;
+
+    /* O: Flag that bandwidth switching is allowed (because low voice activity)             */
+    opus_int allowBandwidthSwitch;
+
+    /* O:   Flag that SILK runs in WB mode without variable LP filter (use for switching between WB/SWB/FB) */
+    opus_int inWBmodeWithoutVariableLP;
+
+    /* O:   Stereo width */
+    opus_int stereoWidth_Q14;
+
+    /* O:   Tells the Opus encoder we're ready to switch                                    */
+    opus_int switchReady;
+
+} silk_EncControlStruct;
+
+/**************************************************************************/
+/* Structure for controlling decoder operation and reading decoder status */
+/**************************************************************************/
+typedef struct {
+    /* I:   Number of channels; 1/2                                                         */
+    opus_int32 nChannelsAPI;
+
+    /* I:   Number of channels; 1/2                                                         */
+    opus_int32 nChannelsInternal;
+
+    /* I:   Output signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000  */
+    opus_int32 API_sampleRate;
+
+    /* I:   Internal sampling rate used, in Hertz; 8000/12000/16000                         */
+    opus_int32 internalSampleRate;
+
+    /* I:   Number of samples per packet in milliseconds; 10/20/40/60                       */
+    opus_int payloadSize_ms;
+
+    /* O:   Pitch lag of previous frame (0 if unvoiced), measured in samples at 48 kHz      */
+    opus_int prevPitchLag;
+} silk_DecControlStruct;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/dec_API.c b/lib/rbcodec/codecs/libopus/silk/dec_API.c
new file mode 100644
index 0000000000..908e6033b1
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/dec_API.c
@@ -0,0 +1,392 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+#include "API.h"
+#include "main.h"
+#include "stack_alloc.h"
+
+/************************/
+/* Decoder Super Struct */
+/************************/
+typedef struct {
+    silk_decoder_state          channel_state[ DECODER_NUM_CHANNELS ];
+    stereo_dec_state                sStereo;
+    opus_int                         nChannelsAPI;
+    opus_int                         nChannelsInternal;
+    opus_int                         prev_decode_only_middle;
+} silk_decoder;
+
+/*********************/
+/* Decoder functions */
+/*********************/
+
+opus_int silk_Get_Decoder_Size(                         /* O    Returns error code                              */
+    opus_int                        *decSizeBytes       /* O    Number of bytes in SILK decoder state           */
+)
+{
+    opus_int ret = SILK_NO_ERROR;
+
+    *decSizeBytes = sizeof( silk_decoder );
+
+    return ret;
+}
+
+/* Reset decoder state */
+opus_int silk_InitDecoder(                              /* O    Returns error code                              */
+    void                            *decState           /* I/O  State                                           */
+)
+{
+    opus_int n, ret = SILK_NO_ERROR;
+    silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state;
+
+    for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
+        ret  = silk_init_decoder( &channel_state[ n ] );
+    }
+
+    return ret;
+}
+
+/* Decode a frame */
+opus_int silk_Decode(                                   /* O    Returns error code                              */
+    void*                           decState,           /* I/O  State                                           */
+    silk_DecControlStruct*          decControl,         /* I/O  Control Structure                               */
+    opus_int                        lostFlag,           /* I    0: no loss, 1 loss, 2 decode fec                */
+    opus_int                        newPacketFlag,      /* I    Indicates first decoder call for this packet    */
+    ec_dec                          *psRangeDec,        /* I/O  Compressor data structure                       */
+    opus_int16                      *samplesOut,        /* O    Decoded output speech vector                    */
+    opus_int32                      *nSamplesOut        /* O    Number of samples decoded                       */
+)
+{
+    opus_int   i, n, decode_only_middle = 0, ret = SILK_NO_ERROR;
+    opus_int32 nSamplesOutDec, LBRR_symbol;
+    opus_int16 *samplesOut1_tmp[ 2 ];
+    VARDECL( opus_int16, samplesOut1_tmp_storage );
+    VARDECL( opus_int16, samplesOut2_tmp );
+    opus_int32 MS_pred_Q13[ 2 ] = { 0 };
+    opus_int16 *resample_out_ptr;
+    silk_decoder *psDec = ( silk_decoder * )decState;
+    silk_decoder_state *channel_state = psDec->channel_state;
+    opus_int has_side;
+    opus_int stereo_to_mono;
+    SAVE_STACK;
+
+    /**********************************/
+    /* Test if first frame in payload */
+    /**********************************/
+    if( newPacketFlag ) {
+        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+            channel_state[ n ].nFramesDecoded = 0;  /* Used to count frames in packet */
+        }
+    }
+
+    /* If Mono -> Stereo transition in bitstream: init state of second channel */
+    if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
+        ret += silk_init_decoder( &channel_state[ 1 ] );
+    }
+
+    stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 &&
+                     ( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz );
+
+    if( channel_state[ 0 ].nFramesDecoded == 0 ) {
+        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+            opus_int fs_kHz_dec;
+            if( decControl->payloadSize_ms == 0 ) {
+                /* Assuming packet loss, use 10 ms */
+                channel_state[ n ].nFramesPerPacket = 1;
+                channel_state[ n ].nb_subfr = 2;
+            } else if( decControl->payloadSize_ms == 10 ) {
+                channel_state[ n ].nFramesPerPacket = 1;
+                channel_state[ n ].nb_subfr = 2;
+            } else if( decControl->payloadSize_ms == 20 ) {
+                channel_state[ n ].nFramesPerPacket = 1;
+                channel_state[ n ].nb_subfr = 4;
+            } else if( decControl->payloadSize_ms == 40 ) {
+                channel_state[ n ].nFramesPerPacket = 2;
+                channel_state[ n ].nb_subfr = 4;
+            } else if( decControl->payloadSize_ms == 60 ) {
+                channel_state[ n ].nFramesPerPacket = 3;
+                channel_state[ n ].nb_subfr = 4;
+            } else {
+                silk_assert( 0 );
+                RESTORE_STACK;
+                return SILK_DEC_INVALID_FRAME_SIZE;
+            }
+            fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1;
+            if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) {
+                silk_assert( 0 );
+                RESTORE_STACK;
+                return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
+            }
+            ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate );
+        }
+    }
+
+    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) {
+        silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) );
+        silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) );
+        silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
+    }
+    psDec->nChannelsAPI      = decControl->nChannelsAPI;
+    psDec->nChannelsInternal = decControl->nChannelsInternal;
+
+    if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
+        ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
+        RESTORE_STACK;
+        return( ret );
+    }
+
+    if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) {
+        /* First decoder call for this payload */
+        /* Decode VAD flags and LBRR flag */
+        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+            for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
+                channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1);
+            }
+            channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1);
+        }
+        /* Decode LBRR flags */
+        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+            silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) );
+            if( channel_state[ n ].LBRR_flag ) {
+                if( channel_state[ n ].nFramesPerPacket == 1 ) {
+                    channel_state[ n ].LBRR_flags[ 0 ] = 1;
+                } else {
+                    LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1;
+                    for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
+                        channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1;
+                    }
+                }
+            }
+        }
+
+        if( lostFlag == FLAG_DECODE_NORMAL ) {
+            /* Regular decoding: skip all LBRR data */
+            for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) {
+                for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+                    if( channel_state[ n ].LBRR_flags[ i ] ) {
+                        opus_int pulses[ MAX_FRAME_LENGTH ];
+                        opus_int condCoding;
+
+                        if( decControl->nChannelsInternal == 2 && n == 0 ) {
+                            silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
+                            if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) {
+                                silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
+                            }
+                        }
+                        /* Use conditional coding if previous frame available */
+                        if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) {
+                            condCoding = CODE_CONDITIONALLY;
+                        } else {
+                            condCoding = CODE_INDEPENDENTLY;
+                        }
+                        silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding );
+                        silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType,
+                            channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length );
+                    }
+                }
+            }
+        }
+    }
+
+    /* Get MS predictor index */
+    if( decControl->nChannelsInternal == 2 ) {
+        if(   lostFlag == FLAG_DECODE_NORMAL ||
+            ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) )
+        {
+            silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
+            /* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */
+            if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ||
+                ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) )
+            {
+                silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
+            } else {
+                decode_only_middle = 0;
+            }
+        } else {
+            for( n = 0; n < 2; n++ ) {
+                MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ];
+            }
+        }
+    }
+
+    /* Reset side channel decoder prediction memory for first frame with side coding */
+    if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) {
+        silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) );
+        silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) );
+        psDec->channel_state[ 1 ].lagPrev        = 100;
+        psDec->channel_state[ 1 ].LastGainIndex  = 10;
+        psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY;
+        psDec->channel_state[ 1 ].first_frame_after_reset = 1;
+    }
+
+    ALLOC( samplesOut1_tmp_storage,
+           decControl->nChannelsInternal*(
+               channel_state[ 0 ].frame_length + 2 ),
+           opus_int16 );
+    samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage;
+    samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage
+                           + channel_state[ 0 ].frame_length + 2;
+
+    if( lostFlag == FLAG_DECODE_NORMAL ) {
+        has_side = !decode_only_middle;
+    } else {
+        has_side = !psDec->prev_decode_only_middle
+              || (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 );
+    }
+    /* Call decoder for one frame */
+    for( n = 0; n < decControl->nChannelsInternal; n++ ) {
+        if( n == 0 || has_side ) {
+            opus_int FrameIndex;
+            opus_int condCoding;
+
+            FrameIndex = channel_state[ 0 ].nFramesDecoded - n;
+            /* Use independent coding if no previous frame available */
+            if( FrameIndex <= 0 ) {
+                condCoding = CODE_INDEPENDENTLY;
+            } else if( lostFlag == FLAG_DECODE_LBRR ) {
+                condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY;
+            } else if( n > 0 && psDec->prev_decode_only_middle ) {
+                /* If we skipped a side frame in this packet, we don't
+                   need LTP scaling; the LTP state is well-defined. */
+                condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
+            } else {
+                condCoding = CODE_CONDITIONALLY;
+            }
+            ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding);
+        } else {
+            silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
+        }
+        channel_state[ n ].nFramesDecoded++;
+    }
+
+    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
+        /* Convert Mid/Side to Left/Right */
+        silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
+    } else {
+        /* Buffering */
+        silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
+        silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
+    }
+
+    /* Number of output samples */
+    *nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) );
+
+    /* Set up pointers to temp buffers */
+    ALLOC( samplesOut2_tmp,
+           decControl->nChannelsAPI == 2 ? *nSamplesOut : 0, opus_int16 );
+    if( decControl->nChannelsAPI == 2 ) {
+        resample_out_ptr = samplesOut2_tmp;
+    } else {
+        resample_out_ptr = samplesOut;
+    }
+
+    for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
+
+        /* Resample decoded signal to API_sampleRate */
+        ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );
+
+        /* Interleave if stereo output and stereo stream */
+        if( decControl->nChannelsAPI == 2 ) {
+            for( i = 0; i < *nSamplesOut; i++ ) {
+                samplesOut[ n + 2 * i ] = resample_out_ptr[ i ];
+            }
+        }
+    }
+
+    /* Create two channel output from mono stream */
+    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) {
+        if ( stereo_to_mono ){
+            /* Resample right channel for newly collapsed stereo just in case
+               we weren't doing collapsing when switching to mono */
+            ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec );
+
+            for( i = 0; i < *nSamplesOut; i++ ) {
+                samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
+            }
+        } else {
+            for( i = 0; i < *nSamplesOut; i++ ) {
+                samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ];
+            }
+        }
+    }
+
+    /* Export pitch lag, measured at 48 kHz sampling rate */
+    if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) {
+        int mult_tab[ 3 ] = { 6, 4, 3 };
+        decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ];
+    } else {
+        decControl->prevPitchLag = 0;
+    }
+
+    if( lostFlag == FLAG_PACKET_LOST ) {
+       /* On packet loss, remove the gain clamping to prevent having the energy "bounce back"
+          if we lose packets when the energy is going down */
+       for ( i = 0; i < psDec->nChannelsInternal; i++ )
+          psDec->channel_state[ i ].LastGainIndex = 10;
+    } else {
+       psDec->prev_decode_only_middle = decode_only_middle;
+    }
+    RESTORE_STACK;
+    return ret;
+}
+
+#if 0
+/* Getting table of contents for a packet */
+opus_int silk_get_TOC(
+    const opus_uint8                *payload,           /* I    Payload data                                */
+    const opus_int                  nBytesIn,           /* I    Number of input bytes                       */
+    const opus_int                  nFramesPerPayload,  /* I    Number of SILK frames per payload           */
+    silk_TOC_struct                 *Silk_TOC           /* O    Type of content                             */
+)
+{
+    opus_int i, flags, ret = SILK_NO_ERROR;
+
+    if( nBytesIn < 1 ) {
+        return -1;
+    }
+    if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) {
+        return -1;
+    }
+
+    silk_memset( Silk_TOC, 0, sizeof( *Silk_TOC ) );
+
+    /* For stereo, extract the flags for the mid channel */
+    flags = silk_RSHIFT( payload[ 0 ], 7 - nFramesPerPayload ) & ( silk_LSHIFT( 1, nFramesPerPayload + 1 ) - 1 );
+
+    Silk_TOC->inbandFECFlag = flags & 1;
+    for( i = nFramesPerPayload - 1; i >= 0 ; i-- ) {
+        flags = silk_RSHIFT( flags, 1 );
+        Silk_TOC->VADFlags[ i ] = flags & 1;
+        Silk_TOC->VADFlag |= flags & 1;
+    }
+
+    return ret;
+}
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_core.c b/lib/rbcodec/codecs/libopus/silk/decode_core.c
new file mode 100644
index 0000000000..f4ed7e0f6d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_core.c
@@ -0,0 +1,238 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+#include "stack_alloc.h"
+
+/**********************************************************/
+/* Core decoder. Performs inverse NSQ operation LTP + LPC */
+/**********************************************************/
+void silk_decode_core(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state                               */
+    silk_decoder_control        *psDecCtrl,                     /* I    Decoder control                             */
+    opus_int16                  xq[],                           /* O    Decoded speech                              */
+    const opus_int              pulses[ MAX_FRAME_LENGTH ]      /* I    Pulse signal                                */
+)
+{
+    opus_int   i, k, lag = 0, start_idx, sLTP_buf_idx, NLSF_interpolation_flag, signalType;
+    opus_int16 *A_Q12, *B_Q14, *pxq, A_Q12_tmp[ MAX_LPC_ORDER ];
+    VARDECL( opus_int16, sLTP );
+    VARDECL( opus_int32, sLTP_Q15 );
+    opus_int32 LTP_pred_Q13, LPC_pred_Q10, Gain_Q10, inv_gain_Q31, gain_adj_Q16, rand_seed, offset_Q10;
+    opus_int32 *pred_lag_ptr, *pexc_Q14, *pres_Q14;
+    VARDECL( opus_int32, res_Q14 );
+    VARDECL( opus_int32, sLPC_Q14 );
+    SAVE_STACK;
+
+    silk_assert( psDec->prev_gain_Q16 != 0 );
+
+    ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 );
+    ALLOC( sLTP_Q15, psDec->ltp_mem_length + psDec->frame_length, opus_int32 );
+    ALLOC( res_Q14, psDec->subfr_length, opus_int32 );
+    ALLOC( sLPC_Q14, psDec->subfr_length + MAX_LPC_ORDER, opus_int32 );
+
+    offset_Q10 = silk_Quantization_Offsets_Q10[ psDec->indices.signalType >> 1 ][ psDec->indices.quantOffsetType ];
+
+    if( psDec->indices.NLSFInterpCoef_Q2 < 1 << 2 ) {
+        NLSF_interpolation_flag = 1;
+    } else {
+        NLSF_interpolation_flag = 0;
+    }
+
+    /* Decode excitation */
+    rand_seed = psDec->indices.Seed;
+    for( i = 0; i < psDec->frame_length; i++ ) {
+        rand_seed = silk_RAND( rand_seed );
+        psDec->exc_Q14[ i ] = silk_LSHIFT( (opus_int32)pulses[ i ], 14 );
+        if( psDec->exc_Q14[ i ] > 0 ) {
+            psDec->exc_Q14[ i ] -= QUANT_LEVEL_ADJUST_Q10 << 4;
+        } else
+        if( psDec->exc_Q14[ i ] < 0 ) {
+            psDec->exc_Q14[ i ] += QUANT_LEVEL_ADJUST_Q10 << 4;
+        }
+        psDec->exc_Q14[ i ] += offset_Q10 << 4;
+        if( rand_seed < 0 ) {
+           psDec->exc_Q14[ i ] = -psDec->exc_Q14[ i ];
+        }
+
+        rand_seed = silk_ADD32_ovflw( rand_seed, pulses[ i ] );
+    }
+
+    /* Copy LPC state */
+    silk_memcpy( sLPC_Q14, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) );
+
+    pexc_Q14 = psDec->exc_Q14;
+    pxq      = xq;
+    sLTP_buf_idx = psDec->ltp_mem_length;
+    /* Loop over subframes */
+    for( k = 0; k < psDec->nb_subfr; k++ ) {
+        pres_Q14 = res_Q14;
+        A_Q12 = psDecCtrl->PredCoef_Q12[ k >> 1 ];
+
+        /* Preload LPC coeficients to array on stack. Gives small performance gain */
+        silk_memcpy( A_Q12_tmp, A_Q12, psDec->LPC_order * sizeof( opus_int16 ) );
+        B_Q14        = &psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER ];
+        signalType   = psDec->indices.signalType;
+
+        Gain_Q10     = silk_RSHIFT( psDecCtrl->Gains_Q16[ k ], 6 );
+        inv_gain_Q31 = silk_INVERSE32_varQ( psDecCtrl->Gains_Q16[ k ], 47 );
+
+        /* Calculate gain adjustment factor */
+        if( psDecCtrl->Gains_Q16[ k ] != psDec->prev_gain_Q16 ) {
+            gain_adj_Q16 =  silk_DIV32_varQ( psDec->prev_gain_Q16, psDecCtrl->Gains_Q16[ k ], 16 );
+
+            /* Scale short term state */
+            for( i = 0; i < MAX_LPC_ORDER; i++ ) {
+                sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, sLPC_Q14[ i ] );
+            }
+        } else {
+            gain_adj_Q16 = (opus_int32)1 << 16;
+        }
+
+        /* Save inv_gain */
+        silk_assert( inv_gain_Q31 != 0 );
+        psDec->prev_gain_Q16 = psDecCtrl->Gains_Q16[ k ];
+
+        /* Avoid abrupt transition from voiced PLC to unvoiced normal decoding */
+        if( psDec->lossCnt && psDec->prevSignalType == TYPE_VOICED &&
+            psDec->indices.signalType != TYPE_VOICED && k < MAX_NB_SUBFR/2 ) {
+
+            silk_memset( B_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) );
+            B_Q14[ LTP_ORDER/2 ] = SILK_FIX_CONST( 0.25, 14 );
+
+            signalType = TYPE_VOICED;
+            psDecCtrl->pitchL[ k ] = psDec->lagPrev;
+        }
+
+        if( signalType == TYPE_VOICED ) {
+            /* Voiced */
+            lag = psDecCtrl->pitchL[ k ];
+
+            /* Re-whitening */
+            if( k == 0 || ( k == 2 && NLSF_interpolation_flag ) ) {
+                /* Rewhiten with new A coefs */
+                start_idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2;
+                silk_assert( start_idx > 0 );
+
+                if( k == 2 ) {
+                    silk_memcpy( &psDec->outBuf[ psDec->ltp_mem_length ], xq, 2 * psDec->subfr_length * sizeof( opus_int16 ) );
+                }
+
+                silk_LPC_analysis_filter( &sLTP[ start_idx ], &psDec->outBuf[ start_idx + k * psDec->subfr_length ],
+                    A_Q12, psDec->ltp_mem_length - start_idx, psDec->LPC_order );
+
+                /* After rewhitening the LTP state is unscaled */
+                if( k == 0 ) {
+                    /* Do LTP downscaling to reduce inter-packet dependency */
+                    inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, psDecCtrl->LTP_scale_Q14 ), 2 );
+                }
+                for( i = 0; i < lag + LTP_ORDER/2; i++ ) {
+                    sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWB( inv_gain_Q31, sLTP[ psDec->ltp_mem_length - i - 1 ] );
+                }
+            } else {
+                /* Update LTP state when Gain changes */
+                if( gain_adj_Q16 != (opus_int32)1 << 16 ) {
+                    for( i = 0; i < lag + LTP_ORDER/2; i++ ) {
+                        sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ sLTP_buf_idx - i - 1 ] );
+                    }
+                }
+            }
+        }
+
+        /* Long-term prediction */
+        if( signalType == TYPE_VOICED ) {
+            /* Set up pointer */
+            pred_lag_ptr = &sLTP_Q15[ sLTP_buf_idx - lag + LTP_ORDER / 2 ];
+            for( i = 0; i < psDec->subfr_length; i++ ) {
+                /* Unrolled loop */
+                /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */
+                LTP_pred_Q13 = 2;
+                LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[  0 ], B_Q14[ 0 ] );
+                LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -1 ], B_Q14[ 1 ] );
+                LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -2 ], B_Q14[ 2 ] );
+                LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -3 ], B_Q14[ 3 ] );
+                LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], B_Q14[ 4 ] );
+                pred_lag_ptr++;
+
+                /* Generate LPC excitation */
+                pres_Q14[ i ] = silk_ADD_LSHIFT32( pexc_Q14[ i ], LTP_pred_Q13, 1 );
+
+                /* Update states */
+                sLTP_Q15[ sLTP_buf_idx ] = silk_LSHIFT( pres_Q14[ i ], 1 );
+                sLTP_buf_idx++;
+            }
+        } else {
+            pres_Q14 = pexc_Q14;
+        }
+
+        for( i = 0; i < psDec->subfr_length; i++ ) {
+            /* Short-term prediction */
+            silk_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 );
+            /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */
+            LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  1 ], A_Q12_tmp[ 0 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  2 ], A_Q12_tmp[ 1 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  3 ], A_Q12_tmp[ 2 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  4 ], A_Q12_tmp[ 3 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  5 ], A_Q12_tmp[ 4 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  6 ], A_Q12_tmp[ 5 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  7 ], A_Q12_tmp[ 6 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  8 ], A_Q12_tmp[ 7 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i -  9 ], A_Q12_tmp[ 8 ] );
+            LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 10 ], A_Q12_tmp[ 9 ] );
+            if( psDec->LPC_order == 16 ) {
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 11 ], A_Q12_tmp[ 10 ] );
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 12 ], A_Q12_tmp[ 11 ] );
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 13 ], A_Q12_tmp[ 12 ] );
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 14 ], A_Q12_tmp[ 13 ] );
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 15 ], A_Q12_tmp[ 14 ] );
+                LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 16 ], A_Q12_tmp[ 15 ] );
+            }
+
+            /* Add prediction to LPC excitation */
+            sLPC_Q14[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT32( pres_Q14[ i ], LPC_pred_Q10, 4 );
+
+            /* Scale with gain */
+            pxq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14[ MAX_LPC_ORDER + i ], Gain_Q10 ), 8 ) );
+        }
+
+        /* DEBUG_STORE_DATA( dec.pcm, pxq, psDec->subfr_length * sizeof( opus_int16 ) ) */
+
+        /* Update LPC filter state */
+        silk_memcpy( sLPC_Q14, &sLPC_Q14[ psDec->subfr_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) );
+        pexc_Q14 += psDec->subfr_length;
+        pxq      += psDec->subfr_length;
+    }
+
+    /* Save LPC state */
+    silk_memcpy( psDec->sLPC_Q14_buf, sLPC_Q14, MAX_LPC_ORDER * sizeof( opus_int32 ) );
+    RESTORE_STACK;
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_frame.c b/lib/rbcodec/codecs/libopus/silk/decode_frame.c
new file mode 100644
index 0000000000..349df506e5
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_frame.c
@@ -0,0 +1,128 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+#include "stack_alloc.h"
+#include "PLC.h"
+
+/****************/
+/* Decode frame */
+/****************/
+opus_int silk_decode_frame(
+    silk_decoder_state          *psDec,                         /* I/O  Pointer to Silk decoder state               */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int16                  pOut[],                         /* O    Pointer to output speech frame              */
+    opus_int32                  *pN,                            /* O    Pointer to size of output frame             */
+    opus_int                    lostFlag,                       /* I    0: no loss, 1 loss, 2 decode fec            */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+)
+{
+    VARDECL( silk_decoder_control, psDecCtrl );
+    opus_int         L, mv_len, ret = 0;
+    VARDECL( opus_int, pulses );
+    SAVE_STACK;
+
+    L = psDec->frame_length;
+    ALLOC( psDecCtrl, 1, silk_decoder_control );
+    ALLOC( pulses, (L + SHELL_CODEC_FRAME_LENGTH - 1) &
+                   ~(SHELL_CODEC_FRAME_LENGTH - 1), opus_int );
+    psDecCtrl->LTP_scale_Q14 = 0;
+
+    /* Safety checks */
+    silk_assert( L > 0 && L <= MAX_FRAME_LENGTH );
+
+    if(   lostFlag == FLAG_DECODE_NORMAL ||
+        ( lostFlag == FLAG_DECODE_LBRR && psDec->LBRR_flags[ psDec->nFramesDecoded ] == 1 ) )
+    {
+        /*********************************************/
+        /* Decode quantization indices of side info  */
+        /*********************************************/
+        silk_decode_indices( psDec, psRangeDec, psDec->nFramesDecoded, lostFlag, condCoding );
+
+        /*********************************************/
+        /* Decode quantization indices of excitation */
+        /*********************************************/
+        silk_decode_pulses( psRangeDec, pulses, psDec->indices.signalType,
+                psDec->indices.quantOffsetType, psDec->frame_length );
+
+        /********************************************/
+        /* Decode parameters and pulse signal       */
+        /********************************************/
+        silk_decode_parameters( psDec, psDecCtrl, condCoding );
+
+        /********************************************************/
+        /* Run inverse NSQ                                      */
+        /********************************************************/
+        silk_decode_core( psDec, psDecCtrl, pOut, pulses );
+
+        /********************************************************/
+        /* Update PLC state                                     */
+        /********************************************************/
+        silk_PLC( psDec, psDecCtrl, pOut, 0 );
+
+        psDec->lossCnt = 0;
+        psDec->prevSignalType = psDec->indices.signalType;
+        silk_assert( psDec->prevSignalType >= 0 && psDec->prevSignalType <= 2 );
+
+        /* A frame has been decoded without errors */
+        psDec->first_frame_after_reset = 0;
+    } else {
+        /* Handle packet loss by extrapolation */
+        silk_PLC( psDec, psDecCtrl, pOut, 1 );
+    }
+
+    /*************************/
+    /* Update output buffer. */
+    /*************************/
+    silk_assert( psDec->ltp_mem_length >= psDec->frame_length );
+    mv_len = psDec->ltp_mem_length - psDec->frame_length;
+    silk_memmove( psDec->outBuf, &psDec->outBuf[ psDec->frame_length ], mv_len * sizeof(opus_int16) );
+    silk_memcpy( &psDec->outBuf[ mv_len ], pOut, psDec->frame_length * sizeof( opus_int16 ) );
+
+    /****************************************************************/
+    /* Ensure smooth connection of extrapolated and good frames     */
+    /****************************************************************/
+    silk_PLC_glue_frames( psDec, pOut, L );
+
+    /************************************************/
+    /* Comfort noise generation / estimation        */
+    /************************************************/
+    silk_CNG( psDec, psDecCtrl, pOut, L );
+
+    /* Update some decoder state variables */
+    psDec->lagPrev = psDecCtrl->pitchL[ psDec->nb_subfr - 1 ];
+
+    /* Set output frame length */
+    *pN = L;
+
+    RESTORE_STACK;
+    return ret;
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_indices.c b/lib/rbcodec/codecs/libopus/silk/decode_indices.c
new file mode 100644
index 0000000000..00eef1de52
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_indices.c
@@ -0,0 +1,151 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Decode side-information parameters from payload */
+void silk_decode_indices(
+    silk_decoder_state          *psDec,                         /* I/O  State                                       */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    FrameIndex,                     /* I    Frame number                                */
+    opus_int                    decode_LBRR,                    /* I    Flag indicating LBRR data is being decoded  */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+)
+{
+    opus_int   i, k, Ix;
+    opus_int   decode_absolute_lagIndex, delta_lagIndex;
+    opus_int16 ec_ix[ MAX_LPC_ORDER ];
+    opus_uint8 pred_Q8[ MAX_LPC_ORDER ];
+
+    /*******************************************/
+    /* Decode signal type and quantizer offset */
+    /*******************************************/
+    if( decode_LBRR || psDec->VAD_flags[ FrameIndex ] ) {
+        Ix = ec_dec_icdf( psRangeDec, silk_type_offset_VAD_iCDF, 8 ) + 2;
+    } else {
+        Ix = ec_dec_icdf( psRangeDec, silk_type_offset_no_VAD_iCDF, 8 );
+    }
+    psDec->indices.signalType      = (opus_int8)silk_RSHIFT( Ix, 1 );
+    psDec->indices.quantOffsetType = (opus_int8)( Ix & 1 );
+
+    /****************/
+    /* Decode gains */
+    /****************/
+    /* First subframe */
+    if( condCoding == CODE_CONDITIONALLY ) {
+        /* Conditional coding */
+        psDec->indices.GainsIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 );
+    } else {
+        /* Independent coding, in two stages: MSB bits followed by 3 LSBs */
+        psDec->indices.GainsIndices[ 0 ]  = (opus_int8)silk_LSHIFT( ec_dec_icdf( psRangeDec, silk_gain_iCDF[ psDec->indices.signalType ], 8 ), 3 );
+        psDec->indices.GainsIndices[ 0 ] += (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform8_iCDF, 8 );
+    }
+
+    /* Remaining subframes */
+    for( i = 1; i < psDec->nb_subfr; i++ ) {
+        psDec->indices.GainsIndices[ i ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 );
+    }
+
+    /**********************/
+    /* Decode LSF Indices */
+    /**********************/
+    psDec->indices.NLSFIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->CB1_iCDF[ ( psDec->indices.signalType >> 1 ) * psDec->psNLSF_CB->nVectors ], 8 );
+    silk_NLSF_unpack( ec_ix, pred_Q8, psDec->psNLSF_CB, psDec->indices.NLSFIndices[ 0 ] );
+    silk_assert( psDec->psNLSF_CB->order == psDec->LPC_order );
+    for( i = 0; i < psDec->psNLSF_CB->order; i++ ) {
+        Ix = ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 );
+        if( Ix == 0 ) {
+            Ix -= ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 );
+        } else if( Ix == 2 * NLSF_QUANT_MAX_AMPLITUDE ) {
+            Ix += ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 );
+        }
+        psDec->indices.NLSFIndices[ i+1 ] = (opus_int8)( Ix - NLSF_QUANT_MAX_AMPLITUDE );
+    }
+
+    /* Decode LSF interpolation factor */
+    if( psDec->nb_subfr == MAX_NB_SUBFR ) {
+        psDec->indices.NLSFInterpCoef_Q2 = (opus_int8)ec_dec_icdf( psRangeDec, silk_NLSF_interpolation_factor_iCDF, 8 );
+    } else {
+        psDec->indices.NLSFInterpCoef_Q2 = 4;
+    }
+
+    if( psDec->indices.signalType == TYPE_VOICED )
+    {
+        /*********************/
+        /* Decode pitch lags */
+        /*********************/
+        /* Get lag index */
+        decode_absolute_lagIndex = 1;
+        if( condCoding == CODE_CONDITIONALLY && psDec->ec_prevSignalType == TYPE_VOICED ) {
+            /* Decode Delta index */
+            delta_lagIndex = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_delta_iCDF, 8 );
+            if( delta_lagIndex > 0 ) {
+                delta_lagIndex = delta_lagIndex - 9;
+                psDec->indices.lagIndex = (opus_int16)( psDec->ec_prevLagIndex + delta_lagIndex );
+                decode_absolute_lagIndex = 0;
+            }
+        }
+        if( decode_absolute_lagIndex ) {
+            /* Absolute decoding */
+            psDec->indices.lagIndex  = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_lag_iCDF, 8 ) * silk_RSHIFT( psDec->fs_kHz, 1 );
+            psDec->indices.lagIndex += (opus_int16)ec_dec_icdf( psRangeDec, psDec->pitch_lag_low_bits_iCDF, 8 );
+        }
+        psDec->ec_prevLagIndex = psDec->indices.lagIndex;
+
+        /* Get countour index */
+        psDec->indices.contourIndex = (opus_int8)ec_dec_icdf( psRangeDec, psDec->pitch_contour_iCDF, 8 );
+
+        /********************/
+        /* Decode LTP gains */
+        /********************/
+        /* Decode PERIndex value */
+        psDec->indices.PERIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_per_index_iCDF, 8 );
+
+        for( k = 0; k < psDec->nb_subfr; k++ ) {
+            psDec->indices.LTPIndex[ k ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_gain_iCDF_ptrs[ psDec->indices.PERIndex ], 8 );
+        }
+
+        /**********************/
+        /* Decode LTP scaling */
+        /**********************/
+        if( condCoding == CODE_INDEPENDENTLY ) {
+            psDec->indices.LTP_scaleIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTPscale_iCDF, 8 );
+        } else {
+            psDec->indices.LTP_scaleIndex = 0;
+        }
+    }
+    psDec->ec_prevSignalType = psDec->indices.signalType;
+
+    /***************/
+    /* Decode seed */
+    /***************/
+    psDec->indices.Seed = (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform4_iCDF, 8 );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_parameters.c b/lib/rbcodec/codecs/libopus/silk/decode_parameters.c
new file mode 100644
index 0000000000..f7c6e2f112
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_parameters.c
@@ -0,0 +1,115 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Decode parameters from payload */
+void silk_decode_parameters(
+    silk_decoder_state          *psDec,                         /* I/O  State                                       */
+    silk_decoder_control        *psDecCtrl,                     /* I/O  Decoder control                             */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+)
+{
+    opus_int   i, k, Ix;
+    opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], pNLSF0_Q15[ MAX_LPC_ORDER ];
+    const opus_int8 *cbk_ptr_Q7;
+
+    /* Dequant Gains */
+    silk_gains_dequant( psDecCtrl->Gains_Q16, psDec->indices.GainsIndices,
+        &psDec->LastGainIndex, condCoding == CODE_CONDITIONALLY, psDec->nb_subfr );
+
+    /****************/
+    /* Decode NLSFs */
+    /****************/
+    silk_NLSF_decode( pNLSF_Q15, psDec->indices.NLSFIndices, psDec->psNLSF_CB );
+
+    /* Convert NLSF parameters to AR prediction filter coefficients */
+    silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 1 ], pNLSF_Q15, psDec->LPC_order );
+
+    /* If just reset, e.g., because internal Fs changed, do not allow interpolation */
+    /* improves the case of packet loss in the first frame after a switch           */
+    if( psDec->first_frame_after_reset == 1 ) {
+        psDec->indices.NLSFInterpCoef_Q2 = 4;
+    }
+
+    if( psDec->indices.NLSFInterpCoef_Q2 < 4 ) {
+        /* Calculation of the interpolated NLSF0 vector from the interpolation factor, */
+        /* the previous NLSF1, and the current NLSF1                                   */
+        for( i = 0; i < psDec->LPC_order; i++ ) {
+            pNLSF0_Q15[ i ] = psDec->prevNLSF_Q15[ i ] + silk_RSHIFT( silk_MUL( psDec->indices.NLSFInterpCoef_Q2,
+                pNLSF_Q15[ i ] - psDec->prevNLSF_Q15[ i ] ), 2 );
+        }
+
+        /* Convert NLSF parameters to AR prediction filter coefficients */
+        silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 0 ], pNLSF0_Q15, psDec->LPC_order );
+    } else {
+        /* Copy LPC coefficients for first half from second half */
+        silk_memcpy( psDecCtrl->PredCoef_Q12[ 0 ], psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) );
+    }
+
+    silk_memcpy( psDec->prevNLSF_Q15, pNLSF_Q15, psDec->LPC_order * sizeof( opus_int16 ) );
+
+    /* After a packet loss do BWE of LPC coefs */
+    if( psDec->lossCnt ) {
+        silk_bwexpander( psDecCtrl->PredCoef_Q12[ 0 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 );
+        silk_bwexpander( psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 );
+    }
+
+    if( psDec->indices.signalType == TYPE_VOICED ) {
+        /*********************/
+        /* Decode pitch lags */
+        /*********************/
+
+        /* Decode pitch values */
+        silk_decode_pitch( psDec->indices.lagIndex, psDec->indices.contourIndex, psDecCtrl->pitchL, psDec->fs_kHz, psDec->nb_subfr );
+
+        /* Decode Codebook Index */
+        cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ psDec->indices.PERIndex ]; /* set pointer to start of codebook */
+
+        for( k = 0; k < psDec->nb_subfr; k++ ) {
+            Ix = psDec->indices.LTPIndex[ k ];
+            for( i = 0; i < LTP_ORDER; i++ ) {
+                psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER + i ] = silk_LSHIFT( cbk_ptr_Q7[ Ix * LTP_ORDER + i ], 7 );
+            }
+        }
+
+        /**********************/
+        /* Decode LTP scaling */
+        /**********************/
+        Ix = psDec->indices.LTP_scaleIndex;
+        psDecCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ Ix ];
+    } else {
+        silk_memset( psDecCtrl->pitchL,      0,             psDec->nb_subfr * sizeof( opus_int   ) );
+        silk_memset( psDecCtrl->LTPCoef_Q14, 0, LTP_ORDER * psDec->nb_subfr * sizeof( opus_int16 ) );
+        psDec->indices.PERIndex  = 0;
+        psDecCtrl->LTP_scale_Q14 = 0;
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_pitch.c b/lib/rbcodec/codecs/libopus/silk/decode_pitch.c
new file mode 100644
index 0000000000..8190a19e51
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_pitch.c
@@ -0,0 +1,77 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/***********************************************************
+* Pitch analyser function
+********************************************************** */
+#include "SigProc_FIX.h"
+#include "pitch_est_defines.h"
+
+void silk_decode_pitch(
+    opus_int16                  lagIndex,           /* I                                                                */
+    opus_int8                   contourIndex,       /* O                                                                */
+    opus_int                    pitch_lags[],       /* O    4 pitch values                                              */
+    const opus_int              Fs_kHz,             /* I    sampling frequency (kHz)                                    */
+    const opus_int              nb_subfr            /* I    number of sub frames                                        */
+)
+{
+    opus_int   lag, k, min_lag, max_lag, cbk_size;
+    const opus_int8 *Lag_CB_ptr;
+
+    if( Fs_kHz == 8 ) {
+        if( nb_subfr == PE_MAX_NB_SUBFR ) {
+            Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ];
+            cbk_size   = PE_NB_CBKS_STAGE2_EXT;
+        } else {
+            silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 );
+            Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ];
+            cbk_size   = PE_NB_CBKS_STAGE2_10MS;
+        }
+    } else {
+        if( nb_subfr == PE_MAX_NB_SUBFR ) {
+            Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ];
+            cbk_size   = PE_NB_CBKS_STAGE3_MAX;
+        } else {
+            silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 );
+            Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ];
+            cbk_size   = PE_NB_CBKS_STAGE3_10MS;
+        }
+    }
+
+    min_lag = silk_SMULBB( PE_MIN_LAG_MS, Fs_kHz );
+    max_lag = silk_SMULBB( PE_MAX_LAG_MS, Fs_kHz );
+    lag = min_lag + lagIndex;
+
+    for( k = 0; k < nb_subfr; k++ ) {
+        pitch_lags[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, contourIndex, cbk_size );
+        pitch_lags[ k ] = silk_LIMIT( pitch_lags[ k ], min_lag, max_lag );
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decode_pulses.c b/lib/rbcodec/codecs/libopus/silk/decode_pulses.c
new file mode 100644
index 0000000000..78fc2032e8
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decode_pulses.c
@@ -0,0 +1,115 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/*********************************************/
+/* Decode quantization indices of excitation */
+/*********************************************/
+void silk_decode_pulses(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    pulses[],                       /* O    Excitation signal                           */
+    const opus_int              signalType,                     /* I    Sigtype                                     */
+    const opus_int              quantOffsetType,                /* I    quantOffsetType                             */
+    const opus_int              frame_length                    /* I    Frame length                                */
+)
+{
+    opus_int   i, j, k, iter, abs_q, nLS, RateLevelIndex;
+    opus_int   sum_pulses[ MAX_NB_SHELL_BLOCKS ], nLshifts[ MAX_NB_SHELL_BLOCKS ];
+    opus_int   *pulses_ptr;
+    const opus_uint8 *cdf_ptr;
+
+    /*********************/
+    /* Decode rate level */
+    /*********************/
+    RateLevelIndex = ec_dec_icdf( psRangeDec, silk_rate_levels_iCDF[ signalType >> 1 ], 8 );
+
+    /* Calculate number of shell blocks */
+    silk_assert( 1 << LOG2_SHELL_CODEC_FRAME_LENGTH == SHELL_CODEC_FRAME_LENGTH );
+    iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH );
+    if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) {
+        silk_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */
+        iter++;
+    }
+
+    /***************************************************/
+    /* Sum-Weighted-Pulses Decoding                    */
+    /***************************************************/
+    cdf_ptr = silk_pulses_per_block_iCDF[ RateLevelIndex ];
+    for( i = 0; i < iter; i++ ) {
+        nLshifts[ i ] = 0;
+        sum_pulses[ i ] = ec_dec_icdf( psRangeDec, cdf_ptr, 8 );
+
+        /* LSB indication */
+        while( sum_pulses[ i ] == MAX_PULSES + 1 ) {
+            nLshifts[ i ]++;
+            /* When we've already got 10 LSBs, we shift the table to not allow (MAX_PULSES + 1) */
+            sum_pulses[ i ] = ec_dec_icdf( psRangeDec,
+                    silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1] + ( nLshifts[ i ] == 10 ), 8 );
+        }
+    }
+
+    /***************************************************/
+    /* Shell decoding                                  */
+    /***************************************************/
+    for( i = 0; i < iter; i++ ) {
+        if( sum_pulses[ i ] > 0 ) {
+            silk_shell_decoder( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], psRangeDec, sum_pulses[ i ] );
+        } else {
+            silk_memset( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof( opus_int ) );
+        }
+    }
+
+    /***************************************************/
+    /* LSB Decoding                                    */
+    /***************************************************/
+    for( i = 0; i < iter; i++ ) {
+        if( nLshifts[ i ] > 0 ) {
+            nLS = nLshifts[ i ];
+            pulses_ptr = &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ];
+            for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) {
+                abs_q = pulses_ptr[ k ];
+                for( j = 0; j < nLS; j++ ) {
+                    abs_q = silk_LSHIFT( abs_q, 1 );
+                    abs_q += ec_dec_icdf( psRangeDec, silk_lsb_iCDF, 8 );
+                }
+                pulses_ptr[ k ] = abs_q;
+            }
+            /* Mark the number of pulses non-zero for sign decoding. */
+            sum_pulses[ i ] |= nLS << 5;
+        }
+    }
+
+    /****************************************/
+    /* Decode and add signs to pulse signal */
+    /****************************************/
+    silk_decode_signs( psRangeDec, pulses, frame_length, signalType, quantOffsetType, sum_pulses );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/decoder_set_fs.c b/lib/rbcodec/codecs/libopus/silk/decoder_set_fs.c
new file mode 100644
index 0000000000..fcc26d7a0e
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/decoder_set_fs.c
@@ -0,0 +1,108 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Set decoder sampling rate */
+opus_int silk_decoder_set_fs(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state pointer                       */
+    opus_int                    fs_kHz,                         /* I    Sampling frequency (kHz)                    */
+    opus_int32                  fs_API_Hz                       /* I    API Sampling frequency (Hz)                 */
+)
+{
+    opus_int frame_length, ret = 0;
+
+    silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
+    silk_assert( psDec->nb_subfr == MAX_NB_SUBFR || psDec->nb_subfr == MAX_NB_SUBFR/2 );
+
+    /* New (sub)frame length */
+    psDec->subfr_length = silk_SMULBB( SUB_FRAME_LENGTH_MS, fs_kHz );
+    frame_length = silk_SMULBB( psDec->nb_subfr, psDec->subfr_length );
+
+    /* Initialize resampler when switching internal or external sampling frequency */
+    if( psDec->fs_kHz != fs_kHz || psDec->fs_API_hz != fs_API_Hz ) {
+        /* Initialize the resampler for dec_API.c preparing resampling from fs_kHz to API_fs_Hz */
+        ret += silk_resampler_init( &psDec->resampler_state, silk_SMULBB( fs_kHz, 1000 ), fs_API_Hz, 0 );
+
+        psDec->fs_API_hz = fs_API_Hz;
+    }
+
+    if( psDec->fs_kHz != fs_kHz || frame_length != psDec->frame_length ) {
+        if( fs_kHz == 8 ) {
+            if( psDec->nb_subfr == MAX_NB_SUBFR ) {
+                psDec->pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
+            } else {
+                psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
+            }
+        } else {
+            if( psDec->nb_subfr == MAX_NB_SUBFR ) {
+                psDec->pitch_contour_iCDF = silk_pitch_contour_iCDF;
+            } else {
+                psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
+            }
+        }
+        if( psDec->fs_kHz != fs_kHz ) {
+            psDec->ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
+            if( fs_kHz == 8 || fs_kHz == 12 ) {
+                psDec->LPC_order = MIN_LPC_ORDER;
+                psDec->psNLSF_CB = &silk_NLSF_CB_NB_MB;
+            } else {
+                psDec->LPC_order = MAX_LPC_ORDER;
+                psDec->psNLSF_CB = &silk_NLSF_CB_WB;
+            }
+            if( fs_kHz == 16 ) {
+                psDec->pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
+            } else if( fs_kHz == 12 ) {
+                psDec->pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
+            } else if( fs_kHz == 8 ) {
+                psDec->pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
+            } else {
+                /* unsupported sampling rate */
+                silk_assert( 0 );
+            }
+            psDec->first_frame_after_reset = 1;
+            psDec->lagPrev                 = 100;
+            psDec->LastGainIndex           = 10;
+            psDec->prevSignalType          = TYPE_NO_VOICE_ACTIVITY;
+            silk_memset( psDec->outBuf, 0, sizeof(psDec->outBuf));
+            silk_memset( psDec->sLPC_Q14_buf, 0, sizeof(psDec->sLPC_Q14_buf) );
+        }
+
+        psDec->fs_kHz       = fs_kHz;
+        psDec->frame_length = frame_length;
+    }
+
+    /* Check that settings are valid */
+    silk_assert( psDec->frame_length > 0 && psDec->frame_length <= MAX_FRAME_LENGTH );
+
+    return ret;
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/define.h b/lib/rbcodec/codecs/libopus/silk/define.h
new file mode 100644
index 0000000000..f74f4869c3
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/define.h
@@ -0,0 +1,235 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_DEFINE_H
+#define SILK_DEFINE_H
+
+#include "errors.h"
+#include "typedef.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/* Max number of encoder channels (1/2) */
+#define ENCODER_NUM_CHANNELS                    2
+/* Number of decoder channels (1/2) */
+#define DECODER_NUM_CHANNELS                    2
+
+#define MAX_FRAMES_PER_PACKET                   3
+
+/* Limits on bitrate */
+#define MIN_TARGET_RATE_BPS                     5000
+#define MAX_TARGET_RATE_BPS                     80000
+#define TARGET_RATE_TAB_SZ                      8
+
+/* LBRR thresholds */
+#define LBRR_NB_MIN_RATE_BPS                    12000
+#define LBRR_MB_MIN_RATE_BPS                    14000
+#define LBRR_WB_MIN_RATE_BPS                    16000
+
+/* DTX settings */
+#define NB_SPEECH_FRAMES_BEFORE_DTX             10      /* eq 200 ms */
+#define MAX_CONSECUTIVE_DTX                     20      /* eq 400 ms */
+
+/* Maximum sampling frequency */
+#define MAX_FS_KHZ                              16
+#define MAX_API_FS_KHZ                          48
+
+/* Signal types */
+#define TYPE_NO_VOICE_ACTIVITY                  0
+#define TYPE_UNVOICED                           1
+#define TYPE_VOICED                             2
+
+/* Conditional coding types */
+#define CODE_INDEPENDENTLY                      0
+#define CODE_INDEPENDENTLY_NO_LTP_SCALING       1
+#define CODE_CONDITIONALLY                      2
+
+/* Settings for stereo processing */
+#define STEREO_QUANT_TAB_SIZE                   16
+#define STEREO_QUANT_SUB_STEPS                  5
+#define STEREO_INTERP_LEN_MS                    8       /* must be even */
+#define STEREO_RATIO_SMOOTH_COEF                0.01    /* smoothing coef for signal norms and stereo width */
+
+/* Range of pitch lag estimates */
+#define PITCH_EST_MIN_LAG_MS                    2       /* 2 ms -> 500 Hz */
+#define PITCH_EST_MAX_LAG_MS                    18      /* 18 ms -> 56 Hz */
+
+/* Maximum number of subframes */
+#define MAX_NB_SUBFR                            4
+
+/* Number of samples per frame */
+#define LTP_MEM_LENGTH_MS                       20
+#define SUB_FRAME_LENGTH_MS                     5
+#define MAX_SUB_FRAME_LENGTH                    ( SUB_FRAME_LENGTH_MS * MAX_FS_KHZ )
+#define MAX_FRAME_LENGTH_MS                     ( SUB_FRAME_LENGTH_MS * MAX_NB_SUBFR )
+#define MAX_FRAME_LENGTH                        ( MAX_FRAME_LENGTH_MS * MAX_FS_KHZ )
+
+/* Milliseconds of lookahead for pitch analysis */
+#define LA_PITCH_MS                             2
+#define LA_PITCH_MAX                            ( LA_PITCH_MS * MAX_FS_KHZ )
+
+/* Order of LPC used in find pitch */
+#define MAX_FIND_PITCH_LPC_ORDER                16
+
+/* Length of LPC window used in find pitch */
+#define FIND_PITCH_LPC_WIN_MS                   ( 20 + (LA_PITCH_MS << 1) )
+#define FIND_PITCH_LPC_WIN_MS_2_SF              ( 10 + (LA_PITCH_MS << 1) )
+#define FIND_PITCH_LPC_WIN_MAX                  ( FIND_PITCH_LPC_WIN_MS * MAX_FS_KHZ )
+
+/* Milliseconds of lookahead for noise shape analysis */
+#define LA_SHAPE_MS                             5
+#define LA_SHAPE_MAX                            ( LA_SHAPE_MS * MAX_FS_KHZ )
+
+/* Maximum length of LPC window used in noise shape analysis */
+#define SHAPE_LPC_WIN_MAX                       ( 15 * MAX_FS_KHZ )
+
+/* dB level of lowest gain quantization level */
+#define MIN_QGAIN_DB                            2
+/* dB level of highest gain quantization level */
+#define MAX_QGAIN_DB                            88
+/* Number of gain quantization levels */
+#define N_LEVELS_QGAIN                          64
+/* Max increase in gain quantization index */
+#define MAX_DELTA_GAIN_QUANT                    36
+/* Max decrease in gain quantization index */
+#define MIN_DELTA_GAIN_QUANT                    -4
+
+/* Quantization offsets (multiples of 4) */
+#define OFFSET_VL_Q10                           32
+#define OFFSET_VH_Q10                           100
+#define OFFSET_UVL_Q10                          100
+#define OFFSET_UVH_Q10                          240
+
+#define QUANT_LEVEL_ADJUST_Q10                  80
+
+/* Maximum numbers of iterations used to stabilize an LPC vector */
+#define MAX_LPC_STABILIZE_ITERATIONS            16
+#define MAX_PREDICTION_POWER_GAIN               1e4f
+#define MAX_PREDICTION_POWER_GAIN_AFTER_RESET   1e2f
+
+#define MAX_LPC_ORDER                           16
+#define MIN_LPC_ORDER                           10
+
+/* Find Pred Coef defines */
+#define LTP_ORDER                               5
+
+/* LTP quantization settings */
+#define NB_LTP_CBKS                             3
+
+/* Flag to use harmonic noise shaping */
+#define USE_HARM_SHAPING                        1
+
+/* Max LPC order of noise shaping filters */
+#define MAX_SHAPE_LPC_ORDER                     16
+
+#define HARM_SHAPE_FIR_TAPS                     3
+
+/* Maximum number of delayed decision states */
+#define MAX_DEL_DEC_STATES                      4
+
+#define LTP_BUF_LENGTH                          512
+#define LTP_MASK                                ( LTP_BUF_LENGTH - 1 )
+
+#define DECISION_DELAY                          32
+#define DECISION_DELAY_MASK                     ( DECISION_DELAY - 1 )
+
+/* Number of subframes for excitation entropy coding */
+#define SHELL_CODEC_FRAME_LENGTH                16
+#define LOG2_SHELL_CODEC_FRAME_LENGTH           4
+#define MAX_NB_SHELL_BLOCKS                     ( MAX_FRAME_LENGTH / SHELL_CODEC_FRAME_LENGTH )
+
+/* Number of rate levels, for entropy coding of excitation */
+#define N_RATE_LEVELS                           10
+
+/* Maximum sum of pulses per shell coding frame */
+#define MAX_PULSES                              16
+
+#define MAX_MATRIX_SIZE                         MAX_LPC_ORDER /* Max of LPC Order and LTP order */
+
+#if( MAX_LPC_ORDER > DECISION_DELAY )
+# define NSQ_LPC_BUF_LENGTH                     MAX_LPC_ORDER
+#else
+# define NSQ_LPC_BUF_LENGTH                     DECISION_DELAY
+#endif
+
+/***************************/
+/* Voice activity detector */
+/***************************/
+#define VAD_N_BANDS                             4
+
+#define VAD_INTERNAL_SUBFRAMES_LOG2             2
+#define VAD_INTERNAL_SUBFRAMES                  ( 1 << VAD_INTERNAL_SUBFRAMES_LOG2 )
+
+#define VAD_NOISE_LEVEL_SMOOTH_COEF_Q16         1024    /* Must be <  4096 */
+#define VAD_NOISE_LEVELS_BIAS                   50
+
+/* Sigmoid settings */
+#define VAD_NEGATIVE_OFFSET_Q5                  128     /* sigmoid is 0 at -128 */
+#define VAD_SNR_FACTOR_Q16                      45000
+
+/* smoothing for SNR measurement */
+#define VAD_SNR_SMOOTH_COEF_Q18                 4096
+
+/* Size of the piecewise linear cosine approximation table for the LSFs */
+#define LSF_COS_TAB_SZ_FIX                      128
+
+/******************/
+/* NLSF quantizer */
+/******************/
+#define NLSF_W_Q                                2
+#define NLSF_VQ_MAX_VECTORS                     32
+#define NLSF_VQ_MAX_SURVIVORS                   32
+#define NLSF_QUANT_MAX_AMPLITUDE                4
+#define NLSF_QUANT_MAX_AMPLITUDE_EXT            10
+#define NLSF_QUANT_LEVEL_ADJ                    0.1
+#define NLSF_QUANT_DEL_DEC_STATES_LOG2          2
+#define NLSF_QUANT_DEL_DEC_STATES               ( 1 << NLSF_QUANT_DEL_DEC_STATES_LOG2 )
+
+/* Transition filtering for mode switching */
+#define TRANSITION_TIME_MS                      5120    /* 5120 = 64 * FRAME_LENGTH_MS * ( TRANSITION_INT_NUM - 1 ) = 64*(20*4)*/
+#define TRANSITION_NB                           3       /* Hardcoded in tables */
+#define TRANSITION_NA                           2       /* Hardcoded in tables */
+#define TRANSITION_INT_NUM                      5       /* Hardcoded in tables */
+#define TRANSITION_FRAMES                       ( TRANSITION_TIME_MS / MAX_FRAME_LENGTH_MS )
+#define TRANSITION_INT_STEPS                    ( TRANSITION_FRAMES  / ( TRANSITION_INT_NUM - 1 ) )
+
+/* BWE factors to apply after packet loss */
+#define BWE_AFTER_LOSS_Q16                      63570
+
+/* Defines for CN generation */
+#define CNG_BUF_MASK_MAX                        255     /* 2^floor(log2(MAX_FRAME_LENGTH))-1    */
+#define CNG_GAIN_SMTH_Q16                       4634    /* 0.25^(1/4)                           */
+#define CNG_NLSF_SMTH_Q16                       16348   /* 0.25                                 */
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/errors.h b/lib/rbcodec/codecs/libopus/silk/errors.h
new file mode 100644
index 0000000000..0591e0091d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/errors.h
@@ -0,0 +1,98 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_ERRORS_H
+#define SILK_ERRORS_H
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/******************/
+/* Error messages */
+/******************/
+#define SILK_NO_ERROR                               0
+
+/**************************/
+/* Encoder error messages */
+/**************************/
+
+/* Input length is not a multiple of 10 ms, or length is longer than the packet length */
+#define SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES        -101
+
+/* Sampling frequency not 8000, 12000 or 16000 Hertz */
+#define SILK_ENC_FS_NOT_SUPPORTED                   -102
+
+/* Packet size not 10, 20, 40, or 60 ms */
+#define SILK_ENC_PACKET_SIZE_NOT_SUPPORTED          -103
+
+/* Allocated payload buffer too short */
+#define SILK_ENC_PAYLOAD_BUF_TOO_SHORT              -104
+
+/* Loss rate not between 0 and 100 percent */
+#define SILK_ENC_INVALID_LOSS_RATE                  -105
+
+/* Complexity setting not valid, use 0...10 */
+#define SILK_ENC_INVALID_COMPLEXITY_SETTING         -106
+
+/* Inband FEC setting not valid, use 0 or 1 */
+#define SILK_ENC_INVALID_INBAND_FEC_SETTING         -107
+
+/* DTX setting not valid, use 0 or 1 */
+#define SILK_ENC_INVALID_DTX_SETTING                -108
+
+/* CBR setting not valid, use 0 or 1 */
+#define SILK_ENC_INVALID_CBR_SETTING                -109
+
+/* Internal encoder error */
+#define SILK_ENC_INTERNAL_ERROR                     -110
+
+/* Internal encoder error */
+#define SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR   -111
+
+/**************************/
+/* Decoder error messages */
+/**************************/
+
+/* Output sampling frequency lower than internal decoded sampling frequency */
+#define SILK_DEC_INVALID_SAMPLING_FREQUENCY         -200
+
+/* Payload size exceeded the maximum allowed 1024 bytes */
+#define SILK_DEC_PAYLOAD_TOO_LARGE                  -201
+
+/* Payload has bit errors */
+#define SILK_DEC_PAYLOAD_ERROR                      -202
+
+/* Payload has bit errors */
+#define SILK_DEC_INVALID_FRAME_SIZE                 -203
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/gain_quant.c b/lib/rbcodec/codecs/libopus/silk/gain_quant.c
new file mode 100644
index 0000000000..e91ec937e1
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/gain_quant.c
@@ -0,0 +1,141 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+#define OFFSET                  ( ( MIN_QGAIN_DB * 128 ) / 6 + 16 * 128 )
+#define SCALE_Q16               ( ( 65536 * ( N_LEVELS_QGAIN - 1 ) ) / ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) )
+#define INV_SCALE_Q16           ( ( 65536 * ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) ) / ( N_LEVELS_QGAIN - 1 ) )
+
+/* Gain scalar quantization with hysteresis, uniform on log scale */
+void silk_gains_quant(
+    opus_int8                   ind[ MAX_NB_SUBFR ],            /* O    gain indices                                */
+    opus_int32                  gain_Q16[ MAX_NB_SUBFR ],       /* I/O  gains (quantized out)                       */
+    opus_int8                   *prev_ind,                      /* I/O  last index in previous frame                */
+    const opus_int              conditional,                    /* I    first gain is delta coded if 1              */
+    const opus_int              nb_subfr                        /* I    number of subframes                         */
+)
+{
+    opus_int k, double_step_size_threshold;
+
+    for( k = 0; k < nb_subfr; k++ ) {
+        /* Convert to log scale, scale, floor() */
+        ind[ k ] = silk_SMULWB( SCALE_Q16, silk_lin2log( gain_Q16[ k ] ) - OFFSET );
+
+        /* Round towards previous quantized gain (hysteresis) */
+        if( ind[ k ] < *prev_ind ) {
+            ind[ k ]++;
+        }
+        ind[ k ] = silk_LIMIT_int( ind[ k ], 0, N_LEVELS_QGAIN - 1 );
+
+        /* Compute delta indices and limit */
+        if( k == 0 && conditional == 0 ) {
+            /* Full index */
+            ind[ k ] = silk_LIMIT_int( ind[ k ], *prev_ind + MIN_DELTA_GAIN_QUANT, N_LEVELS_QGAIN - 1 );
+            *prev_ind = ind[ k ];
+        } else {
+            /* Delta index */
+            ind[ k ] = ind[ k ] - *prev_ind;
+
+            /* Double the quantization step size for large gain increases, so that the max gain level can be reached */
+            double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind;
+            if( ind[ k ] > double_step_size_threshold ) {
+                ind[ k ] = double_step_size_threshold + silk_RSHIFT( ind[ k ] - double_step_size_threshold + 1, 1 );
+            }
+
+            ind[ k ] = silk_LIMIT_int( ind[ k ], MIN_DELTA_GAIN_QUANT, MAX_DELTA_GAIN_QUANT );
+
+            /* Accumulate deltas */
+            if( ind[ k ] > double_step_size_threshold ) {
+                *prev_ind += silk_LSHIFT( ind[ k ], 1 ) - double_step_size_threshold;
+            } else {
+                *prev_ind += ind[ k ];
+            }
+
+            /* Shift to make non-negative */
+            ind[ k ] -= MIN_DELTA_GAIN_QUANT;
+        }
+
+        /* Scale and convert to linear scale */
+        gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */
+    }
+}
+
+/* Gains scalar dequantization, uniform on log scale */
+void silk_gains_dequant(
+    opus_int32                  gain_Q16[ MAX_NB_SUBFR ],       /* O    quantized gains                             */
+    const opus_int8             ind[ MAX_NB_SUBFR ],            /* I    gain indices                                */
+    opus_int8                   *prev_ind,                      /* I/O  last index in previous frame                */
+    const opus_int              conditional,                    /* I    first gain is delta coded if 1              */
+    const opus_int              nb_subfr                        /* I    number of subframes                          */
+)
+{
+    opus_int   k, ind_tmp, double_step_size_threshold;
+
+    for( k = 0; k < nb_subfr; k++ ) {
+        if( k == 0 && conditional == 0 ) {
+            /* Gain index is not allowed to go down more than 16 steps (~21.8 dB) */
+            *prev_ind = silk_max_int( ind[ k ], *prev_ind - 16 );
+        } else {
+            /* Delta index */
+            ind_tmp = ind[ k ] + MIN_DELTA_GAIN_QUANT;
+
+            /* Accumulate deltas */
+            double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind;
+            if( ind_tmp > double_step_size_threshold ) {
+                *prev_ind += silk_LSHIFT( ind_tmp, 1 ) - double_step_size_threshold;
+            } else {
+                *prev_ind += ind_tmp;
+            }
+        }
+        *prev_ind = silk_LIMIT_int( *prev_ind, 0, N_LEVELS_QGAIN - 1 );
+
+        /* Scale and convert to linear scale */
+        gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */
+    }
+}
+
+/* Compute unique identifier of gain indices vector */
+opus_int32 silk_gains_ID(                                       /* O    returns unique identifier of gains          */
+    const opus_int8             ind[ MAX_NB_SUBFR ],            /* I    gain indices                                */
+    const opus_int              nb_subfr                        /* I    number of subframes                         */
+)
+{
+    opus_int   k;
+    opus_int32 gainsID;
+
+    gainsID = 0;
+    for( k = 0; k < nb_subfr; k++ ) {
+        gainsID = silk_ADD_LSHIFT32( ind[ k ], gainsID, 8 );
+    }
+
+    return gainsID;
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/init_decoder.c b/lib/rbcodec/codecs/libopus/silk/init_decoder.c
new file mode 100644
index 0000000000..139cf02014
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/init_decoder.c
@@ -0,0 +1,56 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/************************/
+/* Init Decoder State   */
+/************************/
+opus_int silk_init_decoder(
+    silk_decoder_state          *psDec                          /* I/O  Decoder state pointer                       */
+)
+{
+    /* Clear the entire encoder state, except anything copied */
+    silk_memset( psDec, 0, sizeof( silk_decoder_state ) );
+
+    /* Used to deactivate LSF interpolation */
+    psDec->first_frame_after_reset = 1;
+    psDec->prev_gain_Q16 = 65536;
+
+    /* Reset CNG state */
+    silk_CNG_Reset( psDec );
+
+    /* Reset PLC state */
+    silk_PLC_Reset( psDec );
+
+    return(0);
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/lin2log.c b/lib/rbcodec/codecs/libopus/silk/lin2log.c
new file mode 100644
index 0000000000..68ea030c89
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/lin2log.c
@@ -0,0 +1,46 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+/* Approximation of 128 * log2() (very close inverse of silk_log2lin()) */
+/* Convert input to a log scale    */
+opus_int32 silk_lin2log(
+    const opus_int32            inLin               /* I  input in linear scale                                         */
+)
+{
+    opus_int32 lz, frac_Q7;
+
+    silk_CLZ_FRAC( inLin, &lz, &frac_Q7 );
+
+    /* Piece-wise parabolic approximation */
+    return silk_LSHIFT( 31 - lz, 7 ) + silk_SMLAWB( frac_Q7, silk_MUL( frac_Q7, 128 - frac_Q7 ), 179 );
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/log2lin.c b/lib/rbcodec/codecs/libopus/silk/log2lin.c
new file mode 100644
index 0000000000..d80472c69f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/log2lin.c
@@ -0,0 +1,56 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+/* Approximation of 2^() (very close inverse of silk_lin2log()) */
+/* Convert input to a linear scale    */
+opus_int32 silk_log2lin( 
+    const opus_int32            inLog_Q7            /* I  input on log scale                                            */
+)
+{
+    opus_int32 out, frac_Q7;
+
+    if( inLog_Q7 < 0 ) {
+        return 0;
+    }
+
+    out = silk_LSHIFT( 1, silk_RSHIFT( inLog_Q7, 7 ) );
+    frac_Q7 = inLog_Q7 & 0x7F;
+    if( inLog_Q7 < 2048 ) {
+        /* Piece-wise parabolic approximation */
+        out = silk_ADD_RSHIFT32( out, silk_MUL( out, silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ), 7 );
+    } else {
+        /* Piece-wise parabolic approximation */
+        out = silk_MLA( out, silk_RSHIFT( out, 7 ), silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) );
+    }
+    return out;
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/macros.h b/lib/rbcodec/codecs/libopus/silk/macros.h
new file mode 100644
index 0000000000..d84cd73522
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/macros.h
@@ -0,0 +1,135 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_MACROS_H
+#define SILK_MACROS_H
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/* This is an inline header file for general platform. */
+
+/* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */
+#define silk_SMULWB(a32, b32)            ((((a32) >> 16) * (opus_int32)((opus_int16)(b32))) + ((((a32) & 0x0000FFFF) * (opus_int32)((opus_int16)(b32))) >> 16))
+
+/* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */
+#define silk_SMLAWB(a32, b32, c32)       ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16)))
+
+/* (a32 * (b32 >> 16)) >> 16 */
+#define silk_SMULWT(a32, b32)            (((a32) >> 16) * ((b32) >> 16) + ((((a32) & 0x0000FFFF) * ((b32) >> 16)) >> 16))
+
+/* a32 + (b32 * (c32 >> 16)) >> 16 */
+#define silk_SMLAWT(a32, b32, c32)       ((a32) + (((b32) >> 16) * ((c32) >> 16)) + ((((b32) & 0x0000FFFF) * ((c32) >> 16)) >> 16))
+
+/* (opus_int32)((opus_int16)(a3))) * (opus_int32)((opus_int16)(b32)) output have to be 32bit int */
+#define silk_SMULBB(a32, b32)            ((opus_int32)((opus_int16)(a32)) * (opus_int32)((opus_int16)(b32)))
+
+/* a32 + (opus_int32)((opus_int16)(b32)) * (opus_int32)((opus_int16)(c32)) output have to be 32bit int */
+#define silk_SMLABB(a32, b32, c32)       ((a32) + ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32)))
+
+/* (opus_int32)((opus_int16)(a32)) * (b32 >> 16) */
+#define silk_SMULBT(a32, b32)            ((opus_int32)((opus_int16)(a32)) * ((b32) >> 16))
+
+/* a32 + (opus_int32)((opus_int16)(b32)) * (c32 >> 16) */
+#define silk_SMLABT(a32, b32, c32)       ((a32) + ((opus_int32)((opus_int16)(b32))) * ((c32) >> 16))
+
+/* a64 + (b32 * c32) */
+#define silk_SMLAL(a64, b32, c32)        (silk_ADD64((a64), ((opus_int64)(b32) * (opus_int64)(c32))))
+
+/* (a32 * b32) >> 16 */
+#define silk_SMULWW(a32, b32)            silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16))
+
+/* a32 + ((b32 * c32) >> 16) */
+#define silk_SMLAWW(a32, b32, c32)       silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16))
+
+/* add/subtract with output saturated */
+#define silk_ADD_SAT32(a, b)             ((((opus_uint32)(a) + (opus_uint32)(b)) & 0x80000000) == 0 ?                              \
+                                        ((((a) & (b)) & 0x80000000) != 0 ? silk_int32_MIN : (a)+(b)) :   \
+                                        ((((a) | (b)) & 0x80000000) == 0 ? silk_int32_MAX : (a)+(b)) )
+
+#define silk_SUB_SAT32(a, b)             ((((opus_uint32)(a)-(opus_uint32)(b)) & 0x80000000) == 0 ?                                        \
+                                        (( (a) & ((b)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a)-(b)) :    \
+                                        ((((a)^0x80000000) & (b)  & 0x80000000) ? silk_int32_MAX : (a)-(b)) )
+
+static inline opus_int32 silk_CLZ16(opus_int16 in16)
+{
+    opus_int32 out32 = 0;
+    if( in16 == 0 ) {
+        return 16;
+    }
+    /* test nibbles */
+    if( in16 & 0xFF00 ) {
+        if( in16 & 0xF000 ) {
+            in16 >>= 12;
+        } else {
+            out32 += 4;
+            in16 >>= 8;
+        }
+    } else {
+        if( in16 & 0xFFF0 ) {
+            out32 += 8;
+            in16 >>= 4;
+        } else {
+            out32 += 12;
+        }
+    }
+    /* test bits and return */
+    if( in16 & 0xC ) {
+        if( in16 & 0x8 )
+            return out32 + 0;
+        else
+            return out32 + 1;
+    } else {
+        if( in16 & 0xE )
+            return out32 + 2;
+        else
+            return out32 + 3;
+    }
+}
+
+static inline opus_int32 silk_CLZ32(opus_int32 in32)
+{
+    /* test highest 16 bits and convert to opus_int16 */
+    if( in32 & 0xFFFF0000 ) {
+        return silk_CLZ16((opus_int16)(in32 >> 16));
+    } else {
+        return silk_CLZ16((opus_int16)in32) + 16;
+    }
+}
+
+/* Row based */
+#define matrix_ptr(Matrix_base_adr, row, column, N)         *(Matrix_base_adr + ((row)*(N)+(column)))
+#define matrix_adr(Matrix_base_adr, row, column, N)          (Matrix_base_adr + ((row)*(N)+(column)))
+
+/* Column based */
+#ifndef matrix_c_ptr
+#   define matrix_c_ptr(Matrix_base_adr, row, column, M)    *(Matrix_base_adr + ((row)+(M)*(column)))
+#endif
+
+#endif /* SILK_MACROS_H */
+
diff --git a/lib/rbcodec/codecs/libopus/silk/main.h b/lib/rbcodec/codecs/libopus/silk/main.h
new file mode 100644
index 0000000000..32675f6931
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/main.h
@@ -0,0 +1,434 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_MAIN_H
+#define SILK_MAIN_H
+
+#include "SigProc_FIX.h"
+#include "define.h"
+#include "structs.h"
+#include "tables.h"
+#include "PLC.h"
+#include "control.h"
+#include "debug.h"
+#include "entenc.h"
+#include "entdec.h"
+
+/* Convert Left/Right stereo signal to adaptive Mid/Side representation */
+void silk_stereo_LR_to_MS(
+    stereo_enc_state            *state,                         /* I/O  State                                       */
+    opus_int16                  x1[],                           /* I/O  Left input signal, becomes mid signal       */
+    opus_int16                  x2[],                           /* I/O  Right input signal, becomes side signal     */
+    opus_int8                   ix[ 2 ][ 3 ],                   /* O    Quantization indices                        */
+    opus_int8                   *mid_only_flag,                 /* O    Flag: only mid signal coded                 */
+    opus_int32                  mid_side_rates_bps[],           /* O    Bitrates for mid and side signals           */
+    opus_int32                  total_rate_bps,                 /* I    Total bitrate                               */
+    opus_int                    prev_speech_act_Q8,             /* I    Speech activity level in previous frame     */
+    opus_int                    toMono,                         /* I    Last frame before a stereo->mono transition */
+    opus_int                    fs_kHz,                         /* I    Sample rate (kHz)                           */
+    opus_int                    frame_length                    /* I    Number of samples                           */
+);
+
+/* Convert adaptive Mid/Side representation to Left/Right stereo signal */
+void silk_stereo_MS_to_LR(
+    stereo_dec_state            *state,                         /* I/O  State                                       */
+    opus_int16                  x1[],                           /* I/O  Left input signal, becomes mid signal       */
+    opus_int16                  x2[],                           /* I/O  Right input signal, becomes side signal     */
+    const opus_int32            pred_Q13[],                     /* I    Predictors                                  */
+    opus_int                    fs_kHz,                         /* I    Samples rate (kHz)                          */
+    opus_int                    frame_length                    /* I    Number of samples                           */
+);
+
+/* Find least-squares prediction gain for one signal based on another and quantize it */
+opus_int32 silk_stereo_find_predictor(                          /* O    Returns predictor in Q13                    */
+    opus_int32                  *ratio_Q14,                     /* O    Ratio of residual and mid energies          */
+    const opus_int16            x[],                            /* I    Basis signal                                */
+    const opus_int16            y[],                            /* I    Target signal                               */
+    opus_int32                  mid_res_amp_Q0[],               /* I/O  Smoothed mid, residual norms                */
+    opus_int                    length,                         /* I    Number of samples                           */
+    opus_int                    smooth_coef_Q16                 /* I    Smoothing coefficient                       */
+);
+
+/* Quantize mid/side predictors */
+void silk_stereo_quant_pred(
+    opus_int32                  pred_Q13[],                     /* I/O  Predictors (out: quantized)                 */
+    opus_int8                   ix[ 2 ][ 3 ]                    /* O    Quantization indices                        */
+);
+
+/* Entropy code the mid/side quantization indices */
+void silk_stereo_encode_pred(
+    ec_enc                      *psRangeEnc,                    /* I/O  Compressor data structure                   */
+    opus_int8                   ix[ 2 ][ 3 ]                    /* I    Quantization indices                        */
+);
+
+/* Entropy code the mid-only flag */
+void silk_stereo_encode_mid_only(
+    ec_enc                      *psRangeEnc,                    /* I/O  Compressor data structure                   */
+    opus_int8                   mid_only_flag
+);
+
+/* Decode mid/side predictors */
+void silk_stereo_decode_pred(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int32                  pred_Q13[]                      /* O    Predictors                                  */
+);
+
+/* Decode mid-only flag */
+void silk_stereo_decode_mid_only(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    *decode_only_mid                /* O    Flag that only mid channel has been coded   */
+);
+
+/* Encodes signs of excitation */
+void silk_encode_signs(
+    ec_enc                      *psRangeEnc,                        /* I/O  Compressor data structure                   */
+    const opus_int8             pulses[],                           /* I    pulse signal                                */
+    opus_int                    length,                             /* I    length of input                             */
+    const opus_int              signalType,                         /* I    Signal type                                 */
+    const opus_int              quantOffsetType,                    /* I    Quantization offset type                    */
+    const opus_int              sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
+);
+
+/* Decodes signs of excitation */
+void silk_decode_signs(
+    ec_dec                      *psRangeDec,                        /* I/O  Compressor data structure                   */
+    opus_int                    pulses[],                           /* I/O  pulse signal                                */
+    opus_int                    length,                             /* I    length of input                             */
+    const opus_int              signalType,                         /* I    Signal type                                 */
+    const opus_int              quantOffsetType,                    /* I    Quantization offset type                    */
+    const opus_int              sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
+);
+
+/* Check encoder control struct */
+opus_int check_control_input(
+    silk_EncControlStruct        *encControl                    /* I    Control structure                           */
+);
+
+/* Control internal sampling rate */
+opus_int silk_control_audio_bandwidth(
+    silk_encoder_state          *psEncC,                        /* I/O  Pointer to Silk encoder state               */
+    silk_EncControlStruct       *encControl                     /* I    Control structure                           */
+);
+
+/* Control SNR of redidual quantizer */
+opus_int silk_control_SNR(
+    silk_encoder_state          *psEncC,                        /* I/O  Pointer to Silk encoder state               */
+    opus_int32                  TargetRate_bps                  /* I    Target max bitrate (bps)                    */
+);
+
+/***************/
+/* Shell coder */
+/***************/
+
+/* Encode quantization indices of excitation */
+void silk_encode_pulses(
+    ec_enc                      *psRangeEnc,                    /* I/O  compressor data structure                   */
+    const opus_int              signalType,                     /* I    Signal type                                 */
+    const opus_int              quantOffsetType,                /* I    quantOffsetType                             */
+    opus_int8                   pulses[],                       /* I    quantization indices                        */
+    const opus_int              frame_length                    /* I    Frame length                                */
+);
+
+/* Shell encoder, operates on one shell code frame of 16 pulses */
+void silk_shell_encoder(
+    ec_enc                      *psRangeEnc,                    /* I/O  compressor data structure                   */
+    const opus_int              *pulses0                        /* I    data: nonnegative pulse amplitudes          */
+);
+
+/* Shell decoder, operates on one shell code frame of 16 pulses */
+void silk_shell_decoder(
+    opus_int                    *pulses0,                       /* O    data: nonnegative pulse amplitudes          */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    const opus_int              pulses4                         /* I    number of pulses per pulse-subframe         */
+);
+
+/* Gain scalar quantization with hysteresis, uniform on log scale */
+void silk_gains_quant(
+    opus_int8                   ind[ MAX_NB_SUBFR ],            /* O    gain indices                                */
+    opus_int32                  gain_Q16[ MAX_NB_SUBFR ],       /* I/O  gains (quantized out)                       */
+    opus_int8                   *prev_ind,                      /* I/O  last index in previous frame                */
+    const opus_int              conditional,                    /* I    first gain is delta coded if 1              */
+    const opus_int              nb_subfr                        /* I    number of subframes                         */
+);
+
+/* Gains scalar dequantization, uniform on log scale */
+void silk_gains_dequant(
+    opus_int32                  gain_Q16[ MAX_NB_SUBFR ],       /* O    quantized gains                             */
+    const opus_int8             ind[ MAX_NB_SUBFR ],            /* I    gain indices                                */
+    opus_int8                   *prev_ind,                      /* I/O  last index in previous frame                */
+    const opus_int              conditional,                    /* I    first gain is delta coded if 1              */
+    const opus_int              nb_subfr                        /* I    number of subframes                          */
+);
+
+/* Compute unique identifier of gain indices vector */
+opus_int32 silk_gains_ID(                                       /* O    returns unique identifier of gains          */
+    const opus_int8             ind[ MAX_NB_SUBFR ],            /* I    gain indices                                */
+    const opus_int              nb_subfr                        /* I    number of subframes                         */
+);
+
+/* Interpolate two vectors */
+void silk_interpolate(
+    opus_int16                  xi[ MAX_LPC_ORDER ],            /* O    interpolated vector                         */
+    const opus_int16            x0[ MAX_LPC_ORDER ],            /* I    first vector                                */
+    const opus_int16            x1[ MAX_LPC_ORDER ],            /* I    second vector                               */
+    const opus_int              ifact_Q2,                       /* I    interp. factor, weight on 2nd vector        */
+    const opus_int              d                               /* I    number of parameters                        */
+);
+
+/* LTP tap quantizer */
+void silk_quant_LTP_gains(
+    opus_int16                  B_Q14[ MAX_NB_SUBFR * LTP_ORDER ],          /* I/O  (un)quantized LTP gains         */
+    opus_int8                   cbk_index[ MAX_NB_SUBFR ],                  /* O    Codebook Index                  */
+    opus_int8                   *periodicity_index,                         /* O    Periodicity Index               */
+    const opus_int32            W_Q18[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ],  /* I    Error Weights in Q18            */
+    opus_int                    mu_Q9,                                      /* I    Mu value (R/D tradeoff)         */
+    opus_int                    lowComplexity,                              /* I    Flag for low complexity         */
+    const opus_int              nb_subfr                                    /* I    number of subframes             */
+);
+
+/* Entropy constrained matrix-weighted VQ, for a single input data vector */
+void silk_VQ_WMat_EC(
+    opus_int8                   *ind,                           /* O    index of best codebook vector               */
+    opus_int32                  *rate_dist_Q14,                 /* O    best weighted quant error + mu * rate       */
+    const opus_int16            *in_Q14,                        /* I    input vector to be quantized                */
+    const opus_int32            *W_Q18,                         /* I    weighting matrix                            */
+    const opus_int8             *cb_Q7,                         /* I    codebook                                    */
+    const opus_uint8            *cl_Q5,                         /* I    code length for each codebook vector        */
+    const opus_int              mu_Q9,                          /* I    tradeoff betw. weighted error and rate      */
+    opus_int                    L                               /* I    number of vectors in codebook               */
+);
+
+/************************************/
+/* Noise shaping quantization (NSQ) */
+/************************************/
+void silk_NSQ(
+    const silk_encoder_state    *psEncC,                                    /* I/O  Encoder State                   */
+    silk_nsq_state              *NSQ,                                       /* I/O  NSQ state                       */
+    SideInfoIndices             *psIndices,                                 /* I/O  Quantization Indices            */
+    const opus_int32            x_Q3[],                                     /* I    Prefiltered input signal        */
+    opus_int8                   pulses[],                                   /* O    Quantized pulse signal          */
+    const opus_int16            PredCoef_Q12[ 2 * MAX_LPC_ORDER ],          /* I    Short term prediction coefs     */
+    const opus_int16            LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],    /* I    Long term prediction coefs      */
+    const opus_int16            AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs             */
+    const opus_int              HarmShapeGain_Q14[ MAX_NB_SUBFR ],          /* I    Long term shaping coefs         */
+    const opus_int              Tilt_Q14[ MAX_NB_SUBFR ],                   /* I    Spectral tilt                   */
+    const opus_int32            LF_shp_Q14[ MAX_NB_SUBFR ],                 /* I    Low frequency shaping coefs     */
+    const opus_int32            Gains_Q16[ MAX_NB_SUBFR ],                  /* I    Quantization step sizes         */
+    const opus_int              pitchL[ MAX_NB_SUBFR ],                     /* I    Pitch lags                      */
+    const opus_int              Lambda_Q10,                                 /* I    Rate/distortion tradeoff        */
+    const opus_int              LTP_scale_Q14                               /* I    LTP state scaling               */
+);
+
+/* Noise shaping using delayed decision */
+void silk_NSQ_del_dec(
+    const silk_encoder_state    *psEncC,                                    /* I/O  Encoder State                   */
+    silk_nsq_state              *NSQ,                                       /* I/O  NSQ state                       */
+    SideInfoIndices             *psIndices,                                 /* I/O  Quantization Indices            */
+    const opus_int32            x_Q3[],                                     /* I    Prefiltered input signal        */
+    opus_int8                   pulses[],                                   /* O    Quantized pulse signal          */
+    const opus_int16            PredCoef_Q12[ 2 * MAX_LPC_ORDER ],          /* I    Short term prediction coefs     */
+    const opus_int16            LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],    /* I    Long term prediction coefs      */
+    const opus_int16            AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs             */
+    const opus_int              HarmShapeGain_Q14[ MAX_NB_SUBFR ],          /* I    Long term shaping coefs         */
+    const opus_int              Tilt_Q14[ MAX_NB_SUBFR ],                   /* I    Spectral tilt                   */
+    const opus_int32            LF_shp_Q14[ MAX_NB_SUBFR ],                 /* I    Low frequency shaping coefs     */
+    const opus_int32            Gains_Q16[ MAX_NB_SUBFR ],                  /* I    Quantization step sizes         */
+    const opus_int              pitchL[ MAX_NB_SUBFR ],                     /* I    Pitch lags                      */
+    const opus_int              Lambda_Q10,                                 /* I    Rate/distortion tradeoff        */
+    const opus_int              LTP_scale_Q14                               /* I    LTP state scaling               */
+);
+
+/************/
+/* Silk VAD */
+/************/
+/* Initialize the Silk VAD */
+opus_int silk_VAD_Init(                                         /* O    Return value, 0 if success                  */
+    silk_VAD_state              *psSilk_VAD                     /* I/O  Pointer to Silk VAD state                   */
+);
+
+/* Get speech activity level in Q8 */
+opus_int silk_VAD_GetSA_Q8(                                     /* O    Return value, 0 if success                  */
+    silk_encoder_state          *psEncC,                        /* I/O  Encoder state                               */
+    const opus_int16            pIn[]                           /* I    PCM input                                   */
+);
+
+/* Low-pass filter with variable cutoff frequency based on  */
+/* piece-wise linear interpolation between elliptic filters */
+/* Start by setting transition_frame_no = 1;                */
+void silk_LP_variable_cutoff(
+    silk_LP_state               *psLP,                          /* I/O  LP filter state                             */
+    opus_int16                  *frame,                         /* I/O  Low-pass filtered output signal             */
+    const opus_int              frame_length                    /* I    Frame length                                */
+);
+
+/******************/
+/* NLSF Quantizer */
+/******************/
+/* Limit, stabilize, convert and quantize NLSFs */
+void silk_process_NLSFs(
+    silk_encoder_state          *psEncC,                            /* I/O  Encoder state                               */
+    opus_int16                  PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O    Prediction coefficients                     */
+    opus_int16                  pNLSF_Q15[         MAX_LPC_ORDER ], /* I/O  Normalized LSFs (quant out) (0 - (2^15-1))  */
+    const opus_int16            prev_NLSFq_Q15[    MAX_LPC_ORDER ]  /* I    Previous Normalized LSFs (0 - (2^15-1))     */
+);
+
+opus_int32 silk_NLSF_encode(                                    /* O    Returns RD value in Q25                     */
+          opus_int8             *NLSFIndices,                   /* I    Codebook path vector [ LPC_ORDER + 1 ]      */
+          opus_int16            *pNLSF_Q15,                     /* I/O  Quantized NLSF vector [ LPC_ORDER ]         */
+    const silk_NLSF_CB_struct   *psNLSF_CB,                     /* I    Codebook object                             */
+    const opus_int16            *pW_QW,                         /* I    NLSF weight vector [ LPC_ORDER ]            */
+    const opus_int              NLSF_mu_Q20,                    /* I    Rate weight for the RD optimization         */
+    const opus_int              nSurvivors,                     /* I    Max survivors after first stage             */
+    const opus_int              signalType                      /* I    Signal type: 0/1/2                          */
+);
+
+/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */
+void silk_NLSF_VQ(
+    opus_int32                  err_Q26[],                      /* O    Quantization errors [K]                     */
+    const opus_int16            in_Q15[],                       /* I    Input vectors to be quantized [LPC_order]   */
+    const opus_uint8            pCB_Q8[],                       /* I    Codebook vectors [K*LPC_order]              */
+    const opus_int              K,                              /* I    Number of codebook vectors                  */
+    const opus_int              LPC_order                       /* I    Number of LPCs                              */
+);
+
+/* Delayed-decision quantizer for NLSF residuals */
+opus_int32 silk_NLSF_del_dec_quant(                             /* O    Returns RD value in Q25                     */
+    opus_int8                   indices[],                      /* O    Quantization indices [ order ]              */
+    const opus_int16            x_Q10[],                        /* I    Input [ order ]                             */
+    const opus_int16            w_Q5[],                         /* I    Weights [ order ]                           */
+    const opus_uint8            pred_coef_Q8[],                 /* I    Backward predictor coefs [ order ]          */
+    const opus_int16            ec_ix[],                        /* I    Indices to entropy coding tables [ order ]  */
+    const opus_uint8            ec_rates_Q5[],                  /* I    Rates []                                    */
+    const opus_int              quant_step_size_Q16,            /* I    Quantization step size                      */
+    const opus_int16            inv_quant_step_size_Q6,         /* I    Inverse quantization step size              */
+    const opus_int32            mu_Q20,                         /* I    R/D tradeoff                                */
+    const opus_int16            order                           /* I    Number of input values                      */
+);
+
+/* Unpack predictor values and indices for entropy coding tables */
+void silk_NLSF_unpack(
+          opus_int16            ec_ix[],                        /* O    Indices to entropy tables [ LPC_ORDER ]     */
+          opus_uint8            pred_Q8[],                      /* O    LSF predictor [ LPC_ORDER ]                 */
+    const silk_NLSF_CB_struct   *psNLSF_CB,                     /* I    Codebook object                             */
+    const opus_int              CB1_index                       /* I    Index of vector in first LSF codebook       */
+);
+
+/***********************/
+/* NLSF vector decoder */
+/***********************/
+void silk_NLSF_decode(
+          opus_int16            *pNLSF_Q15,                     /* O    Quantized NLSF vector [ LPC_ORDER ]         */
+          opus_int8             *NLSFIndices,                   /* I    Codebook path vector [ LPC_ORDER + 1 ]      */
+    const silk_NLSF_CB_struct   *psNLSF_CB                      /* I    Codebook object                             */
+);
+
+/****************************************************/
+/* Decoder Functions                                */
+/****************************************************/
+opus_int silk_init_decoder(
+    silk_decoder_state          *psDec                          /* I/O  Decoder state pointer                       */
+);
+
+/* Set decoder sampling rate */
+opus_int silk_decoder_set_fs(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state pointer                       */
+    opus_int                    fs_kHz,                         /* I    Sampling frequency (kHz)                    */
+    opus_int32                  fs_API_Hz                       /* I    API Sampling frequency (Hz)                 */
+);
+
+/****************/
+/* Decode frame */
+/****************/
+opus_int silk_decode_frame(
+    silk_decoder_state          *psDec,                         /* I/O  Pointer to Silk decoder state               */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int16                  pOut[],                         /* O    Pointer to output speech frame              */
+    opus_int32                  *pN,                            /* O    Pointer to size of output frame             */
+    opus_int                    lostFlag,                       /* I    0: no loss, 1 loss, 2 decode fec            */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+);
+
+/* Decode indices from bitstream */
+void silk_decode_indices(
+    silk_decoder_state          *psDec,                         /* I/O  State                                       */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    FrameIndex,                     /* I    Frame number                                */
+    opus_int                    decode_LBRR,                    /* I    Flag indicating LBRR data is being decoded  */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+);
+
+/* Decode parameters from payload */
+void silk_decode_parameters(
+    silk_decoder_state          *psDec,                         /* I/O  State                                       */
+    silk_decoder_control        *psDecCtrl,                     /* I/O  Decoder control                             */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+);
+
+/* Core decoder. Performs inverse NSQ operation LTP + LPC */
+void silk_decode_core(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state                               */
+    silk_decoder_control        *psDecCtrl,                     /* I    Decoder control                             */
+    opus_int16                  xq[],                           /* O    Decoded speech                              */
+    const opus_int              pulses[ MAX_FRAME_LENGTH ]      /* I    Pulse signal                                */
+);
+
+/* Decode quantization indices of excitation (Shell coding) */
+void silk_decode_pulses(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    pulses[],                       /* O    Excitation signal                           */
+    const opus_int              signalType,                     /* I    Sigtype                                     */
+    const opus_int              quantOffsetType,                /* I    quantOffsetType                             */
+    const opus_int              frame_length                    /* I    Frame length                                */
+);
+
+/******************/
+/* CNG */
+/******************/
+
+/* Reset CNG */
+void silk_CNG_Reset(
+    silk_decoder_state          *psDec                          /* I/O  Decoder state                               */
+);
+
+/* Updates CNG estimate, and applies the CNG when packet was lost */
+void silk_CNG(
+    silk_decoder_state          *psDec,                         /* I/O  Decoder state                               */
+    silk_decoder_control        *psDecCtrl,                     /* I/O  Decoder control                             */
+    opus_int16                  frame[],                        /* I/O  Signal                                      */
+    opus_int                    length                          /* I    Length of residual                          */
+);
+
+/* Encoding of various parameters */
+void silk_encode_indices(
+    silk_encoder_state          *psEncC,                        /* I/O  Encoder state                               */
+    ec_enc                      *psRangeEnc,                    /* I/O  Compressor data structure                   */
+    opus_int                    FrameIndex,                     /* I    Frame number                                */
+    opus_int                    encode_LBRR,                    /* I    Flag indicating LBRR data is being encoded  */
+    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
+);
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/pitch_est_defines.h b/lib/rbcodec/codecs/libopus/silk/pitch_est_defines.h
new file mode 100644
index 0000000000..0b6770eb91
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/pitch_est_defines.h
@@ -0,0 +1,88 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_PE_DEFINES_H
+#define SILK_PE_DEFINES_H
+
+#include "SigProc_FIX.h"
+
+/********************************************************/
+/* Definitions for pitch estimator                      */
+/********************************************************/
+
+#define PE_MAX_FS_KHZ               16 /* Maximum sampling frequency used */
+
+#define PE_MAX_NB_SUBFR             4
+#define PE_SUBFR_LENGTH_MS          5   /* 5 ms */
+
+#define PE_LTP_MEM_LENGTH_MS        ( 4 * PE_SUBFR_LENGTH_MS )
+
+#define PE_MAX_FRAME_LENGTH_MS      ( PE_LTP_MEM_LENGTH_MS + PE_MAX_NB_SUBFR * PE_SUBFR_LENGTH_MS )
+#define PE_MAX_FRAME_LENGTH         ( PE_MAX_FRAME_LENGTH_MS * PE_MAX_FS_KHZ )
+#define PE_MAX_FRAME_LENGTH_ST_1    ( PE_MAX_FRAME_LENGTH >> 2 )
+#define PE_MAX_FRAME_LENGTH_ST_2    ( PE_MAX_FRAME_LENGTH >> 1 )
+
+#define PE_MAX_LAG_MS               18           /* 18 ms -> 56 Hz */
+#define PE_MIN_LAG_MS               2            /* 2 ms -> 500 Hz */
+#define PE_MAX_LAG                  ( PE_MAX_LAG_MS * PE_MAX_FS_KHZ )
+#define PE_MIN_LAG                  ( PE_MIN_LAG_MS * PE_MAX_FS_KHZ )
+
+#define PE_D_SRCH_LENGTH            24
+
+#define PE_NB_STAGE3_LAGS           5
+
+#define PE_NB_CBKS_STAGE2           3
+#define PE_NB_CBKS_STAGE2_EXT       11
+
+#define PE_NB_CBKS_STAGE3_MAX       34
+#define PE_NB_CBKS_STAGE3_MID       24
+#define PE_NB_CBKS_STAGE3_MIN       16
+
+#define PE_NB_CBKS_STAGE3_10MS      12
+#define PE_NB_CBKS_STAGE2_10MS      3
+
+#define PE_SHORTLAG_BIAS            0.2f    /* for logarithmic weighting    */
+#define PE_PREVLAG_BIAS             0.2f    /* for logarithmic weighting    */
+#define PE_FLATCONTOUR_BIAS         0.05f
+
+#define SILK_PE_MIN_COMPLEX         0
+#define SILK_PE_MID_COMPLEX         1
+#define SILK_PE_MAX_COMPLEX         2
+
+/* Tables for 20 ms frames */
+extern const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ];
+extern const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ];
+extern const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ];
+extern const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ];
+
+/* Tables for 10 ms frames */
+extern const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ 3 ];
+extern const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 12 ];
+extern const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ];
+
+#endif
+
diff --git a/lib/rbcodec/codecs/libopus/silk/pitch_est_tables.c b/lib/rbcodec/codecs/libopus/silk/pitch_est_tables.c
new file mode 100644
index 0000000000..e191686b0e
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/pitch_est_tables.c
@@ -0,0 +1,99 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "typedef.h"
+#include "pitch_est_defines.h"
+
+const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ PE_NB_CBKS_STAGE2_10MS ] =
+{
+    {0, 1, 0},
+    {0, 0, 1}
+};
+
+const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ PE_NB_CBKS_STAGE3_10MS ] =
+{
+    { 0, 0, 1,-1, 1,-1, 2,-2, 2,-2, 3,-3},
+    { 0, 1, 0, 1,-1, 2,-1, 2,-2, 3,-2, 3}
+};
+
+const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ] =
+{
+    {-3, 7},
+    {-2, 7}
+};
+
+const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ] =
+{
+    {0, 2,-1,-1,-1, 0, 0, 1, 1, 0, 1},
+    {0, 1, 0, 0, 0, 0, 0, 1, 0, 0, 0},
+    {0, 0, 1, 0, 0, 0, 1, 0, 0, 0, 0},
+    {0,-1, 2, 1, 0, 1, 1, 0, 0,-1,-1}
+};
+
+const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ] =
+{
+    {0, 0, 1,-1, 0, 1,-1, 0,-1, 1,-2, 2,-2,-2, 2,-3, 2, 3,-3,-4, 3,-4, 4, 4,-5, 5,-6,-5, 6,-7, 6, 5, 8,-9},
+    {0, 0, 1, 0, 0, 0, 0, 0, 0, 0,-1, 1, 0, 0, 1,-1, 0, 1,-1,-1, 1,-1, 2, 1,-1, 2,-2,-2, 2,-2, 2, 2, 3,-3},
+    {0, 1, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 1,-1, 1, 0, 0, 2, 1,-1, 2,-1,-1, 2,-1, 2, 2,-1, 3,-2,-2,-2, 3},
+    {0, 1, 0, 0, 1, 0, 1,-1, 2,-1, 2,-1, 2, 3,-2, 3,-2,-2, 4, 4,-3, 5,-3,-4, 6,-4, 6, 5,-5, 8,-6,-5,-7, 9}
+};
+
+const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ] =
+{
+    /* Lags to search for low number of stage3 cbks */
+    {
+        {-5,8},
+        {-1,6},
+        {-1,6},
+        {-4,10}
+    },
+    /* Lags to search for middle number of stage3 cbks */
+    {
+        {-6,10},
+        {-2,6},
+        {-1,6},
+        {-5,10}
+    },
+    /* Lags to search for max number of stage3 cbks */
+    {
+        {-9,12},
+        {-3,7},
+        {-2,7},
+        {-7,13}
+    }
+};
+
+const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ] =
+{
+    PE_NB_CBKS_STAGE3_MIN,
+    PE_NB_CBKS_STAGE3_MID,
+    PE_NB_CBKS_STAGE3_MAX
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler.c b/lib/rbcodec/codecs/libopus/silk/resampler.c
new file mode 100644
index 0000000000..ab81b6013f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler.c
@@ -0,0 +1,215 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/*
+ * Matrix of resampling methods used:
+ *                                 Fs_out (kHz)
+ *                        8      12     16     24     48
+ *
+ *               8        C      UF     U      UF     UF
+ *              12        AF     C      UF     U      UF
+ * Fs_in (kHz)  16        D      AF     C      UF     UF
+ *              24        AF     D      AF     C      U
+ *              48        AF     AF     AF     D      C
+ *
+ * C   -> Copy (no resampling)
+ * D   -> Allpass-based 2x downsampling
+ * U   -> Allpass-based 2x upsampling
+ * UF  -> Allpass-based 2x upsampling followed by FIR interpolation
+ * AF  -> AR2 filter followed by FIR interpolation
+ */
+
+#include "resampler_private.h"
+
+/* Tables with delay compensation values to equalize total delay for different modes */
+static const opus_int8 delay_matrix_enc[ 5 ][ 3 ] = {
+/* in  \ out  8  12  16 */
+/*  8 */   {  6,  0,  3 },
+/* 12 */   {  0,  7,  3 },
+/* 16 */   {  0,  1, 10 },
+/* 24 */   {  0,  2,  6 },
+/* 48 */   { 18, 10, 12 }
+};
+
+static const opus_int8 delay_matrix_dec[ 3 ][ 5 ] = {
+/* in  \ out  8  12  16  24  48 */
+/*  8 */   {  4,  0,  2,  0,  0 },
+/* 12 */   {  0,  9,  4,  7,  4 },
+/* 16 */   {  0,  3, 12,  7,  7 }
+};
+
+/* Simple way to make [8000, 12000, 16000, 24000, 48000] to [0, 1, 2, 3, 4] */
+#define rateID(R) ( ( ( ((R)>>12) - ((R)>16000) ) >> ((R)>24000) ) - 1 )
+
+#define USE_silk_resampler_copy                     (0)
+#define USE_silk_resampler_private_up2_HQ_wrapper   (1)
+#define USE_silk_resampler_private_IIR_FIR          (2)
+#define USE_silk_resampler_private_down_FIR         (3)
+
+/* Initialize/reset the resampler state for a given pair of input/output sampling rates */
+opus_int silk_resampler_init(
+    silk_resampler_state_struct *S,                 /* I/O  Resampler state                                             */
+    opus_int32                  Fs_Hz_in,           /* I    Input sampling rate (Hz)                                    */
+    opus_int32                  Fs_Hz_out,          /* I    Output sampling rate (Hz)                                   */
+    opus_int                    forEnc              /* I    If 1: encoder; if 0: decoder                                */
+)
+{
+    opus_int up2x;
+
+    /* Clear state */
+    silk_memset( S, 0, sizeof( silk_resampler_state_struct ) );
+
+    /* Input checking */
+    if( forEnc ) {
+        if( ( Fs_Hz_in  != 8000 && Fs_Hz_in  != 12000 && Fs_Hz_in  != 16000 && Fs_Hz_in  != 24000 && Fs_Hz_in  != 48000 ) ||
+            ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 ) ) {
+            silk_assert( 0 );
+            return -1;
+        }
+        S->inputDelay = delay_matrix_enc[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ];
+    } else {
+        if( ( Fs_Hz_in  != 8000 && Fs_Hz_in  != 12000 && Fs_Hz_in  != 16000 ) ||
+            ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 && Fs_Hz_out != 24000 && Fs_Hz_out != 48000 ) ) {
+            silk_assert( 0 );
+            return -1;
+        }
+        S->inputDelay = delay_matrix_dec[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ];
+    }
+
+    S->Fs_in_kHz  = silk_DIV32_16( Fs_Hz_in,  1000 );
+    S->Fs_out_kHz = silk_DIV32_16( Fs_Hz_out, 1000 );
+
+    /* Number of samples processed per batch */
+    S->batchSize = S->Fs_in_kHz * RESAMPLER_MAX_BATCH_SIZE_MS;
+
+    /* Find resampler with the right sampling ratio */
+    up2x = 0;
+    if( Fs_Hz_out > Fs_Hz_in ) {
+        /* Upsample */
+        if( Fs_Hz_out == silk_MUL( Fs_Hz_in, 2 ) ) {                            /* Fs_out : Fs_in = 2 : 1 */
+            /* Special case: directly use 2x upsampler */
+            S->resampler_function = USE_silk_resampler_private_up2_HQ_wrapper;
+        } else {
+            /* Default resampler */
+            S->resampler_function = USE_silk_resampler_private_IIR_FIR;
+            up2x = 1;
+        }
+    } else if ( Fs_Hz_out < Fs_Hz_in ) {
+        /* Downsample */
+         S->resampler_function = USE_silk_resampler_private_down_FIR;
+        if( silk_MUL( Fs_Hz_out, 4 ) == silk_MUL( Fs_Hz_in, 3 ) ) {             /* Fs_out : Fs_in = 3 : 4 */
+            S->FIR_Fracs = 3;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0;
+            S->Coefs = silk_Resampler_3_4_COEFS;
+        } else if( silk_MUL( Fs_Hz_out, 3 ) == silk_MUL( Fs_Hz_in, 2 ) ) {      /* Fs_out : Fs_in = 2 : 3 */
+            S->FIR_Fracs = 2;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0;
+            S->Coefs = silk_Resampler_2_3_COEFS;
+        } else if( silk_MUL( Fs_Hz_out, 2 ) == Fs_Hz_in ) {                     /* Fs_out : Fs_in = 1 : 2 */
+            S->FIR_Fracs = 1;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR1;
+            S->Coefs = silk_Resampler_1_2_COEFS;
+        } else if( silk_MUL( Fs_Hz_out, 3 ) == Fs_Hz_in ) {                     /* Fs_out : Fs_in = 1 : 3 */
+            S->FIR_Fracs = 1;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
+            S->Coefs = silk_Resampler_1_3_COEFS;
+        } else if( silk_MUL( Fs_Hz_out, 4 ) == Fs_Hz_in ) {                     /* Fs_out : Fs_in = 1 : 4 */
+            S->FIR_Fracs = 1;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
+            S->Coefs = silk_Resampler_1_4_COEFS;
+        } else if( silk_MUL( Fs_Hz_out, 6 ) == Fs_Hz_in ) {                     /* Fs_out : Fs_in = 1 : 6 */
+            S->FIR_Fracs = 1;
+            S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
+            S->Coefs = silk_Resampler_1_6_COEFS;
+        } else {
+            /* None available */
+            silk_assert( 0 );
+            return -1;
+        }
+    } else {
+        /* Input and output sampling rates are equal: copy */
+        S->resampler_function = USE_silk_resampler_copy;
+    }
+
+    /* Ratio of input/output samples */
+    S->invRatio_Q16 = silk_LSHIFT32( silk_DIV32( silk_LSHIFT32( Fs_Hz_in, 14 + up2x ), Fs_Hz_out ), 2 );
+    /* Make sure the ratio is rounded up */
+    while( silk_SMULWW( S->invRatio_Q16, Fs_Hz_out ) < silk_LSHIFT32( Fs_Hz_in, up2x ) ) {
+        S->invRatio_Q16++;
+    }
+
+    return 0;
+}
+
+/* Resampler: convert from one sampling rate to another */
+/* Input and output sampling rate are at most 48000 Hz  */
+opus_int silk_resampler(
+    silk_resampler_state_struct *S,                 /* I/O  Resampler state                                             */
+    opus_int16                  out[],              /* O    Output signal                                               */
+    const opus_int16            in[],               /* I    Input signal                                                */
+    opus_int32                  inLen               /* I    Number of input samples                                     */
+)
+{
+    opus_int nSamples;
+
+    /* Need at least 1 ms of input data */
+    silk_assert( inLen >= S->Fs_in_kHz );
+    /* Delay can't exceed the 1 ms of buffering */
+    silk_assert( S->inputDelay <= S->Fs_in_kHz );
+
+    nSamples = S->Fs_in_kHz - S->inputDelay;
+
+    /* Copy to delay buffer */
+    silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) );
+
+    switch( S->resampler_function ) {
+        case USE_silk_resampler_private_up2_HQ_wrapper:
+            silk_resampler_private_up2_HQ_wrapper( S, out, S->delayBuf, S->Fs_in_kHz );
+            silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
+            break;
+        case USE_silk_resampler_private_IIR_FIR:
+            silk_resampler_private_IIR_FIR( S, out, S->delayBuf, S->Fs_in_kHz );
+            silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
+            break;
+        case USE_silk_resampler_private_down_FIR:
+            silk_resampler_private_down_FIR( S, out, S->delayBuf, S->Fs_in_kHz );
+            silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
+            break;
+        default:
+            silk_memcpy( out, S->delayBuf, S->Fs_in_kHz * sizeof( opus_int16 ) );
+            silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) );
+    }
+
+    /* Copy to delay buffer */
+    silk_memcpy( S->delayBuf, &in[ inLen - S->inputDelay ], S->inputDelay * sizeof( opus_int16 ) );
+
+    return 0;
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_private.h b/lib/rbcodec/codecs/libopus/silk/resampler_private.h
new file mode 100644
index 0000000000..45d342c78d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_private.h
@@ -0,0 +1,88 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_RESAMPLER_PRIVATE_H
+#define SILK_RESAMPLER_PRIVATE_H
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "SigProc_FIX.h"
+#include "resampler_structs.h"
+#include "resampler_rom.h"
+
+/* Number of input samples to process in the inner loop */
+#define RESAMPLER_MAX_BATCH_SIZE_MS             10
+#define RESAMPLER_MAX_FS_KHZ                    48
+#define RESAMPLER_MAX_BATCH_SIZE_IN             ( RESAMPLER_MAX_BATCH_SIZE_MS * RESAMPLER_MAX_FS_KHZ )
+
+/* Description: Hybrid IIR/FIR polyphase implementation of resampling */
+void silk_resampler_private_IIR_FIR(
+    void                            *SS,            /* I/O  Resampler state             */
+    opus_int16                      out[],          /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    opus_int32                      inLen           /* I    Number of input samples     */
+);
+
+/* Description: Hybrid IIR/FIR polyphase implementation of resampling */
+void silk_resampler_private_down_FIR(
+    void                            *SS,            /* I/O  Resampler state             */
+    opus_int16                      out[],          /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    opus_int32                      inLen           /* I    Number of input samples     */
+);
+
+/* Upsample by a factor 2, high quality */
+void silk_resampler_private_up2_HQ_wrapper(
+    void                            *SS,            /* I/O  Resampler state (unused)    */
+    opus_int16                      *out,           /* O    Output signal [ 2 * len ]   */
+    const opus_int16                *in,            /* I    Input signal [ len ]        */
+    opus_int32                      len             /* I    Number of input samples     */
+);
+
+/* Upsample by a factor 2, high quality */
+void silk_resampler_private_up2_HQ(
+    opus_int32                      *S,             /* I/O  Resampler state [ 6 ]       */
+    opus_int16                      *out,           /* O    Output signal [ 2 * len ]   */
+    const opus_int16                *in,            /* I    Input signal [ len ]        */
+    opus_int32                      len             /* I    Number of input samples     */
+);
+
+/* Second order AR filter */
+void silk_resampler_private_AR2(
+    opus_int32                      S[],            /* I/O  State vector [ 2 ]          */
+    opus_int32                      out_Q8[],       /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    const opus_int16                A_Q14[],        /* I    AR coefficients, Q14        */
+    opus_int32                      len             /* I    Signal length               */
+);
+
+#ifdef __cplusplus
+}
+#endif
+#endif /* SILK_RESAMPLER_PRIVATE_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_private_AR2.c b/lib/rbcodec/codecs/libopus/silk/resampler_private_AR2.c
new file mode 100644
index 0000000000..2ec7625e7e
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_private_AR2.c
@@ -0,0 +1,55 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+#include "resampler_private.h"
+
+/* Second order AR filter with single delay elements */
+void silk_resampler_private_AR2(
+    opus_int32                      S[],            /* I/O  State vector [ 2 ]          */
+    opus_int32                      out_Q8[],       /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    const opus_int16                A_Q14[],        /* I    AR coefficients, Q14        */
+    opus_int32                      len             /* I    Signal length               */
+)
+{
+    opus_int32    k;
+    opus_int32    out32;
+
+    for( k = 0; k < len; k++ ) {
+        out32       = silk_ADD_LSHIFT32( S[ 0 ], (opus_int32)in[ k ], 8 );
+        out_Q8[ k ] = out32;
+        out32       = silk_LSHIFT( out32, 2 );
+        S[ 0 ]      = silk_SMLAWB( S[ 1 ], out32, A_Q14[ 0 ] );
+        S[ 1 ]      = silk_SMULWB( out32, A_Q14[ 1 ] );
+    }
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_private_IIR_FIR.c b/lib/rbcodec/codecs/libopus/silk/resampler_private_IIR_FIR.c
new file mode 100644
index 0000000000..105be35f16
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_private_IIR_FIR.c
@@ -0,0 +1,103 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+#include "resampler_private.h"
+
+static inline opus_int16 *silk_resampler_private_IIR_FIR_INTERPOL(
+    opus_int16  *out,
+    opus_int16  *buf,
+    opus_int32  max_index_Q16,
+    opus_int32  index_increment_Q16
+)
+{
+    opus_int32 index_Q16, res_Q15;
+    opus_int16 *buf_ptr;
+    opus_int32 table_index;
+
+    /* Interpolate upsampled signal and store in output array */
+    for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) {
+        table_index = silk_SMULWB( index_Q16 & 0xFFFF, 12 );
+        buf_ptr = &buf[ index_Q16 >> 16 ];
+
+        res_Q15 = silk_SMULBB(          buf_ptr[ 0 ], silk_resampler_frac_FIR_12[      table_index ][ 0 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 1 ], silk_resampler_frac_FIR_12[      table_index ][ 1 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 2 ], silk_resampler_frac_FIR_12[      table_index ][ 2 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 3 ], silk_resampler_frac_FIR_12[      table_index ][ 3 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 4 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 3 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 5 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 2 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 6 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 1 ] );
+        res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 7 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 0 ] );
+        *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q15, 15 ) );
+    }
+    return out;
+}
+/* Upsample using a combination of allpass-based 2x upsampling and FIR interpolation */
+void silk_resampler_private_IIR_FIR(
+    void                            *SS,            /* I/O  Resampler state             */
+    opus_int16                      out[],          /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    opus_int32                      inLen           /* I    Number of input samples     */
+)
+{
+    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
+    opus_int32 nSamplesIn;
+    opus_int32 max_index_Q16, index_increment_Q16;
+    opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
+
+    /* Copy buffered samples to start of buffer */
+    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+
+    /* Iterate over blocks of frameSizeIn input samples */
+    index_increment_Q16 = S->invRatio_Q16;
+    while( 1 ) {
+        nSamplesIn = silk_min( inLen, S->batchSize );
+
+        /* Upsample 2x */
+        silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn );
+
+        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 );         /* + 1 because 2x upsampling */
+        out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 );
+        in += nSamplesIn;
+        inLen -= nSamplesIn;
+
+        if( inLen > 0 ) {
+            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
+            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+        } else {
+            break;
+        }
+    }
+
+    /* Copy last part of filtered signal to the state for the next call */
+    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_private_down_FIR.c b/lib/rbcodec/codecs/libopus/silk/resampler_private_down_FIR.c
new file mode 100644
index 0000000000..5d4cb1f072
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_private_down_FIR.c
@@ -0,0 +1,189 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+#include "resampler_private.h"
+
+static inline opus_int16 *silk_resampler_private_down_FIR_INTERPOL(
+    opus_int16          *out,
+    opus_int32          *buf,
+    const opus_int16    *FIR_Coefs,
+    opus_int            FIR_Order,
+    opus_int            FIR_Fracs,
+    opus_int32          max_index_Q16,
+    opus_int32          index_increment_Q16
+)
+{
+    opus_int32 index_Q16, res_Q6;
+    opus_int32 *buf_ptr;
+    opus_int32 interpol_ind;
+    const opus_int16 *interpol_ptr;
+
+    switch( FIR_Order ) {
+        case RESAMPLER_DOWN_ORDER_FIR0:
+            for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) {
+                /* Integer part gives pointer to buffered input */
+                buf_ptr = buf + silk_RSHIFT( index_Q16, 16 );
+
+                /* Fractional part gives interpolation coefficients */
+                interpol_ind = silk_SMULWB( index_Q16 & 0xFFFF, FIR_Fracs );
+
+                /* Inner product */
+                interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * interpol_ind ];
+                res_Q6 = silk_SMULWB(         buf_ptr[ 0 ], interpol_ptr[ 0 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], interpol_ptr[ 1 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], interpol_ptr[ 2 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], interpol_ptr[ 3 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], interpol_ptr[ 4 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 5 ], interpol_ptr[ 5 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 6 ], interpol_ptr[ 6 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 7 ], interpol_ptr[ 7 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 8 ], interpol_ptr[ 8 ] );
+                interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * ( FIR_Fracs - 1 - interpol_ind ) ];
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 17 ], interpol_ptr[ 0 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 16 ], interpol_ptr[ 1 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 15 ], interpol_ptr[ 2 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 14 ], interpol_ptr[ 3 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 13 ], interpol_ptr[ 4 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 12 ], interpol_ptr[ 5 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 11 ], interpol_ptr[ 6 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 10 ], interpol_ptr[ 7 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[  9 ], interpol_ptr[ 8 ] );
+
+                /* Scale down, saturate and store in output array */
+                *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) );
+            }
+            break;
+        case RESAMPLER_DOWN_ORDER_FIR1:
+            for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) {
+                /* Integer part gives pointer to buffered input */
+                buf_ptr = buf + silk_RSHIFT( index_Q16, 16 );
+
+                /* Inner product */
+                res_Q6 = silk_SMULWB(         silk_ADD32( buf_ptr[  0 ], buf_ptr[ 23 ] ), FIR_Coefs[  0 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  1 ], buf_ptr[ 22 ] ), FIR_Coefs[  1 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  2 ], buf_ptr[ 21 ] ), FIR_Coefs[  2 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  3 ], buf_ptr[ 20 ] ), FIR_Coefs[  3 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  4 ], buf_ptr[ 19 ] ), FIR_Coefs[  4 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  5 ], buf_ptr[ 18 ] ), FIR_Coefs[  5 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  6 ], buf_ptr[ 17 ] ), FIR_Coefs[  6 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  7 ], buf_ptr[ 16 ] ), FIR_Coefs[  7 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  8 ], buf_ptr[ 15 ] ), FIR_Coefs[  8 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  9 ], buf_ptr[ 14 ] ), FIR_Coefs[  9 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 13 ] ), FIR_Coefs[ 10 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 12 ] ), FIR_Coefs[ 11 ] );
+
+                /* Scale down, saturate and store in output array */
+                *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) );
+            }
+            break;
+        case RESAMPLER_DOWN_ORDER_FIR2:
+            for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) {
+                /* Integer part gives pointer to buffered input */
+                buf_ptr = buf + silk_RSHIFT( index_Q16, 16 );
+
+                /* Inner product */
+                res_Q6 = silk_SMULWB(         silk_ADD32( buf_ptr[  0 ], buf_ptr[ 35 ] ), FIR_Coefs[  0 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  1 ], buf_ptr[ 34 ] ), FIR_Coefs[  1 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  2 ], buf_ptr[ 33 ] ), FIR_Coefs[  2 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  3 ], buf_ptr[ 32 ] ), FIR_Coefs[  3 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  4 ], buf_ptr[ 31 ] ), FIR_Coefs[  4 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  5 ], buf_ptr[ 30 ] ), FIR_Coefs[  5 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  6 ], buf_ptr[ 29 ] ), FIR_Coefs[  6 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  7 ], buf_ptr[ 28 ] ), FIR_Coefs[  7 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  8 ], buf_ptr[ 27 ] ), FIR_Coefs[  8 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[  9 ], buf_ptr[ 26 ] ), FIR_Coefs[  9 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 25 ] ), FIR_Coefs[ 10 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 24 ] ), FIR_Coefs[ 11 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 12 ], buf_ptr[ 23 ] ), FIR_Coefs[ 12 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 13 ], buf_ptr[ 22 ] ), FIR_Coefs[ 13 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 14 ], buf_ptr[ 21 ] ), FIR_Coefs[ 14 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 15 ], buf_ptr[ 20 ] ), FIR_Coefs[ 15 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 16 ], buf_ptr[ 19 ] ), FIR_Coefs[ 16 ] );
+                res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 17 ], buf_ptr[ 18 ] ), FIR_Coefs[ 17 ] );
+
+                /* Scale down, saturate and store in output array */
+                *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) );
+            }
+            break;
+        default:
+            silk_assert( 0 );
+    }
+    return out;
+}
+
+/* Resample with a 2nd order AR filter followed by FIR interpolation */
+void silk_resampler_private_down_FIR(
+    void                            *SS,            /* I/O  Resampler state             */
+    opus_int16                      out[],          /* O    Output signal               */
+    const opus_int16                in[],           /* I    Input signal                */
+    opus_int32                      inLen           /* I    Number of input samples     */
+)
+{
+    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
+    opus_int32 nSamplesIn;
+    opus_int32 max_index_Q16, index_increment_Q16;
+    opus_int32 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + SILK_RESAMPLER_MAX_FIR_ORDER ];
+    const opus_int16 *FIR_Coefs;
+
+    /* Copy buffered samples to start of buffer */
+    silk_memcpy( buf, S->sFIR, S->FIR_Order * sizeof( opus_int32 ) );
+
+    FIR_Coefs = &S->Coefs[ 2 ];
+
+    /* Iterate over blocks of frameSizeIn input samples */
+    index_increment_Q16 = S->invRatio_Q16;
+    while( 1 ) {
+        nSamplesIn = silk_min( inLen, S->batchSize );
+
+        /* Second-order AR filter (output in Q8) */
+        silk_resampler_private_AR2( S->sIIR, &buf[ S->FIR_Order ], in, S->Coefs, nSamplesIn );
+
+        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 );
+
+        /* Interpolate filtered signal */
+        out = silk_resampler_private_down_FIR_INTERPOL( out, buf, FIR_Coefs, S->FIR_Order,
+            S->FIR_Fracs, max_index_Q16, index_increment_Q16 );
+
+        in += nSamplesIn;
+        inLen -= nSamplesIn;
+
+        if( inLen > 1 ) {
+            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
+            silk_memcpy( buf, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
+        } else {
+            break;
+        }
+    }
+
+    /* Copy last part of filtered signal to the state for the next call */
+    silk_memcpy( S->sFIR, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_private_up2_HQ.c b/lib/rbcodec/codecs/libopus/silk/resampler_private_up2_HQ.c
new file mode 100644
index 0000000000..6479a154da
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_private_up2_HQ.c
@@ -0,0 +1,113 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+#include "resampler_private.h"
+
+/* Upsample by a factor 2, high quality */
+/* Uses 2nd order allpass filters for the 2x upsampling, followed by a      */
+/* notch filter just above Nyquist.                                         */
+void silk_resampler_private_up2_HQ(
+    opus_int32                      *S,             /* I/O  Resampler state [ 6 ]       */
+    opus_int16                      *out,           /* O    Output signal [ 2 * len ]   */
+    const opus_int16                *in,            /* I    Input signal [ len ]        */
+    opus_int32                      len             /* I    Number of input samples     */
+)
+{
+    opus_int32 k;
+    opus_int32 in32, out32_1, out32_2, Y, X;
+
+    silk_assert( silk_resampler_up2_hq_0[ 0 ] > 0 );
+    silk_assert( silk_resampler_up2_hq_0[ 1 ] > 0 );
+    silk_assert( silk_resampler_up2_hq_0[ 2 ] < 0 );
+    silk_assert( silk_resampler_up2_hq_1[ 0 ] > 0 );
+    silk_assert( silk_resampler_up2_hq_1[ 1 ] > 0 );
+    silk_assert( silk_resampler_up2_hq_1[ 2 ] < 0 );
+
+    /* Internal variables and state are in Q10 format */
+    for( k = 0; k < len; k++ ) {
+        /* Convert to Q10 */
+        in32 = silk_LSHIFT( (opus_int32)in[ k ], 10 );
+
+        /* First all-pass section for even output sample */
+        Y       = silk_SUB32( in32, S[ 0 ] );
+        X       = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 0 ] );
+        out32_1 = silk_ADD32( S[ 0 ], X );
+        S[ 0 ]  = silk_ADD32( in32, X );
+
+        /* Second all-pass section for even output sample */
+        Y       = silk_SUB32( out32_1, S[ 1 ] );
+        X       = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 1 ] );
+        out32_2 = silk_ADD32( S[ 1 ], X );
+        S[ 1 ]  = silk_ADD32( out32_1, X );
+
+        /* Third all-pass section for even output sample */
+        Y       = silk_SUB32( out32_2, S[ 2 ] );
+        X       = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_0[ 2 ] );
+        out32_1 = silk_ADD32( S[ 2 ], X );
+        S[ 2 ]  = silk_ADD32( out32_2, X );
+
+        /* Apply gain in Q15, convert back to int16 and store to output */
+        out[ 2 * k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) );
+
+        /* First all-pass section for odd output sample */
+        Y       = silk_SUB32( in32, S[ 3 ] );
+        X       = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 0 ] );
+        out32_1 = silk_ADD32( S[ 3 ], X );
+        S[ 3 ]  = silk_ADD32( in32, X );
+
+        /* Second all-pass section for odd output sample */
+        Y       = silk_SUB32( out32_1, S[ 4 ] );
+        X       = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 1 ] );
+        out32_2 = silk_ADD32( S[ 4 ], X );
+        S[ 4 ]  = silk_ADD32( out32_1, X );
+
+        /* Third all-pass section for odd output sample */
+        Y       = silk_SUB32( out32_2, S[ 5 ] );
+        X       = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_1[ 2 ] );
+        out32_1 = silk_ADD32( S[ 5 ], X );
+        S[ 5 ]  = silk_ADD32( out32_2, X );
+
+        /* Apply gain in Q15, convert back to int16 and store to output */
+        out[ 2 * k + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) );
+    }
+}
+
+void silk_resampler_private_up2_HQ_wrapper(
+    void                            *SS,            /* I/O  Resampler state (unused)    */
+    opus_int16                      *out,           /* O    Output signal [ 2 * len ]   */
+    const opus_int16                *in,            /* I    Input signal [ len ]        */
+    opus_int32                      len             /* I    Number of input samples     */
+)
+{
+    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
+    silk_resampler_private_up2_HQ( S->sIIR, out, in, len );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_rom.c b/lib/rbcodec/codecs/libopus/silk/resampler_rom.c
new file mode 100644
index 0000000000..bbbd6d1690
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_rom.c
@@ -0,0 +1,96 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/* Filter coefficients for IIR/FIR polyphase resampling     *
+ * Total size: 179 Words (358 Bytes)                        */
+
+#include "resampler_private.h"
+
+/* Matlab code for the notch filter coefficients: */
+/* B = [1, 0.147, 1];  A = [1, 0.107, 0.89]; G = 0.93; freqz(G * B, A, 2^14, 16e3); axis([0, 8000, -10, 1]) */
+/* fprintf('\t%6d, %6d, %6d, %6d\n', round(B(2)*2^16), round(-A(2)*2^16), round((1-A(3))*2^16), round(G*2^15)) */
+/* const opus_int16 silk_resampler_up2_hq_notch[ 4 ] = { 9634,  -7012,   7209,  30474 }; */
+
+/* Tables with IIR and FIR coefficients for fractional downsamplers (123 Words) */
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = {
+        -20694, -13867,
+           -49,     64,     17,   -157,    353,   -496,    163,  11047,  22205,
+           -39,      6,     91,   -170,    186,     23,   -896,   6336,  19928,
+           -19,    -36,    102,    -89,    -24,    328,   -951,   2568,  15909,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = {
+        -14457, -14019,
+            64,    128,   -122,     36,    310,   -768,    584,   9267,  17733,
+            12,    128,     18,   -142,    288,   -117,   -865,   4123,  14459,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_2_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR1 / 2 ] = {
+           616, -14323,
+           -10,     39,     58,    -46,    -84,    120,    184,   -315,   -541,   1284,   5380,   9024,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_3_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = {
+         16102, -15162,
+           -13,      0,     20,     26,      5,    -31,    -43,     -4,     65,     90,      7,   -157,   -248,    -44,    593,   1583,   2612,   3271,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_4_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = {
+         22500, -15099,
+             3,    -14,    -20,    -15,      2,     25,     37,     25,    -16,    -71,   -107,    -79,     50,    292,    623,    982,   1288,   1464,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_6_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = {
+         27540, -15257,
+            17,     12,      8,      1,    -10,    -22,    -30,    -32,    -22,      3,     44,    100,    168,    243,    317,    381,    429,    455,
+};
+
+silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ] = {
+     -2797,  -6507,
+      4697,  10739,
+      1567,   8276,
+};
+
+/* Table with interplation fractions of 1/24, 3/24, 5/24, ... , 23/24 : 23/24 (46 Words) */
+silk_DWORD_ALIGN const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ] = {
+        {  189,  -600,   617, 30567 },
+        {  117,  -159, -1070, 29704 },
+        {   52,   221, -2392, 28276 },
+        {   -4,   529, -3350, 26341 },
+        {  -48,   758, -3956, 23973 },
+        {  -80,   905, -4235, 21254 },
+        {  -99,   972, -4222, 18278 },
+        { -107,   967, -3957, 15143 },
+        { -103,   896, -3487, 11950 },
+        {  -91,   773, -2865,  8798 },
+        {  -71,   611, -2143,  5784 },
+        {  -46,   425, -1375,  2996 },
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_rom.h b/lib/rbcodec/codecs/libopus/silk/resampler_rom.h
new file mode 100644
index 0000000000..473b24a2b5
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_rom.h
@@ -0,0 +1,68 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_FIX_RESAMPLER_ROM_H
+#define SILK_FIX_RESAMPLER_ROM_H
+
+#ifdef  __cplusplus
+extern "C"
+{
+#endif
+
+#include "typedef.h"
+#include "resampler_structs.h"
+
+#define RESAMPLER_DOWN_ORDER_FIR0               18
+#define RESAMPLER_DOWN_ORDER_FIR1               24
+#define RESAMPLER_DOWN_ORDER_FIR2               36
+#define RESAMPLER_ORDER_FIR_12                  8
+
+/* Tables for 2x downsampler */
+static const opus_int16 silk_resampler_down2_0 = 9872;
+static const opus_int16 silk_resampler_down2_1 = 39809 - 65536;
+
+/* Tables for 2x upsampler, high quality */
+static const opus_int16 silk_resampler_up2_hq_0[ 3 ] = { 1746, 14986, 39083 - 65536 };
+static const opus_int16 silk_resampler_up2_hq_1[ 3 ] = { 6854, 25769, 55542 - 65536 };
+
+/* Tables with IIR and FIR coefficients for fractional downsamplers */
+extern const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ];
+extern const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ];
+extern const opus_int16 silk_Resampler_1_2_COEFS[ 2 +     RESAMPLER_DOWN_ORDER_FIR1 / 2 ];
+extern const opus_int16 silk_Resampler_1_3_COEFS[ 2 +     RESAMPLER_DOWN_ORDER_FIR2 / 2 ];
+extern const opus_int16 silk_Resampler_1_4_COEFS[ 2 +     RESAMPLER_DOWN_ORDER_FIR2 / 2 ];
+extern const opus_int16 silk_Resampler_1_6_COEFS[ 2 +     RESAMPLER_DOWN_ORDER_FIR2 / 2 ];
+extern const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ];
+
+/* Table with interplation fractions of 1/24, 3/24, ..., 23/24 */
+extern const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ];
+
+#ifdef  __cplusplus
+}
+#endif
+
+#endif /* SILK_FIX_RESAMPLER_ROM_H */
diff --git a/lib/rbcodec/codecs/libopus/silk/resampler_structs.h b/lib/rbcodec/codecs/libopus/silk/resampler_structs.h
new file mode 100644
index 0000000000..4c28bd0a2f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/resampler_structs.h
@@ -0,0 +1,57 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_RESAMPLER_STRUCTS_H
+#define SILK_RESAMPLER_STRUCTS_H
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define SILK_RESAMPLER_MAX_FIR_ORDER                 36
+#define SILK_RESAMPLER_MAX_IIR_ORDER                 6
+
+typedef struct _silk_resampler_state_struct{
+    opus_int32       sIIR[ SILK_RESAMPLER_MAX_IIR_ORDER ]; /* this must be the first element of this struct */
+    opus_int32       sFIR[ SILK_RESAMPLER_MAX_FIR_ORDER ];
+    opus_int16       delayBuf[ 48 ];
+    opus_int         resampler_function;
+    opus_int         batchSize;
+    opus_int32       invRatio_Q16;
+    opus_int         FIR_Order;
+    opus_int         FIR_Fracs;
+    opus_int         Fs_in_kHz;
+    opus_int         Fs_out_kHz;
+    opus_int         inputDelay;
+    const opus_int16 *Coefs;
+} silk_resampler_state_struct;
+
+#ifdef __cplusplus
+}
+#endif
+#endif /* SILK_RESAMPLER_STRUCTS_H */
+
diff --git a/lib/rbcodec/codecs/libopus/silk/shell_coder.c b/lib/rbcodec/codecs/libopus/silk/shell_coder.c
new file mode 100644
index 0000000000..1cc77fd88e
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/shell_coder.c
@@ -0,0 +1,151 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* shell coder; pulse-subframe length is hardcoded */
+
+static inline void combine_pulses(
+    opus_int         *out,   /* O    combined pulses vector [len] */
+    const opus_int   *in,    /* I    input vector       [2 * len] */
+    const opus_int   len     /* I    number of OUTPUT samples     */
+)
+{
+    opus_int k;
+    for( k = 0; k < len; k++ ) {
+        out[ k ] = in[ 2 * k ] + in[ 2 * k + 1 ];
+    }
+}
+
+static inline void encode_split(
+    ec_enc                      *psRangeEnc,    /* I/O  compressor data structure                   */
+    const opus_int              p_child1,       /* I    pulse amplitude of first child subframe     */
+    const opus_int              p,              /* I    pulse amplitude of current subframe         */
+    const opus_uint8            *shell_table    /* I    table of shell cdfs                         */
+)
+{
+    if( p > 0 ) {
+        ec_enc_icdf( psRangeEnc, p_child1, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 );
+    }
+}
+
+static inline void decode_split(
+    opus_int                    *p_child1,      /* O    pulse amplitude of first child subframe     */
+    opus_int                    *p_child2,      /* O    pulse amplitude of second child subframe    */
+    ec_dec                      *psRangeDec,    /* I/O  Compressor data structure                   */
+    const opus_int              p,              /* I    pulse amplitude of current subframe         */
+    const opus_uint8            *shell_table    /* I    table of shell cdfs                         */
+)
+{
+    if( p > 0 ) {
+        p_child1[ 0 ] = ec_dec_icdf( psRangeDec, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 );
+        p_child2[ 0 ] = p - p_child1[ 0 ];
+    } else {
+        p_child1[ 0 ] = 0;
+        p_child2[ 0 ] = 0;
+    }
+}
+
+/* Shell encoder, operates on one shell code frame of 16 pulses */
+void silk_shell_encoder(
+    ec_enc                      *psRangeEnc,                    /* I/O  compressor data structure                   */
+    const opus_int              *pulses0                        /* I    data: nonnegative pulse amplitudes          */
+)
+{
+    opus_int pulses1[ 8 ], pulses2[ 4 ], pulses3[ 2 ], pulses4[ 1 ];
+
+    /* this function operates on one shell code frame of 16 pulses */
+    silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 );
+
+    /* tree representation per pulse-subframe */
+    combine_pulses( pulses1, pulses0, 8 );
+    combine_pulses( pulses2, pulses1, 4 );
+    combine_pulses( pulses3, pulses2, 2 );
+    combine_pulses( pulses4, pulses3, 1 );
+
+    encode_split( psRangeEnc, pulses3[  0 ], pulses4[ 0 ], silk_shell_code_table3 );
+
+    encode_split( psRangeEnc, pulses2[  0 ], pulses3[ 0 ], silk_shell_code_table2 );
+
+    encode_split( psRangeEnc, pulses1[  0 ], pulses2[ 0 ], silk_shell_code_table1 );
+    encode_split( psRangeEnc, pulses0[  0 ], pulses1[ 0 ], silk_shell_code_table0 );
+    encode_split( psRangeEnc, pulses0[  2 ], pulses1[ 1 ], silk_shell_code_table0 );
+
+    encode_split( psRangeEnc, pulses1[  2 ], pulses2[ 1 ], silk_shell_code_table1 );
+    encode_split( psRangeEnc, pulses0[  4 ], pulses1[ 2 ], silk_shell_code_table0 );
+    encode_split( psRangeEnc, pulses0[  6 ], pulses1[ 3 ], silk_shell_code_table0 );
+
+    encode_split( psRangeEnc, pulses2[  2 ], pulses3[ 1 ], silk_shell_code_table2 );
+
+    encode_split( psRangeEnc, pulses1[  4 ], pulses2[ 2 ], silk_shell_code_table1 );
+    encode_split( psRangeEnc, pulses0[  8 ], pulses1[ 4 ], silk_shell_code_table0 );
+    encode_split( psRangeEnc, pulses0[ 10 ], pulses1[ 5 ], silk_shell_code_table0 );
+
+    encode_split( psRangeEnc, pulses1[  6 ], pulses2[ 3 ], silk_shell_code_table1 );
+    encode_split( psRangeEnc, pulses0[ 12 ], pulses1[ 6 ], silk_shell_code_table0 );
+    encode_split( psRangeEnc, pulses0[ 14 ], pulses1[ 7 ], silk_shell_code_table0 );
+}
+
+
+/* Shell decoder, operates on one shell code frame of 16 pulses */
+void silk_shell_decoder(
+    opus_int                    *pulses0,                       /* O    data: nonnegative pulse amplitudes          */
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    const opus_int              pulses4                         /* I    number of pulses per pulse-subframe         */
+)
+{
+    opus_int pulses3[ 2 ], pulses2[ 4 ], pulses1[ 8 ];
+
+    /* this function operates on one shell code frame of 16 pulses */
+    silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 );
+
+    decode_split( &pulses3[  0 ], &pulses3[  1 ], psRangeDec, pulses4,      silk_shell_code_table3 );
+
+    decode_split( &pulses2[  0 ], &pulses2[  1 ], psRangeDec, pulses3[ 0 ], silk_shell_code_table2 );
+
+    decode_split( &pulses1[  0 ], &pulses1[  1 ], psRangeDec, pulses2[ 0 ], silk_shell_code_table1 );
+    decode_split( &pulses0[  0 ], &pulses0[  1 ], psRangeDec, pulses1[ 0 ], silk_shell_code_table0 );
+    decode_split( &pulses0[  2 ], &pulses0[  3 ], psRangeDec, pulses1[ 1 ], silk_shell_code_table0 );
+
+    decode_split( &pulses1[  2 ], &pulses1[  3 ], psRangeDec, pulses2[ 1 ], silk_shell_code_table1 );
+    decode_split( &pulses0[  4 ], &pulses0[  5 ], psRangeDec, pulses1[ 2 ], silk_shell_code_table0 );
+    decode_split( &pulses0[  6 ], &pulses0[  7 ], psRangeDec, pulses1[ 3 ], silk_shell_code_table0 );
+
+    decode_split( &pulses2[  2 ], &pulses2[  3 ], psRangeDec, pulses3[ 1 ], silk_shell_code_table2 );
+
+    decode_split( &pulses1[  4 ], &pulses1[  5 ], psRangeDec, pulses2[ 2 ], silk_shell_code_table1 );
+    decode_split( &pulses0[  8 ], &pulses0[  9 ], psRangeDec, pulses1[ 4 ], silk_shell_code_table0 );
+    decode_split( &pulses0[ 10 ], &pulses0[ 11 ], psRangeDec, pulses1[ 5 ], silk_shell_code_table0 );
+
+    decode_split( &pulses1[  6 ], &pulses1[  7 ], psRangeDec, pulses2[ 3 ], silk_shell_code_table1 );
+    decode_split( &pulses0[ 12 ], &pulses0[ 13 ], psRangeDec, pulses1[ 6 ], silk_shell_code_table0 );
+    decode_split( &pulses0[ 14 ], &pulses0[ 15 ], psRangeDec, pulses1[ 7 ], silk_shell_code_table0 );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/sort.c b/lib/rbcodec/codecs/libopus/silk/sort.c
new file mode 100644
index 0000000000..f9886b45b2
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/sort.c
@@ -0,0 +1,154 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+/* Insertion sort (fast for already almost sorted arrays):   */
+/* Best case:  O(n)   for an already sorted array            */
+/* Worst case: O(n^2) for an inversely sorted array          */
+/*                                                           */
+/* Shell short:    http://en.wikipedia.org/wiki/Shell_sort   */
+
+#include "SigProc_FIX.h"
+
+void silk_insertion_sort_increasing(
+    opus_int32           *a,             /* I/O   Unsorted / Sorted vector               */
+    opus_int             *idx,           /* O     Index vector for the sorted elements   */
+    const opus_int       L,              /* I     Vector length                          */
+    const opus_int       K               /* I     Number of correctly sorted positions   */
+)
+{
+    opus_int32    value;
+    opus_int        i, j;
+
+    /* Safety checks */
+    silk_assert( K >  0 );
+    silk_assert( L >  0 );
+    silk_assert( L >= K );
+
+    /* Write start indices in index vector */
+    for( i = 0; i < K; i++ ) {
+        idx[ i ] = i;
+    }
+
+    /* Sort vector elements by value, increasing order */
+    for( i = 1; i < K; i++ ) {
+        value = a[ i ];
+        for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) {
+            a[ j + 1 ]   = a[ j ];       /* Shift value */
+            idx[ j + 1 ] = idx[ j ];     /* Shift index */
+        }
+        a[ j + 1 ]   = value;   /* Write value */
+        idx[ j + 1 ] = i;       /* Write index */
+    }
+
+    /* If less than L values are asked for, check the remaining values, */
+    /* but only spend CPU to ensure that the K first values are correct */
+    for( i = K; i < L; i++ ) {
+        value = a[ i ];
+        if( value < a[ K - 1 ] ) {
+            for( j = K - 2; ( j >= 0 ) && ( value < a[ j ] ); j-- ) {
+                a[ j + 1 ]   = a[ j ];       /* Shift value */
+                idx[ j + 1 ] = idx[ j ];     /* Shift index */
+            }
+            a[ j + 1 ]   = value;   /* Write value */
+            idx[ j + 1 ] = i;       /* Write index */
+        }
+    }
+}
+
+#ifdef FIXED_POINT
+/* This function is only used by the fixed-point build */
+void silk_insertion_sort_decreasing_int16(
+    opus_int16                  *a,                 /* I/O   Unsorted / Sorted vector                                   */
+    opus_int                    *idx,               /* O     Index vector for the sorted elements                       */
+    const opus_int              L,                  /* I     Vector length                                              */
+    const opus_int              K                   /* I     Number of correctly sorted positions                       */
+)
+{
+    opus_int i, j;
+    opus_int value;
+
+    /* Safety checks */
+    silk_assert( K >  0 );
+    silk_assert( L >  0 );
+    silk_assert( L >= K );
+
+    /* Write start indices in index vector */
+    for( i = 0; i < K; i++ ) {
+        idx[ i ] = i;
+    }
+
+    /* Sort vector elements by value, decreasing order */
+    for( i = 1; i < K; i++ ) {
+        value = a[ i ];
+        for( j = i - 1; ( j >= 0 ) && ( value > a[ j ] ); j-- ) {
+            a[ j + 1 ]   = a[ j ];     /* Shift value */
+            idx[ j + 1 ] = idx[ j ];   /* Shift index */
+        }
+        a[ j + 1 ]   = value;   /* Write value */
+        idx[ j + 1 ] = i;       /* Write index */
+    }
+
+    /* If less than L values are asked for, check the remaining values, */
+    /* but only spend CPU to ensure that the K first values are correct */
+    for( i = K; i < L; i++ ) {
+        value = a[ i ];
+        if( value > a[ K - 1 ] ) {
+            for( j = K - 2; ( j >= 0 ) && ( value > a[ j ] ); j-- ) {
+                a[ j + 1 ]   = a[ j ];     /* Shift value */
+                idx[ j + 1 ] = idx[ j ];   /* Shift index */
+            }
+            a[ j + 1 ]   = value;   /* Write value */
+            idx[ j + 1 ] = i;       /* Write index */
+        }
+    }
+}
+#endif
+
+void silk_insertion_sort_increasing_all_values_int16(
+     opus_int16                 *a,                 /* I/O   Unsorted / Sorted vector                                   */
+     const opus_int             L                   /* I     Vector length                                              */
+)
+{
+    opus_int    value;
+    opus_int    i, j;
+
+    /* Safety checks */
+    silk_assert( L >  0 );
+
+    /* Sort vector elements by value, increasing order */
+    for( i = 1; i < L; i++ ) {
+        value = a[ i ];
+        for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) {
+            a[ j + 1 ] = a[ j ]; /* Shift value */
+        }
+        a[ j + 1 ] = value; /* Write value */
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/stereo_MS_to_LR.c b/lib/rbcodec/codecs/libopus/silk/stereo_MS_to_LR.c
new file mode 100644
index 0000000000..3705b59686
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/stereo_MS_to_LR.c
@@ -0,0 +1,85 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Convert adaptive Mid/Side representation to Left/Right stereo signal */
+void silk_stereo_MS_to_LR(
+    stereo_dec_state            *state,                         /* I/O  State                                       */
+    opus_int16                  x1[],                           /* I/O  Left input signal, becomes mid signal       */
+    opus_int16                  x2[],                           /* I/O  Right input signal, becomes side signal     */
+    const opus_int32            pred_Q13[],                     /* I    Predictors                                  */
+    opus_int                    fs_kHz,                         /* I    Samples rate (kHz)                          */
+    opus_int                    frame_length                    /* I    Number of samples                           */
+)
+{
+    opus_int   n, denom_Q16, delta0_Q13, delta1_Q13;
+    opus_int32 sum, diff, pred0_Q13, pred1_Q13;
+
+    /* Buffering */
+    silk_memcpy( x1, state->sMid,  2 * sizeof( opus_int16 ) );
+    silk_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) );
+    silk_memcpy( state->sMid,  &x1[ frame_length ], 2 * sizeof( opus_int16 ) );
+    silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) );
+
+    /* Interpolate predictors and add prediction to side channel */
+    pred0_Q13  = state->pred_prev_Q13[ 0 ];
+    pred1_Q13  = state->pred_prev_Q13[ 1 ];
+    denom_Q16  = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
+    delta0_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
+    delta1_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
+    for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
+        pred0_Q13 += delta0_Q13;
+        pred1_Q13 += delta1_Q13;
+        sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );       /* Q11 */
+        sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 );         /* Q8  */
+        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 );        /* Q8  */
+        x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+    }
+    pred0_Q13 = pred_Q13[ 0 ];
+    pred1_Q13 = pred_Q13[ 1 ];
+    for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
+        sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );       /* Q11 */
+        sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 );         /* Q8  */
+        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 );        /* Q8  */
+        x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+    }
+    state->pred_prev_Q13[ 0 ] = pred_Q13[ 0 ];
+    state->pred_prev_Q13[ 1 ] = pred_Q13[ 1 ];
+
+    /* Convert to left/right signals */
+    for( n = 0; n < frame_length; n++ ) {
+        sum  = x1[ n + 1 ] + (opus_int32)x2[ n + 1 ];
+        diff = x1[ n + 1 ] - (opus_int32)x2[ n + 1 ];
+        x1[ n + 1 ] = (opus_int16)silk_SAT16( sum );
+        x2[ n + 1 ] = (opus_int16)silk_SAT16( diff );
+    }
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/stereo_decode_pred.c b/lib/rbcodec/codecs/libopus/silk/stereo_decode_pred.c
new file mode 100644
index 0000000000..8614cb59fb
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/stereo_decode_pred.c
@@ -0,0 +1,73 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "main.h"
+
+/* Decode mid/side predictors */
+void silk_stereo_decode_pred(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int32                  pred_Q13[]                      /* O    Predictors                                  */
+)
+{
+    opus_int   n, ix[ 2 ][ 3 ];
+    opus_int32 low_Q13, step_Q13;
+
+    /* Entropy decoding */
+    n = ec_dec_icdf( psRangeDec, silk_stereo_pred_joint_iCDF, 8 );
+    ix[ 0 ][ 2 ] = silk_DIV32_16( n, 5 );
+    ix[ 1 ][ 2 ] = n - 5 * ix[ 0 ][ 2 ];
+    for( n = 0; n < 2; n++ ) {
+        ix[ n ][ 0 ] = ec_dec_icdf( psRangeDec, silk_uniform3_iCDF, 8 );
+        ix[ n ][ 1 ] = ec_dec_icdf( psRangeDec, silk_uniform5_iCDF, 8 );
+    }
+
+    /* Dequantize */
+    for( n = 0; n < 2; n++ ) {
+        ix[ n ][ 0 ] += 3 * ix[ n ][ 2 ];
+        low_Q13 = silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] ];
+        step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] + 1 ] - low_Q13,
+            SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) );
+        pred_Q13[ n ] = silk_SMLABB( low_Q13, step_Q13, 2 * ix[ n ][ 1 ] + 1 );
+    }
+
+    /* Subtract second from first predictor (helps when actually applying these) */
+    pred_Q13[ 0 ] -= pred_Q13[ 1 ];
+}
+
+/* Decode mid-only flag */
+void silk_stereo_decode_mid_only(
+    ec_dec                      *psRangeDec,                    /* I/O  Compressor data structure                   */
+    opus_int                    *decode_only_mid                /* O    Flag that only mid channel has been coded   */
+)
+{
+    /* Decode flag that only mid channel is coded */
+    *decode_only_mid = ec_dec_icdf( psRangeDec, silk_stereo_only_code_mid_iCDF, 8 );
+}
diff --git a/lib/rbcodec/codecs/libopus/silk/structs.h b/lib/rbcodec/codecs/libopus/silk/structs.h
new file mode 100644
index 0000000000..5d37f6605d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/structs.h
@@ -0,0 +1,324 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_STRUCTS_H
+#define SILK_STRUCTS_H
+
+#include "typedef.h"
+#include "SigProc_FIX.h"
+#include "define.h"
+#include "entenc.h"
+#include "entdec.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/************************************/
+/* Noise shaping quantization state */
+/************************************/
+typedef struct {
+    opus_int16                  xq[           2 * MAX_FRAME_LENGTH ]; /* Buffer for quantized output signal                             */
+    opus_int32                  sLTP_shp_Q14[ 2 * MAX_FRAME_LENGTH ];
+    opus_int32                  sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ];
+    opus_int32                  sAR2_Q14[ MAX_SHAPE_LPC_ORDER ];
+    opus_int32                  sLF_AR_shp_Q14;
+    opus_int                    lagPrev;
+    opus_int                    sLTP_buf_idx;
+    opus_int                    sLTP_shp_buf_idx;
+    opus_int32                  rand_seed;
+    opus_int32                  prev_gain_Q16;
+    opus_int                    rewhite_flag;
+} silk_nsq_state;
+
+/********************************/
+/* VAD state                    */
+/********************************/
+typedef struct {
+    opus_int32                  AnaState[ 2 ];                  /* Analysis filterbank state: 0-8 kHz                                   */
+    opus_int32                  AnaState1[ 2 ];                 /* Analysis filterbank state: 0-4 kHz                                   */
+    opus_int32                  AnaState2[ 2 ];                 /* Analysis filterbank state: 0-2 kHz                                   */
+    opus_int32                  XnrgSubfr[ VAD_N_BANDS ];       /* Subframe energies                                                    */
+    opus_int32                  NrgRatioSmth_Q8[ VAD_N_BANDS ]; /* Smoothed energy level in each band                                   */
+    opus_int16                  HPstate;                        /* State of differentiator in the lowest band                           */
+    opus_int32                  NL[ VAD_N_BANDS ];              /* Noise energy level in each band                                      */
+    opus_int32                  inv_NL[ VAD_N_BANDS ];          /* Inverse noise energy level in each band                              */
+    opus_int32                  NoiseLevelBias[ VAD_N_BANDS ];  /* Noise level estimator bias/offset                                    */
+    opus_int32                  counter;                        /* Frame counter used in the initial phase                              */
+} silk_VAD_state;
+
+/* Variable cut-off low-pass filter state */
+typedef struct {
+    opus_int32                   In_LP_State[ 2 ];           /* Low pass filter state */
+    opus_int32                   transition_frame_no;        /* Counter which is mapped to a cut-off frequency */
+    opus_int                     mode;                       /* Operating mode, <0: switch down, >0: switch up; 0: do nothing           */
+} silk_LP_state;
+
+/* Structure containing NLSF codebook */
+typedef struct {
+    const opus_int16             nVectors;
+    const opus_int16             order;
+    const opus_int16             quantStepSize_Q16;
+    const opus_int16             invQuantStepSize_Q6;
+    const opus_uint8             *CB1_NLSF_Q8;
+    const opus_uint8             *CB1_iCDF;
+    const opus_uint8             *pred_Q8;
+    const opus_uint8             *ec_sel;
+    const opus_uint8             *ec_iCDF;
+    const opus_uint8             *ec_Rates_Q5;
+    const opus_int16             *deltaMin_Q15;
+} silk_NLSF_CB_struct;
+
+typedef struct {
+    opus_int16                   pred_prev_Q13[ 2 ];
+    opus_int16                   sMid[ 2 ];
+    opus_int16                   sSide[ 2 ];
+    opus_int32                   mid_side_amp_Q0[ 4 ];
+    opus_int16                   smth_width_Q14;
+    opus_int16                   width_prev_Q14;
+    opus_int16                   silent_side_len;
+    opus_int8                    predIx[ MAX_FRAMES_PER_PACKET ][ 2 ][ 3 ];
+    opus_int8                    mid_only_flags[ MAX_FRAMES_PER_PACKET ];
+} stereo_enc_state;
+
+typedef struct {
+    opus_int16                   pred_prev_Q13[ 2 ];
+    opus_int16                   sMid[ 2 ];
+    opus_int16                   sSide[ 2 ];
+} stereo_dec_state;
+
+typedef struct {
+    opus_int8                    GainsIndices[ MAX_NB_SUBFR ];
+    opus_int8                    LTPIndex[ MAX_NB_SUBFR ];
+    opus_int8                    NLSFIndices[ MAX_LPC_ORDER + 1 ];
+    opus_int16                   lagIndex;
+    opus_int8                    contourIndex;
+    opus_int8                    signalType;
+    opus_int8                    quantOffsetType;
+    opus_int8                    NLSFInterpCoef_Q2;
+    opus_int8                    PERIndex;
+    opus_int8                    LTP_scaleIndex;
+    opus_int8                    Seed;
+} SideInfoIndices;
+
+/********************************/
+/* Encoder state                */
+/********************************/
+typedef struct {
+    opus_int32                   In_HP_State[ 2 ];                  /* High pass filter state                                           */
+    opus_int32                   variable_HP_smth1_Q15;             /* State of first smoother                                          */
+    opus_int32                   variable_HP_smth2_Q15;             /* State of second smoother                                         */
+    silk_LP_state                sLP;                               /* Low pass filter state                                            */
+    silk_VAD_state               sVAD;                              /* Voice activity detector state                                    */
+    silk_nsq_state               sNSQ;                              /* Noise Shape Quantizer State                                      */
+    opus_int16                   prev_NLSFq_Q15[ MAX_LPC_ORDER ];   /* Previously quantized NLSF vector                                 */
+    opus_int                     speech_activity_Q8;                /* Speech activity                                                  */
+    opus_int                     allow_bandwidth_switch;            /* Flag indicating that switching of internal bandwidth is allowed  */
+    opus_int8                    LBRRprevLastGainIndex;
+    opus_int8                    prevSignalType;
+    opus_int                     prevLag;
+    opus_int                     pitch_LPC_win_length;
+    opus_int                     max_pitch_lag;                     /* Highest possible pitch lag (samples)                             */
+    opus_int32                   API_fs_Hz;                         /* API sampling frequency (Hz)                                      */
+    opus_int32                   prev_API_fs_Hz;                    /* Previous API sampling frequency (Hz)                             */
+    opus_int                     maxInternal_fs_Hz;                 /* Maximum internal sampling frequency (Hz)                         */
+    opus_int                     minInternal_fs_Hz;                 /* Minimum internal sampling frequency (Hz)                         */
+    opus_int                     desiredInternal_fs_Hz;             /* Soft request for internal sampling frequency (Hz)                */
+    opus_int                     fs_kHz;                            /* Internal sampling frequency (kHz)                                */
+    opus_int                     nb_subfr;                          /* Number of 5 ms subframes in a frame                              */
+    opus_int                     frame_length;                      /* Frame length (samples)                                           */
+    opus_int                     subfr_length;                      /* Subframe length (samples)                                        */
+    opus_int                     ltp_mem_length;                    /* Length of LTP memory                                             */
+    opus_int                     la_pitch;                          /* Look-ahead for pitch analysis (samples)                          */
+    opus_int                     la_shape;                          /* Look-ahead for noise shape analysis (samples)                    */
+    opus_int                     shapeWinLength;                    /* Window length for noise shape analysis (samples)                 */
+    opus_int32                   TargetRate_bps;                    /* Target bitrate (bps)                                             */
+    opus_int                     PacketSize_ms;                     /* Number of milliseconds to put in each packet                     */
+    opus_int                     PacketLoss_perc;                   /* Packet loss rate measured by farend                              */
+    opus_int32                   frameCounter;
+    opus_int                     Complexity;                        /* Complexity setting                                               */
+    opus_int                     nStatesDelayedDecision;            /* Number of states in delayed decision quantization                */
+    opus_int                     useInterpolatedNLSFs;              /* Flag for using NLSF interpolation                                */
+    opus_int                     shapingLPCOrder;                   /* Filter order for noise shaping filters                           */
+    opus_int                     predictLPCOrder;                   /* Filter order for prediction filters                              */
+    opus_int                     pitchEstimationComplexity;         /* Complexity level for pitch estimator                             */
+    opus_int                     pitchEstimationLPCOrder;           /* Whitening filter order for pitch estimator                       */
+    opus_int32                   pitchEstimationThreshold_Q16;      /* Threshold for pitch estimator                                    */
+    opus_int                     LTPQuantLowComplexity;             /* Flag for low complexity LTP quantization                         */
+    opus_int                     mu_LTP_Q9;                         /* Rate-distortion tradeoff in LTP quantization                     */
+    opus_int                     NLSF_MSVQ_Survivors;               /* Number of survivors in NLSF MSVQ                                 */
+    opus_int                     first_frame_after_reset;           /* Flag for deactivating NLSF interpolation, pitch prediction       */
+    opus_int                     controlled_since_last_payload;     /* Flag for ensuring codec_control only runs once per packet        */
+    opus_int                     warping_Q16;                       /* Warping parameter for warped noise shaping                       */
+    opus_int                     useCBR;                            /* Flag to enable constant bitrate                                  */
+    opus_int                     prefillFlag;                       /* Flag to indicate that only buffers are prefilled, no coding      */
+    const opus_uint8             *pitch_lag_low_bits_iCDF;          /* Pointer to iCDF table for low bits of pitch lag index            */
+    const opus_uint8             *pitch_contour_iCDF;               /* Pointer to iCDF table for pitch contour index                    */
+    const silk_NLSF_CB_struct    *psNLSF_CB;                        /* Pointer to NLSF codebook                                         */
+    opus_int                     input_quality_bands_Q15[ VAD_N_BANDS ];
+    opus_int                     input_tilt_Q15;
+    opus_int                     SNR_dB_Q7;                         /* Quality setting                                                  */
+
+    opus_int8                    VAD_flags[ MAX_FRAMES_PER_PACKET ];
+    opus_int8                    LBRR_flag;
+    opus_int                     LBRR_flags[ MAX_FRAMES_PER_PACKET ];
+
+    SideInfoIndices              indices;
+    opus_int8                    pulses[ MAX_FRAME_LENGTH ];
+
+    /* Input/output buffering */
+    opus_int16                   inputBuf[ MAX_FRAME_LENGTH + 2 ];  /* Buffer containing input signal                                   */
+    opus_int                     inputBufIx;
+    opus_int                     nFramesPerPacket;
+    opus_int                     nFramesEncoded;                    /* Number of frames analyzed in current packet                      */
+
+    opus_int                     nChannelsAPI;
+    opus_int                     nChannelsInternal;
+    opus_int                     channelNb;
+
+    /* Parameters For LTP scaling Control */
+    opus_int                     frames_since_onset;
+
+    /* Specifically for entropy coding */
+    opus_int                     ec_prevSignalType;
+    opus_int16                   ec_prevLagIndex;
+
+    silk_resampler_state_struct resampler_state;
+
+    /* DTX */
+    opus_int                     useDTX;                            /* Flag to enable DTX                                               */
+    opus_int                     inDTX;                             /* Flag to signal DTX period                                        */
+    opus_int                     noSpeechCounter;                   /* Counts concecutive nonactive frames, used by DTX                 */
+
+    /* Inband Low Bitrate Redundancy (LBRR) data */
+    opus_int                     useInBandFEC;                      /* Saves the API setting for query                                  */
+    opus_int                     LBRR_enabled;                      /* Depends on useInBandFRC, bitrate and packet loss rate            */
+    opus_int                     LBRR_GainIncreases;                /* Gains increment for coding LBRR frames                           */
+    SideInfoIndices              indices_LBRR[ MAX_FRAMES_PER_PACKET ];
+    opus_int8                    pulses_LBRR[ MAX_FRAMES_PER_PACKET ][ MAX_FRAME_LENGTH ];
+} silk_encoder_state;
+
+
+/* Struct for Packet Loss Concealment */
+typedef struct {
+    opus_int32                  pitchL_Q8;                          /* Pitch lag to use for voiced concealment                          */
+    opus_int16                  LTPCoef_Q14[ LTP_ORDER ];           /* LTP coeficients to use for voiced concealment                    */
+    opus_int16                  prevLPC_Q12[ MAX_LPC_ORDER ];
+    opus_int                    last_frame_lost;                    /* Was previous frame lost                                          */
+    opus_int32                  rand_seed;                          /* Seed for unvoiced signal generation                              */
+    opus_int16                  randScale_Q14;                      /* Scaling of unvoiced random signal                                */
+    opus_int32                  conc_energy;
+    opus_int                    conc_energy_shift;
+    opus_int16                  prevLTP_scale_Q14;
+    opus_int32                  prevGain_Q16[ 2 ];
+    opus_int                    fs_kHz;
+    opus_int                    nb_subfr;
+    opus_int                    subfr_length;
+} silk_PLC_struct;
+
+/* Struct for CNG */
+typedef struct {
+    opus_int32                  CNG_exc_buf_Q14[ MAX_FRAME_LENGTH ];
+    opus_int16                  CNG_smth_NLSF_Q15[ MAX_LPC_ORDER ];
+    opus_int32                  CNG_synth_state[ MAX_LPC_ORDER ];
+    opus_int32                  CNG_smth_Gain_Q16;
+    opus_int32                  rand_seed;
+    opus_int                    fs_kHz;
+} silk_CNG_struct;
+
+/********************************/
+/* Decoder state                */
+/********************************/
+typedef struct {
+    opus_int32                  prev_gain_Q16;
+    opus_int32                  exc_Q14[ MAX_FRAME_LENGTH ];
+    opus_int32                  sLPC_Q14_buf[ MAX_LPC_ORDER ];
+    opus_int16                  outBuf[ MAX_FRAME_LENGTH + 2 * MAX_SUB_FRAME_LENGTH ];  /* Buffer for output signal                     */
+    opus_int                    lagPrev;                            /* Previous Lag                                                     */
+    opus_int8                   LastGainIndex;                      /* Previous gain index                                              */
+    opus_int                    fs_kHz;                             /* Sampling frequency in kHz                                        */
+    opus_int32                  fs_API_hz;                          /* API sample frequency (Hz)                                        */
+    opus_int                    nb_subfr;                           /* Number of 5 ms subframes in a frame                              */
+    opus_int                    frame_length;                       /* Frame length (samples)                                           */
+    opus_int                    subfr_length;                       /* Subframe length (samples)                                        */
+    opus_int                    ltp_mem_length;                     /* Length of LTP memory                                             */
+    opus_int                    LPC_order;                          /* LPC order                                                        */
+    opus_int16                  prevNLSF_Q15[ MAX_LPC_ORDER ];      /* Used to interpolate LSFs                                         */
+    opus_int                    first_frame_after_reset;            /* Flag for deactivating NLSF interpolation                         */
+    const opus_uint8            *pitch_lag_low_bits_iCDF;           /* Pointer to iCDF table for low bits of pitch lag index            */
+    const opus_uint8            *pitch_contour_iCDF;                /* Pointer to iCDF table for pitch contour index                    */
+
+    /* For buffering payload in case of more frames per packet */
+    opus_int                    nFramesDecoded;
+    opus_int                    nFramesPerPacket;
+
+    /* Specifically for entropy coding */
+    opus_int                    ec_prevSignalType;
+    opus_int16                  ec_prevLagIndex;
+
+    opus_int                    VAD_flags[ MAX_FRAMES_PER_PACKET ];
+    opus_int                    LBRR_flag;
+    opus_int                    LBRR_flags[ MAX_FRAMES_PER_PACKET ];
+
+    silk_resampler_state_struct resampler_state;
+
+    const silk_NLSF_CB_struct   *psNLSF_CB;                         /* Pointer to NLSF codebook                                         */
+
+    /* Quantization indices */
+    SideInfoIndices             indices;
+
+    /* CNG state */
+    silk_CNG_struct             sCNG;
+
+    /* Stuff used for PLC */
+    opus_int                    lossCnt;
+    opus_int                    prevSignalType;
+
+    silk_PLC_struct sPLC;
+
+} silk_decoder_state;
+
+/************************/
+/* Decoder control      */
+/************************/
+typedef struct {
+    /* Prediction and coding parameters */
+    opus_int                    pitchL[ MAX_NB_SUBFR ];
+    opus_int32                  Gains_Q16[ MAX_NB_SUBFR ];
+    /* Holds interpolated and final coefficients, 4-byte aligned */
+    silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ];
+    opus_int16                  LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ];
+    opus_int                    LTP_scale_Q14;
+} silk_decoder_control;
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/sum_sqr_shift.c b/lib/rbcodec/codecs/libopus/silk/sum_sqr_shift.c
new file mode 100644
index 0000000000..53f70fd8de
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/sum_sqr_shift.c
@@ -0,0 +1,85 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "SigProc_FIX.h"
+
+/* Compute number of bits to right shift the sum of squares of a vector */
+/* of int16s to make it fit in an int32                                 */
+void silk_sum_sqr_shift(
+    opus_int32                  *energy,            /* O   Energy of x, after shifting to the right                     */
+    opus_int                    *shift,             /* O   Number of bits right shift applied to energy                 */
+    const opus_int16            *x,                 /* I   Input vector                                                 */
+    opus_int                    len                 /* I   Length of input vector                                       */
+)
+{
+    opus_int   i, shft;
+    opus_int32 nrg_tmp, nrg;
+
+    nrg  = 0;
+    shft = 0;
+    len--;
+    for( i = 0; i < len; i += 2 ) {
+        nrg = silk_SMLABB_ovflw( nrg, x[ i ], x[ i ] );
+        nrg = silk_SMLABB_ovflw( nrg, x[ i + 1 ], x[ i + 1 ] );
+        if( nrg < 0 ) {
+            /* Scale down */
+            nrg = (opus_int32)silk_RSHIFT_uint( (opus_uint32)nrg, 2 );
+            shft = 2;
+            break;
+        }
+    }
+    for( ; i < len; i += 2 ) {
+        nrg_tmp = silk_SMULBB( x[ i ], x[ i ] );
+        nrg_tmp = silk_SMLABB_ovflw( nrg_tmp, x[ i + 1 ], x[ i + 1 ] );
+        nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, (opus_uint32)nrg_tmp, shft );
+        if( nrg < 0 ) {
+            /* Scale down */
+            nrg = (opus_int32)silk_RSHIFT_uint( (opus_uint32)nrg, 2 );
+            shft += 2;
+        }
+    }
+    if( i == len ) {
+        /* One sample left to process */
+        nrg_tmp = silk_SMULBB( x[ i ], x[ i ] );
+        nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft );
+    }
+
+    /* Make sure to have at least one extra leading zero (two leading zeros in total) */
+    if( nrg & 0xC0000000 ) {
+        nrg = silk_RSHIFT_uint( (opus_uint32)nrg, 2 );
+        shft += 2;
+    }
+
+    /* Output arguments */
+    *shift  = shft;
+    *energy = nrg;
+}
+
diff --git a/lib/rbcodec/codecs/libopus/silk/table_LSF_cos.c b/lib/rbcodec/codecs/libopus/silk/table_LSF_cos.c
new file mode 100644
index 0000000000..b58e3f4728
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/table_LSF_cos.c
@@ -0,0 +1,70 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+/* Cosine approximation table for LSF conversion */
+/* Q12 values (even) */
+const opus_int16 silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ] = {
+            8192,             8190,             8182,             8170,
+            8152,             8130,             8104,             8072,
+            8034,             7994,             7946,             7896,
+            7840,             7778,             7714,             7644,
+            7568,             7490,             7406,             7318,
+            7226,             7128,             7026,             6922,
+            6812,             6698,             6580,             6458,
+            6332,             6204,             6070,             5934,
+            5792,             5648,             5502,             5352,
+            5198,             5040,             4880,             4718,
+            4552,             4382,             4212,             4038,
+            3862,             3684,             3502,             3320,
+            3136,             2948,             2760,             2570,
+            2378,             2186,             1990,             1794,
+            1598,             1400,             1202,             1002,
+             802,              602,              402,              202,
+               0,             -202,             -402,             -602,
+            -802,            -1002,            -1202,            -1400,
+           -1598,            -1794,            -1990,            -2186,
+           -2378,            -2570,            -2760,            -2948,
+           -3136,            -3320,            -3502,            -3684,
+           -3862,            -4038,            -4212,            -4382,
+           -4552,            -4718,            -4880,            -5040,
+           -5198,            -5352,            -5502,            -5648,
+           -5792,            -5934,            -6070,            -6204,
+           -6332,            -6458,            -6580,            -6698,
+           -6812,            -6922,            -7026,            -7128,
+           -7226,            -7318,            -7406,            -7490,
+           -7568,            -7644,            -7714,            -7778,
+           -7840,            -7896,            -7946,            -7994,
+           -8034,            -8072,            -8104,            -8130,
+           -8152,            -8170,            -8182,            -8190,
+           -8192
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/tables.h b/lib/rbcodec/codecs/libopus/silk/tables.h
new file mode 100644
index 0000000000..072b7929d5
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables.h
@@ -0,0 +1,120 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_TABLES_H
+#define SILK_TABLES_H
+
+#include "define.h"
+#include "structs.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/* Entropy coding tables (with size in bytes indicated) */
+extern const opus_uint8  silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ];                                 /* 24 */
+extern const opus_uint8  silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ];   /* 41 */
+
+extern const opus_uint8  silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ];/* 32 */
+extern const opus_uint8  silk_pitch_delta_iCDF[ 21 ];                                               /*  21 */
+extern const opus_uint8  silk_pitch_contour_iCDF[ 34 ];                                             /*  34 */
+extern const opus_uint8  silk_pitch_contour_NB_iCDF[ 11 ];                                          /*  11 */
+extern const opus_uint8  silk_pitch_contour_10_ms_iCDF[ 12 ];                                       /*  12 */
+extern const opus_uint8  silk_pitch_contour_10_ms_NB_iCDF[ 3 ];                                     /*   3 */
+
+extern const opus_uint8  silk_pulses_per_block_iCDF[ N_RATE_LEVELS ][ MAX_PULSES + 2 ];             /* 180 */
+extern const opus_uint8  silk_pulses_per_block_BITS_Q5[ N_RATE_LEVELS - 1 ][ MAX_PULSES + 2 ];      /* 162 */
+
+extern const opus_uint8  silk_rate_levels_iCDF[ 2 ][ N_RATE_LEVELS - 1 ];                           /*  18 */
+extern const opus_uint8  silk_rate_levels_BITS_Q5[ 2 ][ N_RATE_LEVELS - 1 ];                        /*  18 */
+
+extern const opus_uint8  silk_max_pulses_table[ 4 ];                                                /*   4 */
+
+extern const opus_uint8  silk_shell_code_table0[ 152 ];                                             /* 152 */
+extern const opus_uint8  silk_shell_code_table1[ 152 ];                                             /* 152 */
+extern const opus_uint8  silk_shell_code_table2[ 152 ];                                             /* 152 */
+extern const opus_uint8  silk_shell_code_table3[ 152 ];                                             /* 152 */
+extern const opus_uint8  silk_shell_code_table_offsets[ MAX_PULSES + 1 ];                           /*  17 */
+
+extern const opus_uint8  silk_lsb_iCDF[ 2 ];                                                        /*   2 */
+
+extern const opus_uint8  silk_sign_iCDF[ 42 ];                                                      /*  42 */
+
+extern const opus_uint8  silk_uniform3_iCDF[ 3 ];                                                   /*   3 */
+extern const opus_uint8  silk_uniform4_iCDF[ 4 ];                                                   /*   4 */
+extern const opus_uint8  silk_uniform5_iCDF[ 5 ];                                                   /*   5 */
+extern const opus_uint8  silk_uniform6_iCDF[ 6 ];                                                   /*   6 */
+extern const opus_uint8  silk_uniform8_iCDF[ 8 ];                                                   /*   8 */
+
+extern const opus_uint8  silk_NLSF_EXT_iCDF[ 7 ];                                                   /*   7 */
+
+extern const opus_uint8  silk_LTP_per_index_iCDF[ 3 ];                                              /*   3 */
+extern const opus_uint8  * const silk_LTP_gain_iCDF_ptrs[ NB_LTP_CBKS ];                            /*   3 */
+extern const opus_uint8  * const silk_LTP_gain_BITS_Q5_ptrs[ NB_LTP_CBKS ];                         /*   3 */
+extern const opus_int16  silk_LTP_gain_middle_avg_RD_Q14;
+extern const opus_int8   * const silk_LTP_vq_ptrs_Q7[ NB_LTP_CBKS ];                                /* 168 */
+extern const opus_int8   silk_LTP_vq_sizes[ NB_LTP_CBKS ];                                          /*   3 */
+
+extern const opus_uint8  silk_LTPscale_iCDF[ 3 ];                                                   /*   4 */
+extern const opus_int16  silk_LTPScales_table_Q14[ 3 ];                                             /*   6 */
+
+extern const opus_uint8  silk_type_offset_VAD_iCDF[ 4 ];                                            /*   4 */
+extern const opus_uint8  silk_type_offset_no_VAD_iCDF[ 2 ];                                         /*   2 */
+
+extern const opus_int16  silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ];                       /*  32 */
+extern const opus_uint8  silk_stereo_pred_joint_iCDF[ 25 ];                                         /*  25 */
+extern const opus_uint8  silk_stereo_only_code_mid_iCDF[ 2 ];                                       /*   2 */
+
+extern const opus_uint8  * const silk_LBRR_flags_iCDF_ptr[ 2 ];                                     /*  10 */
+
+extern const opus_uint8  silk_NLSF_interpolation_factor_iCDF[ 5 ];                                  /*   5 */
+
+extern const silk_NLSF_CB_struct silk_NLSF_CB_WB;                                                   /* 1040 */
+extern const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB;                                                /* 728 */
+
+/* Piece-wise linear mapping from bitrate in kbps to coding quality in dB SNR */
+extern const opus_int32  silk_TargetRate_table_NB[  TARGET_RATE_TAB_SZ ];                           /*  32 */
+extern const opus_int32  silk_TargetRate_table_MB[  TARGET_RATE_TAB_SZ ];                           /*  32 */
+extern const opus_int32  silk_TargetRate_table_WB[  TARGET_RATE_TAB_SZ ];                           /*  32 */
+extern const opus_int16  silk_SNR_table_Q1[         TARGET_RATE_TAB_SZ ];                           /*  32 */
+
+/* Quantization offsets */
+extern const opus_int16  silk_Quantization_Offsets_Q10[ 2 ][ 2 ];                                   /*   8 */
+
+/* Interpolation points for filter coefficients used in the bandwidth transition smoother */
+extern const opus_int32  silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ];           /*  60 */
+extern const opus_int32  silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ];           /*  60 */
+
+/* Rom table with cosine values */
+extern const opus_int16  silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ];                          /* 258 */
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_LTP.c b/lib/rbcodec/codecs/libopus/silk/tables_LTP.c
new file mode 100644
index 0000000000..6deb9bc556
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_LTP.c
@@ -0,0 +1,272 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+const opus_uint8 silk_LTP_per_index_iCDF[3] = {
+       179,     99,      0
+};
+
+static const opus_uint8 silk_LTP_gain_iCDF_0[8] = {
+        71,     56,     43,     30,     21,     12,      6,      0
+};
+
+static const opus_uint8 silk_LTP_gain_iCDF_1[16] = {
+       199,    165,    144,    124,    109,     96,     84,     71,
+        61,     51,     42,     32,     23,     15,      8,      0
+};
+
+static const opus_uint8 silk_LTP_gain_iCDF_2[32] = {
+       241,    225,    211,    199,    187,    175,    164,    153,
+       142,    132,    123,    114,    105,     96,     88,     80,
+        72,     64,     57,     50,     44,     38,     33,     29,
+        24,     20,     16,     12,      9,      5,      2,      0
+};
+
+const opus_int16 silk_LTP_gain_middle_avg_RD_Q14 = 12304;
+
+static const opus_uint8 silk_LTP_gain_BITS_Q5_0[8] = {
+        15,    131,    138,    138,    155,    155,    173,    173
+};
+
+static const opus_uint8 silk_LTP_gain_BITS_Q5_1[16] = {
+        69,     93,    115,    118,    131,    138,    141,    138,
+       150,    150,    155,    150,    155,    160,    166,    160
+};
+
+static const opus_uint8 silk_LTP_gain_BITS_Q5_2[32] = {
+       131,    128,    134,    141,    141,    141,    145,    145,
+       145,    150,    155,    155,    155,    155,    160,    160,
+       160,    160,    166,    166,    173,    173,    182,    192,
+       182,    192,    192,    192,    205,    192,    205,    224
+};
+
+const opus_uint8 * const silk_LTP_gain_iCDF_ptrs[NB_LTP_CBKS] = {
+    silk_LTP_gain_iCDF_0,
+    silk_LTP_gain_iCDF_1,
+    silk_LTP_gain_iCDF_2
+};
+
+const opus_uint8 * const silk_LTP_gain_BITS_Q5_ptrs[NB_LTP_CBKS] = {
+    silk_LTP_gain_BITS_Q5_0,
+    silk_LTP_gain_BITS_Q5_1,
+    silk_LTP_gain_BITS_Q5_2
+};
+
+static const opus_int8 silk_LTP_gain_vq_0[8][5] =
+{
+{
+         4,      6,     24,      7,      5
+},
+{
+         0,      0,      2,      0,      0
+},
+{
+        12,     28,     41,     13,     -4
+},
+{
+        -9,     15,     42,     25,     14
+},
+{
+         1,     -2,     62,     41,     -9
+},
+{
+       -10,     37,     65,     -4,      3
+},
+{
+        -6,      4,     66,      7,     -8
+},
+{
+        16,     14,     38,     -3,     33
+}
+};
+
+static const opus_int8 silk_LTP_gain_vq_1[16][5] =
+{
+{
+        13,     22,     39,     23,     12
+},
+{
+        -1,     36,     64,     27,     -6
+},
+{
+        -7,     10,     55,     43,     17
+},
+{
+         1,      1,      8,      1,      1
+},
+{
+         6,    -11,     74,     53,     -9
+},
+{
+       -12,     55,     76,    -12,      8
+},
+{
+        -3,      3,     93,     27,     -4
+},
+{
+        26,     39,     59,      3,     -8
+},
+{
+         2,      0,     77,     11,      9
+},
+{
+        -8,     22,     44,     -6,      7
+},
+{
+        40,      9,     26,      3,      9
+},
+{
+        -7,     20,    101,     -7,      4
+},
+{
+         3,     -8,     42,     26,      0
+},
+{
+       -15,     33,     68,      2,     23
+},
+{
+        -2,     55,     46,     -2,     15
+},
+{
+         3,     -1,     21,     16,     41
+}
+};
+
+static const opus_int8 silk_LTP_gain_vq_2[32][5] =
+{
+{
+        -6,     27,     61,     39,      5
+},
+{
+       -11,     42,     88,      4,      1
+},
+{
+        -2,     60,     65,      6,     -4
+},
+{
+        -1,     -5,     73,     56,      1
+},
+{
+        -9,     19,     94,     29,     -9
+},
+{
+         0,     12,     99,      6,      4
+},
+{
+         8,    -19,    102,     46,    -13
+},
+{
+         3,      2,     13,      3,      2
+},
+{
+         9,    -21,     84,     72,    -18
+},
+{
+       -11,     46,    104,    -22,      8
+},
+{
+        18,     38,     48,     23,      0
+},
+{
+       -16,     70,     83,    -21,     11
+},
+{
+         5,    -11,    117,     22,     -8
+},
+{
+        -6,     23,    117,    -12,      3
+},
+{
+         3,     -8,     95,     28,      4
+},
+{
+       -10,     15,     77,     60,    -15
+},
+{
+        -1,      4,    124,      2,     -4
+},
+{
+         3,     38,     84,     24,    -25
+},
+{
+         2,     13,     42,     13,     31
+},
+{
+        21,     -4,     56,     46,     -1
+},
+{
+        -1,     35,     79,    -13,     19
+},
+{
+        -7,     65,     88,     -9,    -14
+},
+{
+        20,      4,     81,     49,    -29
+},
+{
+        20,      0,     75,      3,    -17
+},
+{
+         5,     -9,     44,     92,     -8
+},
+{
+         1,     -3,     22,     69,     31
+},
+{
+        -6,     95,     41,    -12,      5
+},
+{
+        39,     67,     16,     -4,      1
+},
+{
+         0,     -6,    120,     55,    -36
+},
+{
+       -13,     44,    122,      4,    -24
+},
+{
+        81,      5,     11,      3,      7
+},
+{
+         2,      0,      9,     10,     88
+}
+};
+
+const opus_int8 * const silk_LTP_vq_ptrs_Q7[NB_LTP_CBKS] = {
+    (opus_int8 *)&silk_LTP_gain_vq_0[0][0],
+    (opus_int8 *)&silk_LTP_gain_vq_1[0][0],
+    (opus_int8 *)&silk_LTP_gain_vq_2[0][0]
+};
+
+const opus_int8 silk_LTP_vq_sizes[NB_LTP_CBKS] = {
+    8, 16, 32
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_NB_MB.c b/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_NB_MB.c
new file mode 100644
index 0000000000..201a89d110
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_NB_MB.c
@@ -0,0 +1,159 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+static const opus_uint8 silk_NLSF_CB1_NB_MB_Q8[ 320 ] = {
+        12,     35,     60,     83,    108,    132,    157,    180,
+       206,    228,     15,     32,     55,     77,    101,    125,
+       151,    175,    201,    225,     19,     42,     66,     89,
+       114,    137,    162,    184,    209,    230,     12,     25,
+        50,     72,     97,    120,    147,    172,    200,    223,
+        26,     44,     69,     90,    114,    135,    159,    180,
+       205,    225,     13,     22,     53,     80,    106,    130,
+       156,    180,    205,    228,     15,     25,     44,     64,
+        90,    115,    142,    168,    196,    222,     19,     24,
+        62,     82,    100,    120,    145,    168,    190,    214,
+        22,     31,     50,     79,    103,    120,    151,    170,
+       203,    227,     21,     29,     45,     65,    106,    124,
+       150,    171,    196,    224,     30,     49,     75,     97,
+       121,    142,    165,    186,    209,    229,     19,     25,
+        52,     70,     93,    116,    143,    166,    192,    219,
+        26,     34,     62,     75,     97,    118,    145,    167,
+       194,    217,     25,     33,     56,     70,     91,    113,
+       143,    165,    196,    223,     21,     34,     51,     72,
+        97,    117,    145,    171,    196,    222,     20,     29,
+        50,     67,     90,    117,    144,    168,    197,    221,
+        22,     31,     48,     66,     95,    117,    146,    168,
+       196,    222,     24,     33,     51,     77,    116,    134,
+       158,    180,    200,    224,     21,     28,     70,     87,
+       106,    124,    149,    170,    194,    217,     26,     33,
+        53,     64,     83,    117,    152,    173,    204,    225,
+        27,     34,     65,     95,    108,    129,    155,    174,
+       210,    225,     20,     26,     72,     99,    113,    131,
+       154,    176,    200,    219,     34,     43,     61,     78,
+        93,    114,    155,    177,    205,    229,     23,     29,
+        54,     97,    124,    138,    163,    179,    209,    229,
+        30,     38,     56,     89,    118,    129,    158,    178,
+       200,    231,     21,     29,     49,     63,     85,    111,
+       142,    163,    193,    222,     27,     48,     77,    103,
+       133,    158,    179,    196,    215,    232,     29,     47,
+        74,     99,    124,    151,    176,    198,    220,    237,
+        33,     42,     61,     76,     93,    121,    155,    174,
+       207,    225,     29,     53,     87,    112,    136,    154,
+       170,    188,    208,    227,     24,     30,     52,     84,
+       131,    150,    166,    186,    203,    229,     37,     48,
+        64,     84,    104,    118,    156,    177,    201,    230
+};
+
+static const opus_uint8 silk_NLSF_CB1_iCDF_NB_MB[ 64 ] = {
+       212,    178,    148,    129,    108,     96,     85,     82,
+        79,     77,     61,     59,     57,     56,     51,     49,
+        48,     45,     42,     41,     40,     38,     36,     34,
+        31,     30,     21,     12,     10,      3,      1,      0,
+       255,    245,    244,    236,    233,    225,    217,    203,
+       190,    176,    175,    161,    149,    136,    125,    114,
+       102,     91,     81,     71,     60,     52,     43,     35,
+        28,     20,     19,     18,     12,     11,      5,      0
+};
+
+static const opus_uint8 silk_NLSF_CB2_SELECT_NB_MB[ 160 ] = {
+        16,      0,      0,      0,      0,     99,     66,     36,
+        36,     34,     36,     34,     34,     34,     34,     83,
+        69,     36,     52,     34,    116,    102,     70,     68,
+        68,    176,    102,     68,     68,     34,     65,     85,
+        68,     84,     36,    116,    141,    152,    139,    170,
+       132,    187,    184,    216,    137,    132,    249,    168,
+       185,    139,    104,    102,    100,     68,     68,    178,
+       218,    185,    185,    170,    244,    216,    187,    187,
+       170,    244,    187,    187,    219,    138,    103,    155,
+       184,    185,    137,    116,    183,    155,    152,    136,
+       132,    217,    184,    184,    170,    164,    217,    171,
+       155,    139,    244,    169,    184,    185,    170,    164,
+       216,    223,    218,    138,    214,    143,    188,    218,
+       168,    244,    141,    136,    155,    170,    168,    138,
+       220,    219,    139,    164,    219,    202,    216,    137,
+       168,    186,    246,    185,    139,    116,    185,    219,
+       185,    138,    100,    100,    134,    100,    102,     34,
+        68,     68,    100,     68,    168,    203,    221,    218,
+       168,    167,    154,    136,    104,     70,    164,    246,
+       171,    137,    139,    137,    155,    218,    219,    139
+};
+
+static const opus_uint8 silk_NLSF_CB2_iCDF_NB_MB[ 72 ] = {
+       255,    254,    253,    238,     14,      3,      2,      1,
+         0,    255,    254,    252,    218,     35,      3,      2,
+         1,      0,    255,    254,    250,    208,     59,      4,
+         2,      1,      0,    255,    254,    246,    194,     71,
+        10,      2,      1,      0,    255,    252,    236,    183,
+        82,      8,      2,      1,      0,    255,    252,    235,
+       180,     90,     17,      2,      1,      0,    255,    248,
+       224,    171,     97,     30,      4,      1,      0,    255,
+       254,    236,    173,     95,     37,      7,      1,      0
+};
+
+static const opus_uint8 silk_NLSF_CB2_BITS_NB_MB_Q5[ 72 ] = {
+       255,    255,    255,    131,      6,    145,    255,    255,
+       255,    255,    255,    236,     93,     15,     96,    255,
+       255,    255,    255,    255,    194,     83,     25,     71,
+       221,    255,    255,    255,    255,    162,     73,     34,
+        66,    162,    255,    255,    255,    210,    126,     73,
+        43,     57,    173,    255,    255,    255,    201,    125,
+        71,     48,     58,    130,    255,    255,    255,    166,
+       110,     73,     57,     62,    104,    210,    255,    255,
+       251,    123,     65,     55,     68,    100,    171,    255
+};
+
+static const opus_uint8 silk_NLSF_PRED_NB_MB_Q8[ 18 ] = {
+       179,    138,    140,    148,    151,    149,    153,    151,
+       163,    116,     67,     82,     59,     92,     72,    100,
+        89,     92
+};
+
+static const opus_int16 silk_NLSF_DELTA_MIN_NB_MB_Q15[ 11 ] = {
+       250,      3,      6,      3,      3,      3,      4,      3,
+         3,      3,    461
+};
+
+const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB =
+{
+    32,
+    10,
+    SILK_FIX_CONST( 0.18, 16 ),
+    SILK_FIX_CONST( 1.0 / 0.18, 6 ),
+    silk_NLSF_CB1_NB_MB_Q8,
+    silk_NLSF_CB1_iCDF_NB_MB,
+    silk_NLSF_PRED_NB_MB_Q8,
+    silk_NLSF_CB2_SELECT_NB_MB,
+    silk_NLSF_CB2_iCDF_NB_MB,
+    silk_NLSF_CB2_BITS_NB_MB_Q5,
+    silk_NLSF_DELTA_MIN_NB_MB_Q15,
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_WB.c b/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_WB.c
new file mode 100644
index 0000000000..0d9286f39d
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_NLSF_CB_WB.c
@@ -0,0 +1,198 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+static const opus_uint8 silk_NLSF_CB1_WB_Q8[ 512 ] = {
+         7,     23,     38,     54,     69,     85,    100,    116,
+       131,    147,    162,    178,    193,    208,    223,    239,
+        13,     25,     41,     55,     69,     83,     98,    112,
+       127,    142,    157,    171,    187,    203,    220,    236,
+        15,     21,     34,     51,     61,     78,     92,    106,
+       126,    136,    152,    167,    185,    205,    225,    240,
+        10,     21,     36,     50,     63,     79,     95,    110,
+       126,    141,    157,    173,    189,    205,    221,    237,
+        17,     20,     37,     51,     59,     78,     89,    107,
+       123,    134,    150,    164,    184,    205,    224,    240,
+        10,     15,     32,     51,     67,     81,     96,    112,
+       129,    142,    158,    173,    189,    204,    220,    236,
+         8,     21,     37,     51,     65,     79,     98,    113,
+       126,    138,    155,    168,    179,    192,    209,    218,
+        12,     15,     34,     55,     63,     78,     87,    108,
+       118,    131,    148,    167,    185,    203,    219,    236,
+        16,     19,     32,     36,     56,     79,     91,    108,
+       118,    136,    154,    171,    186,    204,    220,    237,
+        11,     28,     43,     58,     74,     89,    105,    120,
+       135,    150,    165,    180,    196,    211,    226,    241,
+         6,     16,     33,     46,     60,     75,     92,    107,
+       123,    137,    156,    169,    185,    199,    214,    225,
+        11,     19,     30,     44,     57,     74,     89,    105,
+       121,    135,    152,    169,    186,    202,    218,    234,
+        12,     19,     29,     46,     57,     71,     88,    100,
+       120,    132,    148,    165,    182,    199,    216,    233,
+        17,     23,     35,     46,     56,     77,     92,    106,
+       123,    134,    152,    167,    185,    204,    222,    237,
+        14,     17,     45,     53,     63,     75,     89,    107,
+       115,    132,    151,    171,    188,    206,    221,    240,
+         9,     16,     29,     40,     56,     71,     88,    103,
+       119,    137,    154,    171,    189,    205,    222,    237,
+        16,     19,     36,     48,     57,     76,     87,    105,
+       118,    132,    150,    167,    185,    202,    218,    236,
+        12,     17,     29,     54,     71,     81,     94,    104,
+       126,    136,    149,    164,    182,    201,    221,    237,
+        15,     28,     47,     62,     79,     97,    115,    129,
+       142,    155,    168,    180,    194,    208,    223,    238,
+         8,     14,     30,     45,     62,     78,     94,    111,
+       127,    143,    159,    175,    192,    207,    223,    239,
+        17,     30,     49,     62,     79,     92,    107,    119,
+       132,    145,    160,    174,    190,    204,    220,    235,
+        14,     19,     36,     45,     61,     76,     91,    108,
+       121,    138,    154,    172,    189,    205,    222,    238,
+        12,     18,     31,     45,     60,     76,     91,    107,
+       123,    138,    154,    171,    187,    204,    221,    236,
+        13,     17,     31,     43,     53,     70,     83,    103,
+       114,    131,    149,    167,    185,    203,    220,    237,
+        17,     22,     35,     42,     58,     78,     93,    110,
+       125,    139,    155,    170,    188,    206,    224,    240,
+         8,     15,     34,     50,     67,     83,     99,    115,
+       131,    146,    162,    178,    193,    209,    224,    239,
+        13,     16,     41,     66,     73,     86,     95,    111,
+       128,    137,    150,    163,    183,    206,    225,    241,
+        17,     25,     37,     52,     63,     75,     92,    102,
+       119,    132,    144,    160,    175,    191,    212,    231,
+        19,     31,     49,     65,     83,    100,    117,    133,
+       147,    161,    174,    187,    200,    213,    227,    242,
+        18,     31,     52,     68,     88,    103,    117,    126,
+       138,    149,    163,    177,    192,    207,    223,    239,
+        16,     29,     47,     61,     76,     90,    106,    119,
+       133,    147,    161,    176,    193,    209,    224,    240,
+        15,     21,     35,     50,     61,     73,     86,     97,
+       110,    119,    129,    141,    175,    198,    218,    237
+};
+
+static const opus_uint8 silk_NLSF_CB1_iCDF_WB[ 64 ] = {
+       225,    204,    201,    184,    183,    175,    158,    154,
+       153,    135,    119,    115,    113,    110,    109,     99,
+        98,     95,     79,     68,     52,     50,     48,     45,
+        43,     32,     31,     27,     18,     10,      3,      0,
+       255,    251,    235,    230,    212,    201,    196,    182,
+       167,    166,    163,    151,    138,    124,    110,    104,
+        90,     78,     76,     70,     69,     57,     45,     34,
+        24,     21,     11,      6,      5,      4,      3,      0
+};
+
+static const opus_uint8 silk_NLSF_CB2_SELECT_WB[ 256 ] = {
+         0,      0,      0,      0,      0,      0,      0,      1,
+       100,    102,    102,     68,     68,     36,     34,     96,
+       164,    107,    158,    185,    180,    185,    139,    102,
+        64,     66,     36,     34,     34,      0,      1,     32,
+       208,    139,    141,    191,    152,    185,    155,    104,
+        96,    171,    104,    166,    102,    102,    102,    132,
+         1,      0,      0,      0,      0,     16,     16,      0,
+        80,    109,     78,    107,    185,    139,    103,    101,
+       208,    212,    141,    139,    173,    153,    123,    103,
+        36,      0,      0,      0,      0,      0,      0,      1,
+        48,      0,      0,      0,      0,      0,      0,     32,
+        68,    135,    123,    119,    119,    103,     69,     98,
+        68,    103,    120,    118,    118,    102,     71,     98,
+       134,    136,    157,    184,    182,    153,    139,    134,
+       208,    168,    248,     75,    189,    143,    121,    107,
+        32,     49,     34,     34,     34,      0,     17,      2,
+       210,    235,    139,    123,    185,    137,    105,    134,
+        98,    135,    104,    182,    100,    183,    171,    134,
+       100,     70,     68,     70,     66,     66,     34,    131,
+        64,    166,    102,     68,     36,      2,      1,      0,
+       134,    166,    102,     68,     34,     34,     66,    132,
+       212,    246,    158,    139,    107,    107,     87,    102,
+       100,    219,    125,    122,    137,    118,    103,    132,
+       114,    135,    137,    105,    171,    106,     50,     34,
+       164,    214,    141,    143,    185,    151,    121,    103,
+       192,     34,      0,      0,      0,      0,      0,      1,
+       208,    109,     74,    187,    134,    249,    159,    137,
+       102,    110,    154,    118,     87,    101,    119,    101,
+         0,      2,      0,     36,     36,     66,     68,     35,
+        96,    164,    102,    100,     36,      0,      2,     33,
+       167,    138,    174,    102,    100,     84,      2,      2,
+       100,    107,    120,    119,     36,    197,     24,      0
+};
+
+static const opus_uint8 silk_NLSF_CB2_iCDF_WB[ 72 ] = {
+       255,    254,    253,    244,     12,      3,      2,      1,
+         0,    255,    254,    252,    224,     38,      3,      2,
+         1,      0,    255,    254,    251,    209,     57,      4,
+         2,      1,      0,    255,    254,    244,    195,     69,
+         4,      2,      1,      0,    255,    251,    232,    184,
+        84,      7,      2,      1,      0,    255,    254,    240,
+       186,     86,     14,      2,      1,      0,    255,    254,
+       239,    178,     91,     30,      5,      1,      0,    255,
+       248,    227,    177,    100,     19,      2,      1,      0
+};
+
+static const opus_uint8 silk_NLSF_CB2_BITS_WB_Q5[ 72 ] = {
+       255,    255,    255,    156,      4,    154,    255,    255,
+       255,    255,    255,    227,    102,     15,     92,    255,
+       255,    255,    255,    255,    213,     83,     24,     72,
+       236,    255,    255,    255,    255,    150,     76,     33,
+        63,    214,    255,    255,    255,    190,    121,     77,
+        43,     55,    185,    255,    255,    255,    245,    137,
+        71,     43,     59,    139,    255,    255,    255,    255,
+       131,     66,     50,     66,    107,    194,    255,    255,
+       166,    116,     76,     55,     53,    125,    255,    255
+};
+
+static const opus_uint8 silk_NLSF_PRED_WB_Q8[ 30 ] = {
+       175,    148,    160,    176,    178,    173,    174,    164,
+       177,    174,    196,    182,    198,    192,    182,     68,
+        62,     66,     60,     72,    117,     85,     90,    118,
+       136,    151,    142,    160,    142,    155
+};
+
+static const opus_int16 silk_NLSF_DELTA_MIN_WB_Q15[ 17 ] = {
+       100,      3,     40,      3,      3,      3,      5,     14,
+        14,     10,     11,      3,      8,      9,      7,      3,
+       347
+};
+
+const silk_NLSF_CB_struct silk_NLSF_CB_WB =
+{
+    32,
+    16,
+    SILK_FIX_CONST( 0.15, 16 ),
+    SILK_FIX_CONST( 1.0 / 0.15, 6 ),
+    silk_NLSF_CB1_WB_Q8,
+    silk_NLSF_CB1_iCDF_WB,
+    silk_NLSF_PRED_WB_Q8,
+    silk_NLSF_CB2_SELECT_WB,
+    silk_NLSF_CB2_iCDF_WB,
+    silk_NLSF_CB2_BITS_WB_Q5,
+    silk_NLSF_DELTA_MIN_WB_Q15,
+};
+
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_gain.c b/lib/rbcodec/codecs/libopus/silk/tables_gain.c
new file mode 100644
index 0000000000..fe6f912b10
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_gain.c
@@ -0,0 +1,63 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+const opus_uint8 silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ] =
+{
+{
+       224,    112,     44,     15,      3,      2,      1,      0
+},
+{
+       254,    237,    192,    132,     70,     23,      4,      0
+},
+{
+       255,    252,    226,    155,     61,     11,      2,      0
+}
+};
+
+const opus_uint8 silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ] = {
+       250,    245,    234,    203,     71,     50,     42,     38,
+        35,     33,     31,     29,     28,     27,     26,     25,
+        24,     23,     22,     21,     20,     19,     18,     17,
+        16,     15,     14,     13,     12,     11,     10,      9,
+         8,      7,      6,      5,      4,      3,      2,      1,
+         0
+};
+
+#ifdef __cplusplus
+}
+#endif
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_other.c b/lib/rbcodec/codecs/libopus/silk/tables_other.c
new file mode 100644
index 0000000000..5119ebd39f
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_other.c
@@ -0,0 +1,138 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "structs.h"
+#include "define.h"
+#include "tables.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/* Piece-wise linear mapping from bitrate in kbps to coding quality in dB SNR */
+const opus_int32 silk_TargetRate_table_NB[ TARGET_RATE_TAB_SZ ] = {
+    0,      8000,   9400,   11500,  13500,  17500,  25000,  MAX_TARGET_RATE_BPS
+};
+const opus_int32 silk_TargetRate_table_MB[ TARGET_RATE_TAB_SZ ] = {
+    0,      9000,   12000,  14500,  18500,  24500,  35500,  MAX_TARGET_RATE_BPS
+};
+const opus_int32 silk_TargetRate_table_WB[ TARGET_RATE_TAB_SZ ] = {
+    0,      10500,  14000,  17000,  21500,  28500,  42000,  MAX_TARGET_RATE_BPS
+};
+const opus_int16 silk_SNR_table_Q1[ TARGET_RATE_TAB_SZ ] = {
+    18,     29,     38,     40,     46,     52,     62,     84
+};
+
+/* Tables for stereo predictor coding */
+const opus_int16 silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ] = {
+    -13732, -10050, -8266, -7526, -6500, -5000, -2950,  -820,
+       820,   2950,  5000,  6500,  7526,  8266, 10050, 13732
+};
+const opus_uint8  silk_stereo_pred_joint_iCDF[ 25 ] = {
+    249, 247, 246, 245, 244,
+    234, 210, 202, 201, 200,
+    197, 174,  82,  59,  56,
+     55,  54,  46,  22,  12,
+     11,  10,   9,   7,   0
+};
+const opus_uint8  silk_stereo_only_code_mid_iCDF[ 2 ] = { 64, 0 };
+
+/* Tables for LBRR flags */
+static const opus_uint8 silk_LBRR_flags_2_iCDF[ 3 ] = { 203, 150, 0 };
+static const opus_uint8 silk_LBRR_flags_3_iCDF[ 7 ] = { 215, 195, 166, 125, 110, 82, 0 };
+const opus_uint8 * const silk_LBRR_flags_iCDF_ptr[ 2 ] = {
+    silk_LBRR_flags_2_iCDF,
+    silk_LBRR_flags_3_iCDF
+};
+
+/* Table for LSB coding */
+const opus_uint8 silk_lsb_iCDF[ 2 ] = { 120, 0 };
+
+/* Tables for LTPScale */
+const opus_uint8 silk_LTPscale_iCDF[ 3 ] = { 128, 64, 0 };
+
+/* Tables for signal type and offset coding */
+const opus_uint8 silk_type_offset_VAD_iCDF[ 4 ] = {
+       232,    158,    10,      0
+};
+const opus_uint8 silk_type_offset_no_VAD_iCDF[ 2 ] = {
+       230,      0
+};
+
+/* Tables for NLSF interpolation factor */
+const opus_uint8 silk_NLSF_interpolation_factor_iCDF[ 5 ] = { 243, 221, 192, 181, 0 };
+
+/* Quantization offsets */
+const opus_int16  silk_Quantization_Offsets_Q10[ 2 ][ 2 ] = {
+    { OFFSET_UVL_Q10, OFFSET_UVH_Q10 }, { OFFSET_VL_Q10, OFFSET_VH_Q10 }
+};
+
+/* Table for LTPScale */
+const opus_int16 silk_LTPScales_table_Q14[ 3 ] = { 15565, 12288, 8192 };
+
+/* Uniform entropy tables */
+const opus_uint8 silk_uniform3_iCDF[ 3 ] = { 171, 85, 0 };
+const opus_uint8 silk_uniform4_iCDF[ 4 ] = { 192, 128, 64, 0 };
+const opus_uint8 silk_uniform5_iCDF[ 5 ] = { 205, 154, 102, 51, 0 };
+const opus_uint8 silk_uniform6_iCDF[ 6 ] = { 213, 171, 128, 85, 43, 0 };
+const opus_uint8 silk_uniform8_iCDF[ 8 ] = { 224, 192, 160, 128, 96, 64, 32, 0 };
+
+const opus_uint8 silk_NLSF_EXT_iCDF[ 7 ] = { 100, 40, 16, 7, 3, 1, 0 };
+
+/*  Elliptic/Cauer filters designed with 0.1 dB passband ripple,
+        80 dB minimum stopband attenuation, and
+        [0.95 : 0.15 : 0.35] normalized cut off frequencies. */
+
+/* Interpolation points for filter coefficients used in the bandwidth transition smoother */
+const opus_int32 silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ] =
+{
+{    250767114,  501534038,  250767114  },
+{    209867381,  419732057,  209867381  },
+{    170987846,  341967853,  170987846  },
+{    131531482,  263046905,  131531482  },
+{     89306658,  178584282,   89306658  }
+};
+
+/* Interpolation points for filter coefficients used in the bandwidth transition smoother */
+const opus_int32 silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ] =
+{
+{    506393414,  239854379  },
+{    411067935,  169683996  },
+{    306733530,  116694253  },
+{    185807084,   77959395  },
+{     35497197,   57401098  }
+};
+
+#ifdef __cplusplus
+}
+#endif
+
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_pitch_lag.c b/lib/rbcodec/codecs/libopus/silk/tables_pitch_lag.c
new file mode 100644
index 0000000000..e795a23cd0
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_pitch_lag.c
@@ -0,0 +1,69 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+const opus_uint8 silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ] = {
+       253,    250,    244,    233,    212,    182,    150,    131,
+       120,    110,     98,     85,     72,     60,     49,     40,
+        32,     25,     19,     15,     13,     11,      9,      8,
+         7,      6,      5,      4,      3,      2,      1,      0
+};
+
+const opus_uint8 silk_pitch_delta_iCDF[21] = {
+       210,    208,    206,    203,    199,    193,    183,    168,
+       142,    104,     74,     52,     37,     27,     20,     14,
+        10,      6,      4,      2,      0
+};
+
+const opus_uint8 silk_pitch_contour_iCDF[34] = {
+       223,    201,    183,    167,    152,    138,    124,    111,
+        98,     88,     79,     70,     62,     56,     50,     44,
+        39,     35,     31,     27,     24,     21,     18,     16,
+        14,     12,     10,      8,      6,      4,      3,      2,
+         1,      0
+};
+
+const opus_uint8 silk_pitch_contour_NB_iCDF[11] = {
+       188,    176,    155,    138,    119,     97,     67,     43,
+        26,     10,      0
+};
+
+const opus_uint8 silk_pitch_contour_10_ms_iCDF[12] = {
+       165,    119,     80,     61,     47,     35,     27,     20,
+        14,      9,      4,      0
+};
+
+const opus_uint8 silk_pitch_contour_10_ms_NB_iCDF[3] = {
+       113,     63,      0
+};
+
+
diff --git a/lib/rbcodec/codecs/libopus/silk/tables_pulses_per_block.c b/lib/rbcodec/codecs/libopus/silk/tables_pulses_per_block.c
new file mode 100644
index 0000000000..0c9b18be6c
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/tables_pulses_per_block.c
@@ -0,0 +1,264 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "tables.h"
+
+const opus_uint8 silk_max_pulses_table[ 4 ] = {
+         8,     10,     12,     16
+};
+
+const opus_uint8 silk_pulses_per_block_iCDF[ 10 ][ 18 ] = {
+{
+       125,     51,     26,     18,     15,     12,     11,     10,
+         9,      8,      7,      6,      5,      4,      3,      2,
+         1,      0
+},
+{
+       198,    105,     45,     22,     15,     12,     11,     10,
+         9,      8,      7,      6,      5,      4,      3,      2,
+         1,      0
+},
+{
+       213,    162,    116,     83,     59,     43,     32,     24,
+        18,     15,     12,      9,      7,      6,      5,      3,
+         2,      0
+},
+{
+       239,    187,    116,     59,     28,     16,     11,     10,
+         9,      8,      7,      6,      5,      4,      3,      2,
+         1,      0
+},
+{
+       250,    229,    188,    135,     86,     51,     30,     19,
+        13,     10,      8,      6,      5,      4,      3,      2,
+         1,      0
+},
+{
+       249,    235,    213,    185,    156,    128,    103,     83,
+        66,     53,     42,     33,     26,     21,     17,     13,
+        10,      0
+},
+{
+       254,    249,    235,    206,    164,    118,     77,     46,
+        27,     16,     10,      7,      5,      4,      3,      2,
+         1,      0
+},
+{
+       255,    253,    249,    239,    220,    191,    156,    119,
+        85,     57,     37,     23,     15,     10,      6,      4,
+         2,      0
+},
+{
+       255,    253,    251,    246,    237,    223,    203,    179,
+       152,    124,     98,     75,     55,     40,     29,     21,
+        15,      0
+},
+{
+       255,    254,    253,    247,    220,    162,    106,     67,
+        42,     28,     18,     12,      9,      6,      4,      3,
+         2,      0
+}
+};
+
+const opus_uint8 silk_pulses_per_block_BITS_Q5[ 9 ][ 18 ] = {
+{
+        31,     57,    107,    160,    205,    205,    255,    255,
+       255,    255,    255,    255,    255,    255,    255,    255,
+       255,    255
+},
+{
+        69,     47,     67,    111,    166,    205,    255,    255,
+       255,    255,    255,    255,    255,    255,    255,    255,
+       255,    255
+},
+{
+        82,     74,     79,     95,    109,    128,    145,    160,
+       173,    205,    205,    205,    224,    255,    255,    224,
+       255,    224
+},
+{
+       125,     74,     59,     69,     97,    141,    182,    255,
+       255,    255,    255,    255,    255,    255,    255,    255,
+       255,    255
+},
+{
+       173,    115,     85,     73,     76,     92,    115,    145,
+       173,    205,    224,    224,    255,    255,    255,    255,
+       255,    255
+},
+{
+       166,    134,    113,    102,    101,    102,    107,    118,
+       125,    138,    145,    155,    166,    182,    192,    192,
+       205,    150
+},
+{
+       224,    182,    134,    101,     83,     79,     85,     97,
+       120,    145,    173,    205,    224,    255,    255,    255,
+       255,    255
+},
+{
+       255,    224,    192,    150,    120,    101,     92,     89,
+        93,    102,    118,    134,    160,    182,    192,    224,
+       224,    224
+},
+{
+       255,    224,    224,    182,    155,    134,    118,    109,
+       104,    102,    106,    111,    118,    131,    145,    160,
+       173,    131
+}
+};
+
+const opus_uint8 silk_rate_levels_iCDF[ 2 ][ 9 ] =
+{
+{
+       241,    190,    178,    132,     87,     74,     41,     14,
+         0
+},
+{
+       223,    193,    157,    140,    106,     57,     39,     18,
+         0
+}
+};
+
+const opus_uint8 silk_rate_levels_BITS_Q5[ 2 ][ 9 ] =
+{
+{
+       131,     74,    141,     79,     80,    138,     95,    104,
+       134
+},
+{
+        95,     99,     91,    125,     93,     76,    123,    115,
+       123
+}
+};
+
+const opus_uint8 silk_shell_code_table0[ 152 ] = {
+       128,      0,    214,     42,      0,    235,    128,     21,
+         0,    244,    184,     72,     11,      0,    248,    214,
+       128,     42,      7,      0,    248,    225,    170,     80,
+        25,      5,      0,    251,    236,    198,    126,     54,
+        18,      3,      0,    250,    238,    211,    159,     82,
+        35,     15,      5,      0,    250,    231,    203,    168,
+       128,     88,     53,     25,      6,      0,    252,    238,
+       216,    185,    148,    108,     71,     40,     18,      4,
+         0,    253,    243,    225,    199,    166,    128,     90,
+        57,     31,     13,      3,      0,    254,    246,    233,
+       212,    183,    147,    109,     73,     44,     23,     10,
+         2,      0,    255,    250,    240,    223,    198,    166,
+       128,     90,     58,     33,     16,      6,      1,      0,
+       255,    251,    244,    231,    210,    181,    146,    110,
+        75,     46,     25,     12,      5,      1,      0,    255,
+       253,    248,    238,    221,    196,    164,    128,     92,
+        60,     35,     18,      8,      3,      1,      0,    255,
+       253,    249,    242,    229,    208,    180,    146,    110,
+        76,     48,     27,     14,      7,      3,      1,      0
+};
+
+const opus_uint8 silk_shell_code_table1[ 152 ] = {
+       129,      0,    207,     50,      0,    236,    129,     20,
+         0,    245,    185,     72,     10,      0,    249,    213,
+       129,     42,      6,      0,    250,    226,    169,     87,
+        27,      4,      0,    251,    233,    194,    130,     62,
+        20,      4,      0,    250,    236,    207,    160,     99,
+        47,     17,      3,      0,    255,    240,    217,    182,
+       131,     81,     41,     11,      1,      0,    255,    254,
+       233,    201,    159,    107,     61,     20,      2,      1,
+         0,    255,    249,    233,    206,    170,    128,     86,
+        50,     23,      7,      1,      0,    255,    250,    238,
+       217,    186,    148,    108,     70,     39,     18,      6,
+         1,      0,    255,    252,    243,    226,    200,    166,
+       128,     90,     56,     30,     13,      4,      1,      0,
+       255,    252,    245,    231,    209,    180,    146,    110,
+        76,     47,     25,     11,      4,      1,      0,    255,
+       253,    248,    237,    219,    194,    163,    128,     93,
+        62,     37,     19,      8,      3,      1,      0,    255,
+       254,    250,    241,    226,    205,    177,    145,    111,
+        79,     51,     30,     15,      6,      2,      1,      0
+};
+
+const opus_uint8 silk_shell_code_table2[ 152 ] = {
+       129,      0,    203,     54,      0,    234,    129,     23,
+         0,    245,    184,     73,     10,      0,    250,    215,
+       129,     41,      5,      0,    252,    232,    173,     86,
+        24,      3,      0,    253,    240,    200,    129,     56,
+        15,      2,      0,    253,    244,    217,    164,     94,
+        38,     10,      1,      0,    253,    245,    226,    189,
+       132,     71,     27,      7,      1,      0,    253,    246,
+       231,    203,    159,    105,     56,     23,      6,      1,
+         0,    255,    248,    235,    213,    179,    133,     85,
+        47,     19,      5,      1,      0,    255,    254,    243,
+       221,    194,    159,    117,     70,     37,     12,      2,
+         1,      0,    255,    254,    248,    234,    208,    171,
+       128,     85,     48,     22,      8,      2,      1,      0,
+       255,    254,    250,    240,    220,    189,    149,    107,
+        67,     36,     16,      6,      2,      1,      0,    255,
+       254,    251,    243,    227,    201,    166,    128,     90,
+        55,     29,     13,      5,      2,      1,      0,    255,
+       254,    252,    246,    234,    213,    183,    147,    109,
+        73,     43,     22,     10,      4,      2,      1,      0
+};
+
+const opus_uint8 silk_shell_code_table3[ 152 ] = {
+       130,      0,    200,     58,      0,    231,    130,     26,
+         0,    244,    184,     76,     12,      0,    249,    214,
+       130,     43,      6,      0,    252,    232,    173,     87,
+        24,      3,      0,    253,    241,    203,    131,     56,
+        14,      2,      0,    254,    246,    221,    167,     94,
+        35,      8,      1,      0,    254,    249,    232,    193,
+       130,     65,     23,      5,      1,      0,    255,    251,
+       239,    211,    162,     99,     45,     15,      4,      1,
+         0,    255,    251,    243,    223,    186,    131,     74,
+        33,     11,      3,      1,      0,    255,    252,    245,
+       230,    202,    158,    105,     57,     24,      8,      2,
+         1,      0,    255,    253,    247,    235,    214,    179,
+       132,     84,     44,     19,      7,      2,      1,      0,
+       255,    254,    250,    240,    223,    196,    159,    112,
+        69,     36,     15,      6,      2,      1,      0,    255,
+       254,    253,    245,    231,    209,    176,    136,     93,
+        55,     27,     11,      3,      2,      1,      0,    255,
+       254,    253,    252,    239,    221,    194,    158,    117,
+        76,     42,     18,      4,      3,      2,      1,      0
+};
+
+const opus_uint8 silk_shell_code_table_offsets[ 17 ] = {
+         0,      0,      2,      5,      9,     14,     20,     27,
+        35,     44,     54,     65,     77,     90,    104,    119,
+       135
+};
+
+const opus_uint8 silk_sign_iCDF[ 42 ] = {
+       254,     49,     67,     77,     82,     93,     99,
+       198,     11,     18,     24,     31,     36,     45,
+       255,     46,     66,     78,     87,     94,    104,
+       208,     14,     21,     32,     42,     51,     66,
+       255,     94,    104,    109,    112,    115,    118,
+       248,     53,     69,     80,     88,     95,    102
+};
diff --git a/lib/rbcodec/codecs/libopus/silk/typedef.h b/lib/rbcodec/codecs/libopus/silk/typedef.h
new file mode 100644
index 0000000000..da98123701
--- /dev/null
+++ b/lib/rbcodec/codecs/libopus/silk/typedef.h
@@ -0,0 +1,77 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS”
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifndef SILK_TYPEDEF_H
+#define SILK_TYPEDEF_H
+
+#include "opus_types.h"
+
+#ifndef FIXED_POINT
+# include <float.h>
+# define silk_float      float
+# define silk_float_MAX  FLT_MAX
+#endif
+
+#define silk_int64_MAX   ((opus_int64)0x7FFFFFFFFFFFFFFFLL)   /*  2^63 - 1 */
+#define silk_int64_MIN   ((opus_int64)0x8000000000000000LL)   /* -2^63 */
+#define silk_int32_MAX   0x7FFFFFFF                           /*  2^31 - 1 =  2147483647 */
+#define silk_int32_MIN   ((opus_int32)0x80000000)             /* -2^31     = -2147483648 */
+#define silk_int16_MAX   0x7FFF                               /*  2^15 - 1 =  32767 */
+#define silk_int16_MIN   ((opus_int16)0x8000)                 /* -2^15     = -32768 */
+#define silk_int8_MAX    0x7F                                 /*  2^7 - 1  =  127 */
+#define silk_int8_MIN    ((opus_int8)0x80)                    /* -2^7      = -128 */
+#define silk_uint8_MAX   0xFF                                 /*  2^8 - 1 = 255 */
+
+#define silk_TRUE        1
+#define silk_FALSE       0
+
+/* assertions */
+#if (defined _WIN32 && !defined _WINCE && !defined(__GNUC__) && !defined(NO_ASSERTS))
+# ifndef silk_assert
+#  include <crtdbg.h>      /* ASSERTE() */
+#  define silk_assert(COND)   _ASSERTE(COND)
+# endif
+#else
+# ifdef ENABLE_ASSERTIONS
+#  include <stdio.h>
+#  include <stdlib.h>
+#define silk_fatal(str) _silk_fatal(str, __FILE__, __LINE__);
+#ifdef __GNUC__
+__attribute__((noreturn))
+#endif
+static inline void _silk_fatal(const char *str, const char *file, int line)
+{
+   fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str);
+   abort();
+}
+#  define silk_assert(COND) {if (!(COND)) {silk_fatal("assertion failed: " #COND);}}
+# else
+#  define silk_assert(COND)
+# endif
+#endif
+
+#endif /* SILK_TYPEDEF_H */
diff --git a/lib/rbcodec/codecs/opus.c b/lib/rbcodec/codecs/opus.c
new file mode 100644
index 0000000000..19bdb8daae
--- /dev/null
+++ b/lib/rbcodec/codecs/opus.c
@@ -0,0 +1,461 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2012 Frederik M.J. Vestre
+ * Based on speex.c codec interface:
+ * Copyright (C) 2006 Frederik M.J. Vestre
+ * Based on vorbis.c codec interface:
+ * Copyright (C) 2002 Björn Stenberg
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codeclib.h"
+#include "inttypes.h"
+#include "libopus/opus.h"
+#include "libopus/opus_header.h"
+
+
+#include "libopus/ogg/ogg.h"
+#ifdef SIMULATOR
+#include <tlsf.h>
+#endif
+
+CODEC_HEADER
+
+#define SEEK_REWIND 3840    /* 80 ms @ 48 kHz */
+
+/* the opus pseudo stack pointer */
+extern char *global_stack;
+
+/* Room for 120 ms of stereo audio at 48 kHz */
+#define MAX_FRAME_SIZE  (2*120*48)
+#define CHUNKSIZE       (16*1024)
+#define SEEK_CHUNKSIZE 7*CHUNKSIZE
+
+static int get_more_data(ogg_sync_state *oy)
+{
+    int bytes;
+    char *buffer;
+
+    buffer = (char *)ogg_sync_buffer(oy, CHUNKSIZE);
+    bytes = ci->read_filebuf(buffer, CHUNKSIZE);
+    ogg_sync_wrote(oy,bytes);
+
+    return bytes;
+}
+/* The read/seek functions track absolute position within the stream */
+static int64_t get_next_page(ogg_sync_state *oy, ogg_page *og,
+                                 int64_t boundary)
+{
+    int64_t localoffset = ci->curpos;
+    long more;
+    long ret;
+
+    if (boundary > 0)
+        boundary += ci->curpos;
+
+    while (1) {
+        more = ogg_sync_pageseek(oy,og);
+
+        if (more < 0) {
+            /* skipped n bytes */
+            localoffset-=more;
+        } else {
+            if (more == 0) {
+                /* send more data */
+                if(!boundary)return(-1);
+                {
+                    ret = get_more_data(oy);
+                    if (ret == 0)
+                        return(-2);
+
+                    if (ret < 0)
+                        return(-3);
+                }
+            } else {
+                /* got a page.  Return the offset at the page beginning,
+                   advance the internal offset past the page end */
+
+                int64_t ret=localoffset;
+
+                return(ret);
+            }
+        }
+    }
+}
+
+static int64_t seek_backwards(ogg_sync_state *oy, ogg_page *og,
+                                  int64_t wantedpos)
+{
+    int64_t crofs;
+    int64_t *curoffset=&crofs;
+    *curoffset=ci->curpos;
+    int64_t begin=*curoffset;
+    int64_t end=begin;
+    int64_t ret;
+    int64_t offset=-1;
+    int64_t avgpagelen=-1;
+    int64_t lastgranule=-1;
+
+    short time = -1;
+
+    while (offset == -1) {
+
+        begin -= SEEK_CHUNKSIZE;
+
+        if (begin < 0) {
+            if (time < 0) {
+                begin = 0;
+                time++;
+            } else {
+                LOGF("Can't seek that early:%lld\n",begin);
+                return -3;  /* too early */
+            }
+        }
+
+        *curoffset = begin;
+
+        ci->seek_buffer(*curoffset);
+
+        ogg_sync_reset(oy);
+
+        lastgranule = -1;
+
+        while (*curoffset < end) {
+            ret = get_next_page(oy,og,end-*curoffset);
+
+            if (ret > 0) {
+                if (lastgranule != -1) {
+                    if (avgpagelen < 0)
+                        avgpagelen = (ogg_page_granulepos(og)-lastgranule);
+                    else
+                       avgpagelen=((ogg_page_granulepos(og)-lastgranule)
+                                   + avgpagelen) / 2;
+                }
+
+                lastgranule=ogg_page_granulepos(og);
+
+                if ((lastgranule - (avgpagelen/4)) < wantedpos &&
+                    (lastgranule + avgpagelen + (avgpagelen/4)) > wantedpos) {
+
+                    /*wanted offset found Yeay!*/
+
+                    /*LOGF("GnPagefound:%d,%d,%d,%d\n",ret,
+                           lastgranule,wantedpos,avgpagelen);*/
+
+                    return ret;
+
+                } else if (lastgranule > wantedpos) {  /*too late, seek more*/
+                    if (offset != -1) {
+                        LOGF("Toolate, returnanyway:%lld,%lld,%lld,%lld\n",
+                             ret,lastgranule,wantedpos,avgpagelen);
+                        return ret;
+                    }
+                    break;
+                } else{ /*if (ogg_page_granulepos(&og)<wantedpos)*/
+                    /*too early*/
+                    offset = ret;
+                    continue;
+                }
+            } else if (ret == -3) 
+                return(-3);
+            else if (ret<=0)
+                break;
+            else if (*curoffset < end) {
+                /*this should not be possible*/
+
+                //LOGF("Seek:get_earlier_page:Offset:not_cached by granule:"\"%d,%d,%d,%d,%d\n",*curoffset,end,begin,wantedpos,curpos);
+
+                offset=ret;
+            }
+        }
+    }
+    return -1;
+}
+
+static int speex_seek_page_granule(int64_t pos, int64_t curpos,
+                                   ogg_sync_state *oy,
+                                   int64_t headerssize)
+{
+    /* TODO: Someone may want to try to implement seek to packet, 
+             instead of just to page (should be more accurate, not be any 
+             faster) */
+
+    int64_t crofs;
+    int64_t *curbyteoffset = &crofs;
+    *curbyteoffset = ci->curpos;
+    int64_t curoffset;
+    curoffset = *curbyteoffset;
+    int64_t offset = 0;
+    ogg_page og = {0,0,0,0};
+    int64_t avgpagelen = -1;
+    int64_t lastgranule = -1;
+
+    if(abs(pos-curpos)>10000 && headerssize>0 && curoffset-headerssize>10000) {
+        /* if seeking for more that 10sec,
+           headersize is known & more than 10kb is played,
+           try to guess a place to seek from the number of
+           bytes playe for this position, this works best when 
+           the bitrate is relativly constant.
+         */
+
+        curoffset = (((*curbyteoffset-headerssize) * pos)/curpos)*98/100;
+        if (curoffset < 0)
+            curoffset=0;
+
+        //int64_t toffset=curoffset;
+
+        ci->seek_buffer(curoffset);
+
+        ogg_sync_reset(oy);
+
+        offset = get_next_page(oy,&og,-1);
+
+        if (offset < 0) { /* could not find new page,use old offset */
+            LOGF("Seek/guess/fault:%lld->-<-%d,%lld:%lld,%d,%ld,%d\n",
+                 curpos,0,pos,offset,0,
+                 ci->curpos,/*stream_length*/0);
+
+            curoffset = *curbyteoffset;
+
+            ci->seek_buffer(curoffset);
+
+            ogg_sync_reset(oy);
+        } else {
+            if (ogg_page_granulepos(&og) == 0 && pos > 5000) {
+                LOGF("SEEK/guess/fault:%lld->-<-%lld,%lld:%lld,%d,%ld,%d\n",
+                     curpos,ogg_page_granulepos(&og),pos,
+                     offset,0,ci->curpos,/*stream_length*/0);
+
+                curoffset = *curbyteoffset;
+
+                ci->seek_buffer(curoffset);
+
+                ogg_sync_reset(oy);
+            } else {
+                curoffset = offset;
+                curpos = ogg_page_granulepos(&og);
+            }
+        }
+    }
+
+    /* which way do we want to seek? */
+
+    if (curpos > pos) {  /* backwards */
+        offset = seek_backwards(oy,&og,pos);
+
+        if (offset > 0) {
+            *curbyteoffset = curoffset;
+            return 1;
+        }
+    } else {  /* forwards */
+
+        while ( (offset = get_next_page(oy,&og,-1)) > 0) {
+            if (lastgranule != -1) {
+               if (avgpagelen < 0)
+                   avgpagelen = (ogg_page_granulepos(&og) - lastgranule);
+               else
+                   avgpagelen = ((ogg_page_granulepos(&og) - lastgranule)
+                                 + avgpagelen) / 2;
+            }
+
+            lastgranule = ogg_page_granulepos(&og);
+
+            if ( ((lastgranule - (avgpagelen/4)) < pos && ( lastgranule + 
+                  avgpagelen + (avgpagelen / 4)) > pos) ||
+                 lastgranule > pos) {
+
+                /*wanted offset found Yeay!*/
+
+                *curbyteoffset = offset;
+
+                return offset;
+            }
+        }
+    }
+
+    ci->seek_buffer(*curbyteoffset);
+
+    ogg_sync_reset(oy);
+
+    LOGF("Seek failed:%lld\n", offset);
+
+    return -1;
+}
+
+
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    (void)reason;
+
+    return CODEC_OK;
+}
+
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
+    int error = CODEC_ERROR;
+    intptr_t param;
+    ogg_sync_state oy;
+    ogg_page og;
+    ogg_packet op;
+    ogg_stream_state os;
+    int64_t page_granule = 0;
+    int stream_init = 0;
+    int sample_rate = 48000;
+    OpusDecoder *st = NULL;
+    OpusHeader header;
+    int ret;
+    unsigned long strtoffset = ci->id3->offset;
+    int skip = 0;
+    int64_t seek_target;
+    uint64_t granule_pos;
+
+    /* reset our simple malloc */
+    if (codec_init()) {
+        goto done;
+    }
+    global_stack = 0;
+
+    /* pre-init the ogg_sync_state buffer, so it won't need many reallocs */
+    ogg_sync_init(&oy);
+    oy.storage = 64*1024;
+    oy.data = codec_malloc(oy.storage);
+
+    /* allocate output buffer */
+    uint16_t *output = (uint16_t*) codec_malloc(MAX_FRAME_SIZE*sizeof(uint16_t));
+
+    ci->seek_buffer(0);
+    ci->set_elapsed(0);
+
+    while (1) {
+        enum codec_command_action action = ci->get_command(&param);
+
+        if (action == CODEC_ACTION_HALT)
+            break;
+
+        if (action == CODEC_ACTION_SEEK_TIME) {
+            if (st != NULL) {
+                /* calculate granule to seek to (including seek rewind) */
+                seek_target = (48LL * param) + header.preskip;
+                skip = MIN(seek_target, SEEK_REWIND);
+                seek_target -= skip;
+
+                LOGF("Opus seek page:%lld,%lld,%ld\n",
+                            seek_target, page_granule, (long)param);
+                speex_seek_page_granule(seek_target, page_granule, &oy, 0);
+            }
+
+            ci->set_elapsed(param);
+            ci->seek_complete();
+        }
+
+        /*Get the ogg buffer for writing*/
+        if (get_more_data(&oy) < 1) {
+            goto done;
+        }
+
+        /* Loop for all complete pages we got (most likely only one) */
+        while (ogg_sync_pageout(&oy, &og) == 1) {
+            if (stream_init == 0) {
+                ogg_stream_init(&os, ogg_page_serialno(&og));
+                stream_init = 1;
+            }
+
+            /* Add page to the bitstream */
+            ogg_stream_pagein(&os, &og);
+
+            page_granule = ogg_page_granulepos(&og);
+            granule_pos = page_granule;
+
+            while ((ogg_stream_packetout(&os, &op) == 1) && !op.e_o_s) {
+                if (op.packetno == 0){
+                    /* identification header */
+                
+                    if (opus_header_parse(op.packet, op.bytes, &header) == 0) {
+                        LOGF("Could not parse header");
+                        goto done;
+                    }
+                    skip = header.preskip;
+
+                    st = opus_decoder_create(sample_rate, header.channels, &ret);
+                    if (ret != OPUS_OK) {
+                        LOGF("opus_decoder_create failed %d", ret);
+                        goto done;
+                    }
+                    LOGF("Decoder inited");
+
+                    codec_set_replaygain(ci->id3);
+
+                    opus_decoder_ctl(st, OPUS_SET_GAIN(header.gain));
+
+                    ci->configure(DSP_SET_FREQUENCY, sample_rate);
+                    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
+                    ci->configure(DSP_SET_STEREO_MODE, (header.channels == 2) ?
+                        STEREO_INTERLEAVED : STEREO_MONO);
+
+                } else if (op.packetno == 1) {
+                    /* Comment header */
+                } else {
+                    if (strtoffset) {
+                        ci->seek_buffer(strtoffset);
+                        ogg_sync_reset(&oy);
+                        strtoffset = 0;
+                        break;//next page
+                    }
+
+                    /* report progress */
+                    ci->set_elapsed((granule_pos - header.preskip) / 48);
+
+                    /* Decode audio packets */
+                    ret = opus_decode(st, op.packet, op.bytes, output, MAX_FRAME_SIZE, 0);
+
+                    if (ret > 0) {
+                        if (skip > 0) {
+                            if (ret <= skip) {
+                                /* entire output buffer is skipped */
+                                skip -= ret;
+                                ret = 0;
+                            } else {
+                                /* part of output buffer is played */
+                                ret -= skip;
+                                ci->pcmbuf_insert(&output[skip * header.channels], NULL, ret);
+                                skip = 0;
+                            }
+                        } else {
+                            /* entire buffer is played */
+                            ci->pcmbuf_insert(output, NULL, ret);
+                        }
+                        granule_pos += ret;
+                    } else {
+                        if (ret < 0) {
+                            LOGF("opus_decode failed %d", ret);
+                            goto done;
+                        }
+                        break;
+                    }
+                }
+            }
+        }
+    }
+    LOGF("Returned OK");
+    error = CODEC_OK;
+done:
+    return error;
+}
+
diff --git a/lib/rbcodec/metadata/metadata.c b/lib/rbcodec/metadata/metadata.c
index 668ece97cf..5caf933b55 100644
--- a/lib/rbcodec/metadata/metadata.c
+++ b/lib/rbcodec/metadata/metadata.c
@@ -233,6 +233,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
     /* KSS (MSX computer KSS Music File) */
     [AFMT_KSS] =
         AFMT_ENTRY("KSS", "kss", NULL, get_kss_metadata,   "kss\0"),
+    /* Opus */
+    [AFMT_OPUS] =
+        AFMT_ENTRY("Opus", "opus", NULL, get_ogg_metadata,   "opus\0"),
 #endif
 };
 
diff --git a/lib/rbcodec/metadata/metadata.h b/lib/rbcodec/metadata/metadata.h
index 4b4a8337fb..18cfce7523 100644
--- a/lib/rbcodec/metadata/metadata.h
+++ b/lib/rbcodec/metadata/metadata.h
@@ -89,6 +89,7 @@ enum
     AFMT_SGC,          /* SGC (Sega Master System, Game Gear, Coleco Vision Sound Format) */
     AFMT_VGM,         /* VGM (Video Game Music Format) */
     AFMT_KSS,          /* KSS (MSX computer KSS Music File) */
+    AFMT_OPUS,         /* Opus (see http://www.opus-codec.org ) */
 #endif
 
     /* add new formats at any index above this line to have a sensible order -
diff --git a/lib/rbcodec/metadata/ogg.c b/lib/rbcodec/metadata/ogg.c
index f3231c6f8a..5d302442ed 100644
--- a/lib/rbcodec/metadata/ogg.c
+++ b/lib/rbcodec/metadata/ogg.c
@@ -102,6 +102,20 @@ bool get_ogg_metadata(int fd, struct mp3entry* id3)
             return false;
         }
     }
+    else if (memcmp(&buf[28], "OpusHead", 8) == 0)
+    {
+        id3->codectype = AFMT_OPUS;
+        id3->frequency = 48000;
+        id3->vbr = true;
+
+// FIXME handle an actual channel mapping table
+        /* Comments are in second Ogg page (byte 108 onwards for Speex) */
+        if (lseek(fd, 47, SEEK_SET) < 0)
+        {
+            DEBUGF("Couldnotseektoogg");
+            return false;
+        }
+    }
     else
     {
         /* Unsupported format, try to print the marker, catches Ogg/FLAC at least */
diff --git a/lib/rbcodec/metadata/vorbis.c b/lib/rbcodec/metadata/vorbis.c
index d020808c56..66aa85022f 100644
--- a/lib/rbcodec/metadata/vorbis.c
+++ b/lib/rbcodec/metadata/vorbis.c
@@ -250,7 +250,7 @@ static bool file_init(struct file* file, int fd, int type, int remaining)
     memset(file, 0, sizeof(*file));
     file->fd = fd;
     
-    if (type == AFMT_OGG_VORBIS || type == AFMT_SPEEX)
+    if (type == AFMT_OGG_VORBIS || type == AFMT_SPEEX || type == AFMT_OPUS)
     {
         if (!file_read_page_header(file))
         {
@@ -276,6 +276,22 @@ static bool file_init(struct file* file, int fd, int type, int remaining)
             return false;
         }
     }
+    else if (type == AFMT_OPUS)
+    {
+        char buffer[8];
+
+        /* Read comment header */
+        if (file_read(file, buffer, sizeof(buffer)) < (ssize_t) sizeof(buffer))
+        {
+            return false;
+        }
+
+        /* Should be equal to "OpusTags" */
+        if (memcmp(buffer, "OpusTags", 8) != 0)
+        {
+            return false;
+        }
+    }
     else if (type == AFMT_FLAC)
     {
         file->packet_remaining = remaining;