The first part of Mohamed Tarek's Google Summer of Code work to implement RealAudio support in Rockbox. This is a self-contained Cook decoder using the original ffmpeg (still floating point) decoder and a new RM parser started by me in 2008 and continued by MT over the past few months. This is the equivalent of libcook.patch1 from FS#10182, but with further cleaning by both MT and me to minimise the differences to the original ffmpeg files. See README.rockbox for more details.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@20883 a1c6a512-1295-4272-9138-f99709370657

16 files changed, 925 insertions, 111 deletions

diff --git a/apps/codecs/libcook/Makefile.test b/apps/codecs/libcook/Makefile.test
new file mode 100644
index 0000000000..98ee7ffcb4
--- /dev/null
+++ b/apps/codecs/libcook/Makefile.test
@@ -0,0 +1,10 @@
+CFLAGS = -Wall -O3
+OBJS = bitstream.o cook.o fft.o libavutil/log.o mdct.o libavutil/mem.o libavutil/lfg.o libavutil/md5.o rm2wav.o
+cooktest: $(OBJS)
+        gcc -o cooktest $(OBJS) -lm
+
+.c.o :
+        $(CC) $(CFLAGS) -c -o $@ $<
+
+clean:
+        rm -f cooktest $(OBJS) *~
diff --git a/apps/codecs/libcook/README.rockbox b/apps/codecs/libcook/README.rockbox
new file mode 100644
index 0000000000..9447b96be5
--- /dev/null
+++ b/apps/codecs/libcook/README.rockbox
@@ -0,0 +1,40 @@
+Library: libcook
+Imported by : Mohamed Tarek
+
+These files comprise a rm parser and a cook decoder based on the decoder
+from ffmpeg.
+
+LICENSING INFORMATION
+
+ffmpeg is licensed under the Lesser GNU General Public License and the
+file cook.c is Copyright 2003 Sascha Sommer and 2005 Benjamin Larsson.
+
+IMPORT DETAILS
+
+The decoder is based on ffmpeg-svn r18079.
+
+The file libavcodec/cook.c was modified to remove all ffmpeg-specific
+code and to use the current rm parser.
+
+In initializing random_state in COOKContext, ffmpeg used a random_seed()
+function that was weird in the way it gets the random number. So it was
+decided to not use this function at all,and the value was initialized to 1,
+because according to Benjamin Larsson random_seed() could just be 
+replaced by any value.
+
+The current files contain lots of code which is either not needed by
+the decoder or totally disabled (#if 0 .. #endif) to enable
+compiling. This was during the isolation of the decoder from ffmpeg,
+the intention was to take as few files as possible to be able to
+compile cook.c and the related files outside ffmpeg.
+
+The decoder still uses floating point and relies on dynamic allocations 
+in some parts of it. It's still not ready to be ported to rockbox.
+
+TESTING
+
+The test program should compile in any Unix-like environment using the
+command "make -f Makefile.test".
+
+Running "./cooktest file.rm" will decode the audio data to a WAV file
+called "output.wav" in the current directory.
diff --git a/apps/codecs/libcook/avcodec.h b/apps/codecs/libcook/avcodec.h
index a520340162..08eb800d4b 100644
--- a/apps/codecs/libcook/avcodec.h
+++ b/apps/codecs/libcook/avcodec.h
@@ -406,6 +406,7 @@ enum SampleFormat {
 #define FF_MIN_BUFFER_SIZE 16384
 
 
+#if 0/* MT : DELETE THIS LINE.*/
 /**
  * motion estimation type.
  */
@@ -3429,4 +3430,6 @@ void av_register_hwaccel(AVHWAccel *hwaccel);
  */
 AVHWAccel *av_hwaccel_next(AVHWAccel *hwaccel);
 
+#endif/* MT : DELETE THIS LINE.*/
+
 #endif /* AVCODEC_AVCODEC_H */
diff --git a/apps/codecs/libcook/bitstream.c b/apps/codecs/libcook/bitstream.c
index 6781fe6cb0..f32e88fdbf 100644
--- a/apps/codecs/libcook/bitstream.c
+++ b/apps/codecs/libcook/bitstream.c
@@ -73,6 +73,7 @@ void ff_put_string(PutBitContext * pbc, const char *s, int put_zero)
         put_bits(pbc, 8, 0);
 }
 
+#if 0
 void ff_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
 {
     const uint16_t *srcw= (const uint16_t*)src;
@@ -94,6 +95,7 @@ void ff_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
 
     put_bits(pb, bits, AV_RB16(&srcw[words])>>(16-bits));
 }
+#endif
 
 /* VLC decoding */
 
diff --git a/apps/codecs/libcook/bitstream.h b/apps/codecs/libcook/bitstream.h
index 3670285904..e3e3d77595 100644
--- a/apps/codecs/libcook/bitstream.h
+++ b/apps/codecs/libcook/bitstream.h
@@ -33,7 +33,7 @@
 #include "libavutil/common.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/log.h"
-#include "mathops.h"
+//#include "mathops.h"
 
 #if defined(ALT_BITSTREAM_READER_LE) && !defined(ALT_BITSTREAM_READER)
 #   define ALT_BITSTREAM_READER
@@ -707,12 +707,14 @@ static inline unsigned int get_bits_long(GetBitContext *s, int n){
     }
 }
 
+#if 0
 /**
  * reads 0-32 bits as a signed integer.
  */
 static inline int get_sbits_long(GetBitContext *s, int n) {
     return sign_extend(get_bits_long(s, n), n);
 }
+#endif
 
 /**
  * shows 0-32 bits.
diff --git a/apps/codecs/libcook/cook.c b/apps/codecs/libcook/cook.c
index cee69fe14a..856004f31b 100644
--- a/apps/codecs/libcook/cook.c
+++ b/apps/codecs/libcook/cook.c
@@ -45,15 +45,30 @@
 #include <math.h>
 #include <stddef.h>
 #include <stdio.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <unistd.h>
 
 #include "libavutil/lfg.h"
-#include "libavutil/random_seed.h"
-#include "avcodec.h"
 #include "bitstream.h"
 #include "dsputil.h"
 #include "bytestream.h"
 
 #include "cookdata.h"
+#include "rm2wav.h"
+
+/* The following table is taken from libavutil/mathematics.c */
+const uint8_t ff_log2_tab[256]={
+        0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
+        5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
+        6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+        6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+        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,
+        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
+};
 
 /* the different Cook versions */
 #define MONO            0x1000001
@@ -64,6 +79,8 @@
 #define SUBBAND_SIZE    20
 #define MAX_SUBPACKETS   5
 //#define COOKDEBUG
+//#define DUMP_RAW_FRAMES
+#define DEBUGF(message,args ...) av_log(NULL,AV_LOG_ERROR,message,## args)
 
 typedef struct {
     int *now;
@@ -93,8 +110,10 @@ typedef struct cook {
 
     void (* saturate_output) (struct cook *q, int chan, int16_t *out);
 
-    AVCodecContext*     avctx;
     GetBitContext       gb;
+    int                 frame_number;
+    int                 block_align;
+    int                 extradata_size;
     /* stream data */
     int                 nb_channels;
     int                 joint_stereo;
@@ -112,7 +131,6 @@ typedef struct cook {
     int                 cookversion;
     /* states */
     AVLFG               random_state;
-
     /* transform data */
     MDCTContext         mdct_ctx;
     float*              mlt_window;
@@ -138,7 +156,7 @@ typedef struct cook {
     /* data buffers */
 
     uint8_t*            decoded_bytes_buffer;
-    DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
+    float mono_mdct_output[2048] __attribute__ ((aligned(16))); //DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
     float               mono_previous_buffer1[1024];
     float               mono_previous_buffer2[1024];
     float               decode_buffer_1[1024];
@@ -214,7 +232,7 @@ static av_cold int init_cook_vlc_tables(COOKContext *q) {
             envelope_quant_index_huffbits[i], 1, 1,
             envelope_quant_index_huffcodes[i], 2, 2, 0);
     }
-    av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n");
+    av_log(NULL,AV_LOG_DEBUG,"sqvh VLC init\n");
     for (i=0 ; i<7 ; i++) {
         result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
             cvh_huffbits[i], 1, 1,
@@ -225,10 +243,10 @@ static av_cold int init_cook_vlc_tables(COOKContext *q) {
         result |= init_vlc (&q->ccpl, 6, (1<<q->js_vlc_bits)-1,
             ccpl_huffbits[q->js_vlc_bits-2], 1, 1,
             ccpl_huffcodes[q->js_vlc_bits-2], 2, 2, 0);
-        av_log(q->avctx,AV_LOG_DEBUG,"Joint-stereo VLC used.\n");
+        av_log(NULL,AV_LOG_DEBUG,"Joint-stereo VLC used.\n");
     }
 
-    av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
+    av_log(NULL,AV_LOG_ERROR,"VLC tables initialized. Result = %d\n",result);
     return result;
 }
 
@@ -249,8 +267,8 @@ static av_cold int init_cook_mlt(COOKContext *q) {
       av_free(q->mlt_window);
       return -1;
     }
-    av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
-           av_log2(mlt_size)+1);
+    av_log(NULL,AV_LOG_ERROR,"MDCT initialized, order = %d. mlt_window = %d\n",
+           av_log2(mlt_size)+1,sizeof(q->mlt_window)*mlt_size);
 
     return 0;
 }
@@ -317,11 +335,11 @@ static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes)
  * Cook uninit
  */
 
-static av_cold int cook_decode_close(AVCodecContext *avctx)
+static av_cold int cook_decode_close(COOKContext *q)
 {
     int i;
-    COOKContext *q = avctx->priv_data;
-    av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
+    //COOKContext *q = avctx->priv_data;
+    av_log(NULL,AV_LOG_ERROR, "Deallocating memory.\n");
 
     /* Free allocated memory buffers. */
     av_free(q->mlt_window);
@@ -341,7 +359,7 @@ static av_cold int cook_decode_close(AVCodecContext *avctx)
         free_vlc(&q->ccpl);
     }
 
-    av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
+    av_log(NULL,AV_LOG_ERROR,"Memory deallocated.\n");
 
     return 0;
 }
@@ -864,6 +882,8 @@ static void joint_decode(COOKContext *q, float* mlt_buffer1,
  * @param gain_ptr          array of current/prev gain pointers
  */
 
+#define FFSWAP(type,a,b) do{type SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0)
+
 static inline void
 decode_bytes_and_gain(COOKContext *q, const uint8_t *inbuffer,
                       cook_gains *gains_ptr)
@@ -938,9 +958,9 @@ static int decode_subpacket(COOKContext *q, const uint8_t *inbuffer,
                             int sub_packet_size, int16_t *outbuffer) {
     /* packet dump */
 //    for (i=0 ; i<sub_packet_size ; i++) {
-//        av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
+//        av_log(NULL, AV_LOG_ERROR, "%02x", inbuffer[i]);
 //    }
-//    av_log(q->avctx, AV_LOG_ERROR, "\n");
+//    av_log(NULL, AV_LOG_ERROR, "\n");
 
     decode_bytes_and_gain(q, inbuffer, &q->gains1);
 
@@ -974,37 +994,37 @@ static int decode_subpacket(COOKContext *q, const uint8_t *inbuffer,
 /**
  * Cook frame decoding
  *
- * @param avctx     pointer to the AVCodecContext
+ * @param rmctx     pointer to the RMContext
  */
 
-static int cook_decode_frame(AVCodecContext *avctx,
-            void *data, int *data_size,
-            const uint8_t *buf, int buf_size) {
-    COOKContext *q = avctx->priv_data;
+static int cook_decode_frame(RMContext *rmctx,COOKContext *q,
+            int16_t *outbuffer, int *data_size,
+            const uint8_t *inbuffer, int buf_size) {
+    //COOKContext *q = avctx->priv_data;
+    //COOKContext *q;
 
-    if (buf_size < avctx->block_align)
+    if (buf_size < rmctx->block_align)
         return buf_size;
 
-    *data_size = decode_subpacket(q, buf, avctx->block_align, data);
+    *data_size = decode_subpacket(q, inbuffer, rmctx->block_align, outbuffer);
 
     /* Discard the first two frames: no valid audio. */
-    if (avctx->frame_number < 2) *data_size = 0;
+    if (rmctx->frame_number < 2) *data_size = 0;
 
-    return avctx->block_align;
+    return rmctx->block_align;
 }
 
 #ifdef COOKDEBUG
 static void dump_cook_context(COOKContext *q)
 {
     //int i=0;
-#define PRINT(a,b) av_log(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
-    av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
-    av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->cookversion);
+#define PRINT(a,b) av_log(NULL,AV_LOG_ERROR," %s = %d\n", a, b);
+    av_log(NULL,AV_LOG_ERROR,"COOKextradata\n");
+    av_log(NULL,AV_LOG_ERROR,"cookversion=%x\n",q->cookversion);
     if (q->cookversion > STEREO) {
         PRINT("js_subband_start",q->js_subband_start);
         PRINT("js_vlc_bits",q->js_vlc_bits);
     }
-    av_log(q->avctx,AV_LOG_ERROR,"COOKContext\n");
     PRINT("nb_channels",q->nb_channels);
     PRINT("bit_rate",q->bit_rate);
     PRINT("sample_rate",q->sample_rate);
@@ -1031,76 +1051,60 @@ static av_cold int cook_count_channels(unsigned int mask){
 
 /**
  * Cook initialization
- *
- * @param avctx     pointer to the AVCodecContext
  */
 
-static av_cold int cook_decode_init(AVCodecContext *avctx)
-{
-    COOKContext *q = avctx->priv_data;
-    const uint8_t *edata_ptr = avctx->extradata;
-    q->avctx = avctx;
-
-    /* Take care of the codec specific extradata. */
-    if (avctx->extradata_size <= 0) {
-        av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
-        return -1;
-    } else {
-        /* 8 for mono, 16 for stereo, ? for multichannel
-           Swap to right endianness so we don't need to care later on. */
-        av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
-        if (avctx->extradata_size >= 8){
-            q->cookversion = bytestream_get_be32(&edata_ptr);
-            q->samples_per_frame =  bytestream_get_be16(&edata_ptr);
-            q->subbands = bytestream_get_be16(&edata_ptr);
-        }
-        if (avctx->extradata_size >= 16){
-            bytestream_get_be32(&edata_ptr);    //Unknown unused
-            q->js_subband_start = bytestream_get_be16(&edata_ptr);
-            q->js_vlc_bits = bytestream_get_be16(&edata_ptr);
-        }
+static av_cold int cook_decode_init(RMContext *rmctx, COOKContext *q)
+{   
+    /* cook extradata */
+    q->cookversion = rmctx->cook_version;
+    q->samples_per_frame =  rmctx->samples_pf_pc;
+    q->subbands = rmctx->nb_subbands;
+    q->extradata_size = rmctx->extradata_size;
+    if (q->extradata_size >= 16){
+        q->js_subband_start = rmctx->js_subband_start;
+        q->js_vlc_bits = rmctx->js_vlc_bits;
     }
 
-    /* Take data from the AVCodecContext (RM container). */
-    q->sample_rate = avctx->sample_rate;
-    q->nb_channels = avctx->channels;
-    q->bit_rate = avctx->bit_rate;
+    /* Take data from the RMContext (RM container). */
+    q->sample_rate = rmctx->sample_rate;
+    q->nb_channels = rmctx->nb_channels;
+    q->bit_rate = rmctx->bit_rate;  
 
     /* Initialize RNG. */
-    av_lfg_init(&q->random_state, ff_random_get_seed());
+    av_lfg_init(&q->random_state, 1);
 
     /* Initialize extradata related variables. */
     q->samples_per_channel = q->samples_per_frame / q->nb_channels;
-    q->bits_per_subpacket = avctx->block_align * 8;
+    q->bits_per_subpacket = rmctx->block_align * 8;
 
     /* Initialize default data states. */
     q->log2_numvector_size = 5;
     q->total_subbands = q->subbands;
 
     /* Initialize version-dependent variables */
-    av_log(avctx,AV_LOG_DEBUG,"q->cookversion=%x\n",q->cookversion);
+    av_log(NULL,AV_LOG_DEBUG,"q->cookversion=%x\n",q->cookversion);
     q->joint_stereo = 0;
     switch (q->cookversion) {
         case MONO:
             if (q->nb_channels != 1) {
-                av_log(avctx,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
+                av_log(NULL,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
                 return -1;
             }
-            av_log(avctx,AV_LOG_DEBUG,"MONO\n");
+            av_log(NULL,AV_LOG_DEBUG,"MONO\n");
             break;
         case STEREO:
             if (q->nb_channels != 1) {
                 q->bits_per_subpacket = q->bits_per_subpacket/2;
             }
-            av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
+            av_log(NULL,AV_LOG_DEBUG,"STEREO\n");
             break;
         case JOINT_STEREO:
             if (q->nb_channels != 2) {
-                av_log(avctx,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
+                av_log(NULL,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
                 return -1;
             }
-            av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
-            if (avctx->extradata_size >= 16){
+            av_log(NULL,AV_LOG_ERROR,"JOINT_STEREO\n");
+            if (q->extradata_size >= 16){
                 q->total_subbands = q->subbands + q->js_subband_start;
                 q->joint_stereo = 1;
             }
@@ -1112,11 +1116,11 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
             }
             break;
         case MC_COOK:
-            av_log(avctx,AV_LOG_ERROR,"MC_COOK not supported!\n");
+            av_log(NULL,AV_LOG_ERROR,"MC_COOK not supported!\n");
             return -1;
             break;
         default:
-            av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
+            av_log(NULL,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
             return -1;
             break;
     }
@@ -1133,22 +1137,24 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         return -1;
 
 
-    if(avctx->block_align >= UINT_MAX/2)
+    if(q->block_align >= UINT_MAX/2)
         return -1;
 
     /* Pad the databuffer with:
        DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
-       FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
+       INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
+
+#define INPUT_BUFFER_PADDING_SIZE 8
     if (q->nb_channels==2 && q->joint_stereo==0) {
         q->decoded_bytes_buffer =
-          av_mallocz(avctx->block_align/2
-                     + DECODE_BYTES_PAD2(avctx->block_align/2)
-                     + FF_INPUT_BUFFER_PADDING_SIZE);
+          av_mallocz(rmctx->block_align/2
+                     + DECODE_BYTES_PAD2(q->block_align/2)
+                     + INPUT_BUFFER_PADDING_SIZE);
     } else {
         q->decoded_bytes_buffer =
-          av_mallocz(avctx->block_align
-                     + DECODE_BYTES_PAD1(avctx->block_align)
-                     + FF_INPUT_BUFFER_PADDING_SIZE);
+          av_mallocz(rmctx->block_align
+                     + DECODE_BYTES_PAD1(q->block_align)
+                     + INPUT_BUFFER_PADDING_SIZE);
     }
     if (q->decoded_bytes_buffer == NULL)
         return -1;
@@ -1173,25 +1179,23 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
 
     /* Try to catch some obviously faulty streams, othervise it might be exploitable */
     if (q->total_subbands > 53) {
-        av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
+        av_log(NULL,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
         return -1;
     }
     if (q->subbands > 50) {
-        av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n");
+        av_log(NULL,AV_LOG_ERROR,"subbands > 50, report sample!\n");
         return -1;
     }
     if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
     } else {
-        av_log(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
+        av_log(NULL,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
         return -1;
     }
     if ((q->js_vlc_bits > 6) || (q->js_vlc_bits < 0)) {
-        av_log(avctx,AV_LOG_ERROR,"q->js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->js_vlc_bits);
+        av_log(NULL,AV_LOG_ERROR,"q->js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->js_vlc_bits);
         return -1;
     }
 
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
 
 #ifdef COOKDEBUG
     dump_cook_context(q);
@@ -1200,14 +1204,86 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
 }
 
 
-AVCodec cook_decoder =
+int main(int argc, char *argv[])
 {
-    .name = "cook",
-    .type = CODEC_TYPE_AUDIO,
-    .id = CODEC_ID_COOK,
-    .priv_data_size = sizeof(COOKContext),
-    .init = cook_decode_init,
-    .close = cook_decode_close,
-    .decode = cook_decode_frame,
-    .long_name = NULL_IF_CONFIG_SMALL("COOK"),
-};
+    int fd, fd_dec;
+    int res, datasize,x,i;
+    int nb_frames = 0;
+#ifdef DUMP_RAW_FRAMES 
+    char filename[15];
+    int fd_out;
+#endif
+    int16_t outbuf[2048];
+    uint8_t inbuf[1024];
+    uint16_t fs,sps,h;
+    uint32_t packet_count;
+    COOKContext q;
+    RMContext rmctx;
+    RMPacket pkt;
+
+    memset(&q,0,sizeof(COOKContext));
+    memset(&rmctx,0,sizeof(RMContext));
+    memset(&pkt,0,sizeof(RMPacket));
+
+    if (argc != 2) {
+        av_log(NULL,AV_LOG_ERROR,"Incorrect number of arguments\n");
+        return -1;
+    }
+
+    fd = open(argv[1],O_RDONLY);
+    if (fd < 0) {
+        av_log(NULL,AV_LOG_ERROR,"Error opening file %s\n", argv[1]);
+        return -1;
+    }
+    
+    fd_dec = open_wav("output.wav");
+    if (fd_dec < 0) {
+        av_log(NULL,AV_LOG_ERROR,"Error creating output file\n");
+        return -1;
+    }
+    res = real_parse_header(fd, &rmctx);
+    packet_count = rmctx.nb_packets;
+    rmctx.audio_framesize = rmctx.block_align;
+    rmctx.block_align = rmctx.sub_packet_size;
+    fs = rmctx.audio_framesize;
+    sps= rmctx.block_align;
+    h = rmctx.sub_packet_h;
+    cook_decode_init(&rmctx,&q);
+    av_log(NULL,AV_LOG_ERROR,"nb_frames = %d\n",nb_frames);
+    x = 0;
+    if(packet_count % h)
+    {
+        packet_count += h - (packet_count % h);
+        rmctx.nb_packets = packet_count;
+    }
+    while(packet_count)
+    {  
+        
+        memset(pkt.data,0,sizeof(pkt.data));
+        rm_get_packet(fd, &rmctx, &pkt);
+        DEBUGF("total frames = %d packet count = %d output counter = %d \n",rmctx.audio_pkt_cnt*(fs/sps), packet_count,rmctx.audio_pkt_cnt);
+        for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
+        { 
+            /* output raw audio frames that are sent to the decoder into separate files */
+            #ifdef DUMP_RAW_FRAMES 
+              snprintf(filename,sizeof(filename),"dump%d.raw",++x);
+              fd_out = open(filename,O_WRONLY|O_CREAT|O_APPEND);           
+              write(fd_out,pkt.data+i*sps,sps);  
+              close(fd_out);
+            #endif
+
+            memcpy(inbuf,pkt.data+i*sps,sps);
+            nb_frames = cook_decode_frame(&rmctx,&q, outbuf, &datasize, inbuf , rmctx.block_align);
+            rmctx.frame_number++;
+            write(fd_dec,outbuf,datasize);        
+        }
+        packet_count -= rmctx.audio_pkt_cnt;
+        rmctx.audio_pkt_cnt = 0;
+    } 
+    cook_decode_close(&q);
+    close_wav(fd_dec,&rmctx);
+    close(fd);
+
+
+  return 0;
+}
diff --git a/apps/codecs/libcook/dsputil.h b/apps/codecs/libcook/dsputil.h
index e9c2bfc77e..4573c17a62 100644
--- a/apps/codecs/libcook/dsputil.h
+++ b/apps/codecs/libcook/dsputil.h
@@ -36,6 +36,7 @@
 
 //#define DEBUG
 /* dct code */
+#if 0 /*MT : DELETE THIS LINE.*/
 typedef short DCTELEM;
 typedef int DWTELEM;
 typedef short IDWTELEM;
@@ -590,8 +591,10 @@ void dsputil_init_ppc(DSPContext* c, AVCodecContext *avctx);
 void dsputil_init_sh4(DSPContext* c, AVCodecContext *avctx);
 void dsputil_init_vis(DSPContext* c, AVCodecContext *avctx);
 
+#endif /*MT : DELETE THIS LINE ONLY. */
 #define DECLARE_ALIGNED_16(t, v) DECLARE_ALIGNED(16, t, v)
 
+#if 0 /*MT : DELETE THIS LINE ONLY. */
 #if HAVE_MMX
 
 #undef emms_c
@@ -644,10 +647,12 @@ extern int mm_flags;
 
 #endif
 
+#endif /* MT : DELETE THIS LINE ONLY */
 #ifndef DECLARE_ALIGNED_8
 #   define DECLARE_ALIGNED_8(t, v) DECLARE_ALIGNED(8, t, v)
 #endif
 
+#if 0 /* MT : DELETE THIS LINE ONLY */
 #ifndef STRIDE_ALIGN
 #   define STRIDE_ALIGN 8
 #endif
@@ -657,6 +662,7 @@ void get_psnr(uint8_t *orig_image[3], uint8_t *coded_image[3],
               int orig_linesize[3], int coded_linesize,
               AVCodecContext *avctx);
 
+#endif /*MT : DELETE THIS LINE.*/
 /* FFT computation */
 
 /* NOTE: soon integer code will be added, so you must use the
@@ -715,6 +721,7 @@ static inline void ff_fft_calc(FFTContext *s, FFTComplex *z)
 }
 void ff_fft_end(FFTContext *s);
 
+#endif /*MT : DELETE THIS LINE.*/
 /* MDCT computation */
 
 typedef struct MDCTContext {
@@ -735,6 +742,7 @@ static inline void ff_imdct_half(MDCTContext *s, FFTSample *output, const FFTSam
     s->fft.imdct_half(s, output, input);
 }
 
+#if 0 /* MT : DELETE THIS LINE. */
 /**
  * Generate a Kaiser-Bessel Derived Window.
  * @param   window  pointer to half window
@@ -742,6 +750,7 @@ static inline void ff_imdct_half(MDCTContext *s, FFTSample *output, const FFTSam
  * @param   n       size of half window
  */
 void ff_kbd_window_init(float *window, float alpha, int n);
+#endif /* MT : DELETE THIS LINE.*/
 
 /**
  * Generate a sine window.
@@ -769,6 +778,7 @@ void ff_imdct_half_sse(MDCTContext *s, FFTSample *output, const FFTSample *input
 void ff_mdct_calc(MDCTContext *s, FFTSample *out, const FFTSample *input);
 void ff_mdct_end(MDCTContext *s);
 
+#if 0 /* MT : DELETE THIS LINE.*/
 /* Real Discrete Fourier Transform */
 
 enum RDFTransformType {
diff --git a/apps/codecs/libcook/ffmpeg_config.h b/apps/codecs/libcook/ffmpeg_config.h
new file mode 100644
index 0000000000..707e14df52
--- /dev/null
+++ b/apps/codecs/libcook/ffmpeg_config.h
@@ -0,0 +1,14 @@
+/* Automatically generated by configure - do not modify */
+#ifndef _FFMPEG_CONFIG_H
+#define _FFMPEG_CONFIG_H
+// CHECK THIS : #include "codecs.h"
+
+#ifdef CPU_ARM
+#define CONFIG_ALIGN 1
+#endif
+
+#ifdef ROCKBOX_BIG_ENDIAN
+#define WORDS_BIGENDIAN
+#endif
+
+#endif
diff --git a/apps/codecs/libcook/libavutil/avutil.h b/apps/codecs/libcook/libavutil/avutil.h
index c57e69f10c..c07e44d660 100644
--- a/apps/codecs/libcook/libavutil/avutil.h
+++ b/apps/codecs/libcook/libavutil/avutil.h
@@ -54,10 +54,10 @@
 unsigned avutil_version(void);
 
 #include "common.h"
-#include "mathematics.h"
-#include "rational.h"
-#include "intfloat_readwrite.h"
+//#include "mathematics.h"
+//#include "rational.h"
+//#include "intfloat_readwrite.h"
 #include "log.h"
-#include "pixfmt.h"
+//#include "pixfmt.h"
 
 #endif /* AVUTIL_AVUTIL_H */
diff --git a/apps/codecs/libcook/libavutil/bswap.h b/apps/codecs/libcook/libavutil/bswap.h
index cf68c43c72..9175cb24a5 100644
--- a/apps/codecs/libcook/libavutil/bswap.h
+++ b/apps/codecs/libcook/libavutil/bswap.h
@@ -27,7 +27,7 @@
 #define AVUTIL_BSWAP_H
 
 #include <stdint.h>
-#include "config.h"
+//#include "ffmpeg_config.h"
 #include "common.h"
 
 #if   ARCH_ARM
diff --git a/apps/codecs/libcook/libavutil/common.h b/apps/codecs/libcook/libavutil/common.h
index 15eaf9849d..949f093d35 100644
--- a/apps/codecs/libcook/libavutil/common.h
+++ b/apps/codecs/libcook/libavutil/common.h
@@ -278,9 +278,9 @@ static inline av_const float av_clipf(float a, float amin, float amax)
 
 #include "mem.h"
 
-#ifdef HAVE_AV_CONFIG_H
-#    include "config.h"
+//#ifdef HAVE_AV_CONFIG_H
+//#    include "ffmpeg_config.h"
 #    include "internal.h"
-#endif /* HAVE_AV_CONFIG_H */
+//#endif /* HAVE_AV_CONFIG_H */
 
 #endif /* AVUTIL_COMMON_H */
diff --git a/apps/codecs/libcook/libavutil/internal.h b/apps/codecs/libcook/libavutil/internal.h
index 4191aa8e52..f8f9418a76 100644
--- a/apps/codecs/libcook/libavutil/internal.h
+++ b/apps/codecs/libcook/libavutil/internal.h
@@ -34,10 +34,10 @@
 #include <stdint.h>
 #include <stddef.h>
 #include <assert.h>
-#include "config.h"
+//#include "ffmpeg_config.h"
 #include "common.h"
 #include "mem.h"
-#include "timer.h"
+//#include "timer.h"
 
 #ifndef attribute_align_arg
 #if (!defined(__ICC) || __ICC > 1100) && AV_GCC_VERSION_AT_LEAST(4,2)
@@ -223,10 +223,10 @@ if((y)<(x)){\
 #define realloc please_use_av_realloc
 #undef  time
 #define time time_is_forbidden_due_to_security_issues
-#undef  rand
-#define rand rand_is_forbidden_due_to_state_trashing_use_av_random
-#undef  srand
-#define srand srand_is_forbidden_due_to_state_trashing_use_av_random_init
+//#undef  rand
+//#define rand rand_is_forbidden_due_to_state_trashing_use_av_random
+//#undef  srand
+//#define srand srand_is_forbidden_due_to_state_trashing_use_av_random_init
 #undef  random
 #define random random_is_forbidden_due_to_state_trashing_use_av_random
 #undef  sprintf
diff --git a/apps/codecs/libcook/libavutil/intreadwrite.h b/apps/codecs/libcook/libavutil/intreadwrite.h
index 7c5909ea51..d27a50061e 100644
--- a/apps/codecs/libcook/libavutil/intreadwrite.h
+++ b/apps/codecs/libcook/libavutil/intreadwrite.h
@@ -20,7 +20,7 @@
 #define AVUTIL_INTREADWRITE_H
 
 #include <stdint.h>
-#include "config.h"
+//#include "ffmpeg_config.h"
 #include "bswap.h"
 
 #ifdef __GNUC__
diff --git a/apps/codecs/libcook/libavutil/mem.c b/apps/codecs/libcook/libavutil/mem.c
index 741450b53f..7307df2384 100644
--- a/apps/codecs/libcook/libavutil/mem.c
+++ b/apps/codecs/libcook/libavutil/mem.c
@@ -24,7 +24,6 @@
  * default memory allocator for libavutil
  */
 
-#include "config.h"
 
 #include <limits.h>
 #include <stdlib.h>
diff --git a/apps/codecs/libcook/rm2wav.c b/apps/codecs/libcook/rm2wav.c
new file mode 100644
index 0000000000..4ef1ec4f66
--- /dev/null
+++ b/apps/codecs/libcook/rm2wav.c
@@ -0,0 +1,584 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2008 Dave Chapman
+ * Copyright (C) 2009 Mohamed Tarek
+ *
+ * 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 <stdio.h>
+#include <string.h>
+#include <stdint.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <unistd.h>
+
+#include "rm2wav.h"
+
+#define MAX_PATH 260
+#define PKT_HEADER_SIZE 12
+#define RAW_AUDIO_DATA packet_length-PKT_HEADER_SIZE
+
+
+#if 0
+#define DEBUG 
+#define DEBUGF printf
+#else
+#define DEBUGF(...)
+#endif
+
+/* ASF codec IDs */
+#define CODEC_ID_WMAV1 0x160
+#define CODEC_ID_WMAV2 0x161
+
+static unsigned char wav_header[44]={
+    'R','I','F','F',//  0 - ChunkID
+    0,0,0,0,        //  4 - ChunkSize (filesize-8)
+    'W','A','V','E',//  8 - Format
+    'f','m','t',' ',// 12 - SubChunkID
+    16,0,0,0,       // 16 - SubChunk1ID  // 16 for PCM
+    1,0,            // 20 - AudioFormat (1=Uncompressed)
+    2,0,            // 22 - NumChannels
+    0,0,0,0,        // 24 - SampleRate in Hz
+    0,0,0,0,        // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
+    4,0,            // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
+    16,0,           // 34 - BitsPerSample
+    'd','a','t','a',// 36 - Subchunk2ID
+    0,0,0,0         // 40 - Subchunk2Size
+};
+
+int open_wav(char* filename) {
+    int fd,res;
+
+    fd=open(filename,O_CREAT|O_WRONLY|O_TRUNC,S_IRUSR|S_IWUSR);
+    if (fd >= 0) {
+        res = write(fd,wav_header,sizeof(wav_header));
+    }
+
+    return(fd);
+}
+
+void close_wav(int fd, RMContext *rmctx) {
+    int x,res;
+    int filesize;
+    int bytes_per_sample = 2;
+    int samples_per_frame = rmctx->samples_pf_pc;
+    int nb_channels = rmctx->nb_channels;
+    int sample_rate = rmctx->sample_rate;
+    int nb_frames = rmctx->audio_framesize/rmctx->block_align * rmctx->nb_packets - 2; // first 2 frames have no valid audio; skipped in output
+
+    filesize= samples_per_frame*bytes_per_sample*nb_frames +44;
+    printf("Filesize = %d\n",filesize);
+
+    // ChunkSize
+    x=filesize-8;
+    wav_header[4]=(x&0xff);
+    wav_header[5]=(x&0xff00)>>8;
+    wav_header[6]=(x&0xff0000)>>16;
+    wav_header[7]=(x&0xff000000)>>24;
+
+    // Number of channels
+    wav_header[22]=nb_channels;
+
+    // Samplerate
+    wav_header[24]=sample_rate&0xff;
+    wav_header[25]=(sample_rate&0xff00)>>8;
+    wav_header[26]=(sample_rate&0xff0000)>>16;
+    wav_header[27]=(sample_rate&0xff000000)>>24;
+
+    // ByteRate
+    x=sample_rate*bytes_per_sample*nb_channels;
+    wav_header[28]=(x&0xff);
+    wav_header[29]=(x&0xff00)>>8;
+    wav_header[30]=(x&0xff0000)>>16;
+    wav_header[31]=(x&0xff000000)>>24;
+
+    // BlockAlign
+    wav_header[32]=rmctx->block_align;//2*rmctx->nb_channels;
+
+    // Bits per sample
+    wav_header[34]=16;
+    
+    // Subchunk2Size
+    x=filesize-44;
+    wav_header[40]=(x&0xff);
+    wav_header[41]=(x&0xff00)>>8;
+    wav_header[42]=(x&0xff0000)>>16;
+    wav_header[43]=(x&0xff000000)>>24;
+
+    lseek(fd,0,SEEK_SET);
+    res = write(fd,wav_header,sizeof(wav_header));
+    close(fd);
+}
+           
+/* Some Rockbox-like functions (these should be implemented in metadata_common.[ch] */
+struct cook_extradata {
+    uint32_t cook_version;
+    uint16_t samples_pf_pc;    /* samples per frame per channel */
+    uint16_t nb_subbands;      /* number of subbands in the frequency domain */
+
+    /* extra 8 bytes for stereo data */
+    uint32_t unused;
+    uint16_t js_subband_start; /* joint stereo subband start */
+    uint16_t js_vlc_bits;
+};
+
+static int read_uint8(int fd, uint8_t* buf)
+{
+    unsigned char tmp[1];
+    int res;
+
+    res=read(fd, tmp, 1);
+    *buf = tmp[0];
+    return res;
+}
+
+static int read_uint16be(int fd, uint16_t* buf)
+{
+    unsigned char tmp[2];
+    int res;
+
+    res=read(fd, tmp, 2);
+    *buf = (tmp[0] << 8) | tmp[1];
+    return res;
+}
+
+static int read_uint32be(int fd, uint32_t* buf)
+{
+    unsigned char tmp[4];
+    int res;
+
+    res=read(fd, tmp, 4);
+    *buf = (tmp[0] << 24) | (tmp[1] << 16) | (tmp[2] << 8) | tmp[3];
+    return res;
+}
+
+off_t filesize(int fd)
+{
+  struct stat buf;
+
+  if (fstat(fd,&buf) == -1) {
+    return -1;
+  } else {
+    return buf.st_size;
+  }
+}
+
+int read_cook_extradata(int fd, RMContext *rmctx) {
+    read_uint32be(fd, &rmctx->cook_version);
+    read_uint16be(fd, &rmctx->samples_pf_pc);
+    read_uint16be(fd, &rmctx->nb_subbands);
+    if(rmctx->extradata_size == 16) {
+        read_uint32be(fd, &rmctx->unused);
+        read_uint16be(fd, &rmctx->js_subband_start);
+        read_uint16be(fd, &rmctx->js_vlc_bits);
+    }
+    return rmctx->extradata_size; /* for 'skipped' */
+}
+
+void print_cook_extradata(RMContext *rmctx) {
+
+    printf("            cook_version = 0x%08x\n", rmctx->cook_version);
+    printf("            samples_per_frame_per_channel = %d\n", rmctx->samples_pf_pc);
+    printf("            number_of_subbands_in_freq_domain = %d\n", rmctx->nb_subbands);
+    if(rmctx->extradata_size == 16) {
+        printf("            joint_stereo_subband_start = %d\n",rmctx->js_subband_start);
+        printf("            joint_stereo_vlc_bits = %d\n", rmctx->js_vlc_bits);
+    }
+} 
+
+
+struct real_object_t
+{
+    uint32_t fourcc;
+    uint32_t size;
+    uint16_t version;
+};
+
+#define FOURCC(a,b,c,d) (((a)<<24) | ((b) << 16) | ((c) << 8) | (d))
+
+static int real_read_object_header(int fd, struct real_object_t* obj)
+{
+    int n;
+
+    if ((n = read_uint32be(fd, &obj->fourcc)) <= 0) return n;
+    if ((n = read_uint32be(fd, &obj->size)) <= 0) return n;
+    if ((n = read_uint16be(fd, &obj->version)) <= 0) return n;
+
+    return 1;
+}
+
+static char* fourcc2str(uint32_t f)
+{
+    static char res[5];
+
+    res[0] = (f & 0xff000000) >> 24;
+    res[1] = (f & 0xff0000) >> 16;
+    res[2] = (f & 0xff00) >> 8;
+    res[3] = (f & 0xff);
+    res[4] = 0;
+
+    return res;
+}
+
+static int read_str(int fd, char* buf)
+{
+    uint8_t len;
+    int res;
+
+    res = read(fd, &len, 1);
+    res = read(fd, buf, len);
+    buf[len]=0;
+
+    return len+1;
+}
+
+static int real_read_audio_stream_info(int fd, RMContext *rmctx)
+{
+    int skipped = 0;
+    uint32_t version;
+    struct real_object_t obj;
+    memset(&obj,0,sizeof(obj));
+    uint32_t header_size;
+    uint16_t flavor;
+    uint32_t coded_framesize;
+    uint32_t unknown1;
+    uint32_t unknown2;
+    uint32_t unknown3;
+    uint16_t unknown4;
+    uint16_t unknown5;
+    uint16_t unknown6;
+    uint16_t unknown7;
+    uint32_t unknown8;
+    uint8_t  interleaver_id_length;
+    uint32_t interleaver_id;
+    uint8_t  fourcc_length;
+    uint32_t fourcc = 0;
+    uint8_t  unknown9;
+    uint16_t unknown10;
+    uint8_t  unknown11;
+
+    read_uint32be(fd, &version);
+    skipped += 4;
+
+    printf("    version=0x%04x\n",((version >> 16) & 0xff));
+    if (((version >> 16) & 0xff) == 3) {
+        /* Very old version */
+    } else {
+       real_read_object_header(fd, &obj);
+       skipped += 10;
+       read_uint32be(fd, &header_size);
+       skipped += 4;
+       /* obj.size will be filled with an unknown value, replaced with header_size */
+       printf("    Object: %s, size: %d bytes, version: 0x%04x\n",fourcc2str(obj.fourcc),header_size,obj.version);
+
+       read_uint16be(fd, &flavor);
+       read_uint32be(fd, &coded_framesize);
+       read_uint32be(fd, &unknown1);
+       read_uint32be(fd, &unknown2);
+       read_uint32be(fd, &unknown3);
+       read_uint16be(fd, &rmctx->sub_packet_h);
+       read_uint16be(fd, &rmctx->block_align);
+       read_uint16be(fd, &rmctx->sub_packet_size);
+       read_uint16be(fd, &unknown4);
+       skipped += 26;
+       if (((version >> 16) & 0xff) == 5)
+       {
+           read_uint16be(fd, &unknown5);
+           read_uint16be(fd, &unknown6);
+           read_uint16be(fd, &unknown7);
+           skipped += 6;
+       }
+       read_uint16be(fd, &rmctx->sample_rate);
+       read_uint32be(fd, &unknown8);
+       read_uint16be(fd, &rmctx->nb_channels);
+       skipped += 8;
+       if (((version >> 16) & 0xff) == 4)
+       {
+           read_uint8(fd, &interleaver_id_length);
+           read_uint32be(fd, &interleaver_id);
+           read_uint8(fd, &fourcc_length);
+           read_uint32be(fd, &fourcc);
+           skipped += 10;
+       }
+       if (((version >> 16) & 0xff) == 5)
+       {
+           read_uint32be(fd, &interleaver_id);
+           read_uint32be(fd, &fourcc);
+           skipped += 8;
+       }
+       read_uint8(fd,&unknown9);
+       read_uint16be(fd,&unknown10);
+       skipped += 3;
+       if (((version >> 16) & 0xff) == 5)
+       {
+           read_uint8(fd, &unknown11);
+           skipped += 1;
+       }
+  
+       read_uint32be(fd, &rmctx->extradata_size);
+       skipped += 4;
+       if(!strncmp(fourcc2str(fourcc),"cook",4))
+           skipped += read_cook_extradata(fd, rmctx); 
+       
+       
+       printf("        flavor = %d\n",flavor);
+       printf("        coded_frame_size = %d\n",coded_framesize);
+       printf("        sub_packet_h = %d\n",rmctx->sub_packet_h);
+       printf("        frame_size = %d\n",rmctx->block_align);
+       printf("        sub_packet_size = %d\n",rmctx->sub_packet_size);
+       printf("        sample_rate= %d\n",rmctx->sample_rate);
+       printf("        channels= %d\n",rmctx->nb_channels);
+       printf("        fourcc = %s\n",fourcc2str(fourcc));
+       printf("        codec_extra_data_length = %d\n",rmctx->extradata_size);
+       printf("        codec_extradata :\n");
+       print_cook_extradata(rmctx);
+
+    }
+
+    return skipped;
+}
+
+int real_parse_header(int fd, RMContext *rmctx)
+{
+    struct real_object_t obj;
+    memset(&obj,0,sizeof(obj));
+    int res;
+    int skipped;
+    off_t curpos;
+
+    uint32_t unknown1;
+    uint32_t unknown2;
+
+    uint32_t max_bitrate;
+    uint32_t avg_bitrate = 0;
+    uint32_t max_packet_size;
+    uint32_t avg_packet_size;
+    uint32_t packet_count;
+    uint32_t duration;
+    uint32_t preroll;
+    uint32_t index_offset;
+    uint32_t data_offset;
+    uint16_t num_streams;
+    uint16_t flags = 0;
+
+    uint16_t stream_id;
+    uint32_t start_time;
+    char desc[256];
+    char mimetype[256];
+    uint32_t codec_data_size;
+    uint32_t v;
+
+    char title[256];
+    char author[256];
+    char copyright[256];
+    char comment[256];
+
+    uint32_t next_data_off;
+    uint8_t  header_end;
+
+    curpos = lseek(fd, 0, SEEK_SET);    
+    res = real_read_object_header(fd, &obj);
+
+    if (obj.fourcc == FOURCC('.','r','a',0xfd))
+    {
+        /* Very old .ra format - not yet supported */
+        return -1;
+    } 
+    else if (obj.fourcc != FOURCC('.','R','M','F'))
+    {
+        return -1;
+    }
+
+    read_uint32be(fd, &unknown1);
+    read_uint32be(fd, &unknown2);
+
+    printf("Object: %s, size: %d bytes, version: 0x%04x, pos: %d\n",fourcc2str(obj.fourcc),(int)obj.size,obj.version,(int)curpos);
+    printf("    unknown1=%d (0x%08x)\n",unknown1,unknown1);
+    printf("    unknown2=%d (0x%08x)\n",unknown2,unknown2);
+
+    res = real_read_object_header(fd, &obj);
+    header_end = 0;
+    while(res)
+    {
+        printf("Object: %s, size: %d bytes, version: 0x%04x, pos: %d\n",fourcc2str(obj.fourcc),(int)obj.size,obj.version,(int)curpos);
+        skipped = 10;
+        if(obj.fourcc == FOURCC('I','N','D','X'))
+                break;
+        switch (obj.fourcc)
+        {
+            case FOURCC('P','R','O','P'): /* File properties */
+                read_uint32be(fd, &max_bitrate);
+                read_uint32be(fd, &rmctx->bit_rate); /*avg bitrate*/
+                read_uint32be(fd, &max_packet_size);
+                read_uint32be(fd, &avg_packet_size);
+                read_uint32be(fd, &packet_count);
+                read_uint32be(fd, &duration);
+                read_uint32be(fd, &preroll);
+                read_uint32be(fd, &index_offset);
+                read_uint32be(fd, &data_offset);
+                read_uint16be(fd, &num_streams);
+                read_uint16be(fd, &rmctx->flags);
+                skipped += 40;
+
+                printf("    max_bitrate = %d\n",max_bitrate);
+                printf("    avg_bitrate = %d\n",avg_bitrate);
+                printf("    max_packet_size = %d\n",max_packet_size);
+                printf("    avg_packet_size = %d\n",avg_packet_size);
+                printf("    packet_count = %d\n",packet_count);
+                printf("    duration = %d\n",duration);
+                printf("    preroll = %d\n",preroll);
+                printf("    index_offset = %d\n",index_offset);
+                printf("    data_offset = %d\n",data_offset);
+                printf("    num_streams = %d\n",num_streams);
+                printf("    flags=0x%04x\n",flags);
+                break;
+
+            case FOURCC('C','O','N','T'):
+                /* Four strings - Title, Author, Copyright, Comment */
+                skipped += read_str(fd,title);
+                skipped += read_str(fd,author);
+                skipped += read_str(fd,copyright);
+                skipped += read_str(fd,comment);
+
+                printf("    title=\"%s\"\n",title);
+                printf("    author=\"%s\"\n",author);
+                printf("    copyright=\"%s\"\n",copyright);
+                printf("    comment=\"%s\"\n",comment);
+                break;
+
+            case FOURCC('M','D','P','R'): /* Media properties */
+                read_uint16be(fd,&stream_id);
+                skipped += 2;
+                read_uint32be(fd,&max_bitrate);
+                skipped += 4;
+                read_uint32be(fd,&avg_bitrate);
+                skipped += 4;
+                read_uint32be(fd,&max_packet_size);
+                skipped += 4;
+                read_uint32be(fd,&avg_packet_size);
+                skipped += 4;
+                read_uint32be(fd,&start_time);
+                skipped += 4;
+                read_uint32be(fd,&preroll);
+                skipped += 4;
+                read_uint32be(fd,&duration);
+                skipped += 4;
+                skipped += read_str(fd,desc);
+                skipped += read_str(fd,mimetype);
+                read_uint32be(fd,&codec_data_size);
+                skipped += 4;
+                //From ffmpeg: codec_pos = url_ftell(pb);
+                read_uint32be(fd,&v);
+                skipped += 4;
+
+                printf("    stream_id = 0x%04x\n",stream_id);
+                printf("    max_bitrate = %d\n",max_bitrate);
+                printf("    avg_bitrate = %d\n",avg_bitrate);
+                printf("    max_packet_size = %d\n",max_packet_size);
+                printf("    avg_packet_size = %d\n",avg_packet_size);
+                printf("    start_time = %d\n",start_time);
+                printf("    preroll = %d\n",preroll);
+                printf("    duration = %d\n",duration);
+                printf("    desc=\"%s\"\n",desc);
+                printf("    mimetype=\"%s\"\n",mimetype);
+                printf("    codec_data_size = %d\n",codec_data_size);
+                printf("    v=\"%s\"\n", fourcc2str(v));
+
+                if (v == FOURCC('.','r','a',0xfd))
+                {
+                    skipped += real_read_audio_stream_info(fd, rmctx);
+                }
+
+                break;
+
+            case FOURCC('D','A','T','A'):    
+                
+              read_uint32be(fd,&rmctx->nb_packets);
+              skipped += 4;
+              read_uint32be(fd,&next_data_off);
+              skipped += 4;
+              if (!rmctx->nb_packets && (rmctx->flags & 4))
+                  rmctx->nb_packets = 3600 * 25;                
+              printf("    data_nb_packets = %d\n",rmctx->nb_packets);
+              printf("    next DATA offset = %d\n",next_data_off);
+              header_end = 1;                   
+                break; 
+        }
+        if(header_end) break;
+        curpos = lseek(fd, obj.size - skipped, SEEK_CUR);
+        res = real_read_object_header(fd, &obj);
+    }
+
+
+    return 0;
+}
+
+void rm_get_packet(int fd,RMContext *rmctx, RMPacket *pkt)
+{   
+    uint8_t unknown,packet_group;
+    uint16_t x, place;
+    uint16_t sps = rmctx->sub_packet_size;
+    uint16_t h = rmctx->sub_packet_h;
+    uint16_t y = rmctx->sub_packet_cnt;
+    uint16_t w = rmctx->audio_framesize;
+    do
+    {
+        y = rmctx->sub_packet_cnt;
+        read_uint16be(fd,&pkt->version);
+        read_uint16be(fd,&pkt->length);
+        read_uint16be(fd,&pkt->stream_number);
+        read_uint32be(fd,&pkt->timestamp);
+        DEBUGF("    version = %d\n"
+               "    length  = %d\n"
+               "    stream  = %d\n"
+               "    timestmp= %d\n",pkt->version,pkt->length,pkt->stream_number,pkt->timestamp);
+
+        //getchar();
+        if(pkt->version == 0)
+        {
+            read_uint8(fd,&packet_group);
+            read_uint8(fd,&pkt->flags);
+        }
+        if(pkt->version == 1)
+            read_uint8(fd,&unknown);
+
+        if (pkt->flags & 2) /* keyframe */
+            y = rmctx->sub_packet_cnt = 0;
+        if (!y) /* if keyframe update playback elapsed time */
+            rmctx->audiotimestamp = pkt->timestamp;
+
+        for(x = 0 ; x < w/sps; x++)
+        {
+            place = sps*(h*x+((h+1)/2)*(y&1)+(y>>1)); 
+            read(fd,pkt->data+place, sps);
+            //DEBUGF("place = %d data[place] = %d\n",place,pkt->data[place]);
+        }
+        rmctx->audio_pkt_cnt++;
+    }while(++(rmctx->sub_packet_cnt) < h);
+
+    //return pkt->data;
+}
+#ifdef DEBUG
+void dump_rm_context(RMContext *rmctx)
+{
+    DEBUGF("block_align = %d\n", rmctx->block_align);
+    DEBUGF("nb_channels = %d\n", rmctx->nb_channels);
+    DEBUGF("sample_rate = %d\n", rmctx->sample_rate);
+    DEBUGF("bit_rate    = %d\n", rmctx->bit_rate   );
+}
+#endif
diff --git a/apps/codecs/libcook/rm2wav.h b/apps/codecs/libcook/rm2wav.h
new file mode 100644
index 0000000000..09b20c7fa6
--- /dev/null
+++ b/apps/codecs/libcook/rm2wav.h
@@ -0,0 +1,74 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2009 Mohamed Tarek
+ *
+ * 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 <stdio.h>
+#include <stdint.h>
+
+typedef struct rm_packet
+{
+    uint8_t  data[30000]; /* Reordered data. No malloc, hence the size */
+    uint16_t version;
+    uint16_t length;
+    uint32_t timestamp;
+    uint16_t stream_number;
+    uint8_t  flags;
+}RMPacket;
+    
+typedef struct rm_context 
+{
+    /* Demux Context */
+    int old_format;
+    int current_stream;
+    int remaining_len;
+    int audio_stream_num; /*< Stream number for audio packets*/
+    int audio_pkt_cnt; /* Output packet counter*/
+
+    /* Stream Variables */
+    uint32_t audiotimestamp; /* Audio packet timestamp*/
+    uint16_t sub_packet_cnt; /* Subpacket counter, used while reading */
+    uint16_t sub_packet_size, sub_packet_h, coded_framesize; /* Descrambling parameters from container */
+    uint16_t audio_framesize; /* Audio frame size from container */
+    uint16_t sub_packet_lengths[16]; /* Length of each subpacket */
+
+    /* Codec Context */
+    uint16_t block_align;
+    uint32_t nb_packets;
+    int frame_number;
+    uint32_t extradata_size;
+    uint16_t sample_rate;
+    uint16_t nb_channels;
+    uint32_t bit_rate;
+    uint16_t flags;
+
+    /*cook extradata*/
+    uint32_t cook_version;
+    uint16_t samples_pf_pc;    /* samples per frame per channel */
+    uint16_t nb_subbands;      /* number of subbands in the frequency domain */
+    /* extra 8 bytes for stereo data */
+    uint32_t unused;
+    uint16_t js_subband_start; /* joint stereo subband start */
+    uint16_t js_vlc_bits;    
+
+} RMContext;
+
+int open_wav(char* filename);
+void close_wav(int fd, RMContext *rmctx);
+int real_parse_header(int fd, RMContext *rmctx);
+void rm_get_packet(int fd,RMContext *rmctx, RMPacket *pkt);