SPC Codec: Refactor for CPU and clean up some things.

CPU optimization gets its own files in which to fill-in optimizable
routines.

Some pointless #if 0's for profiling need removal. Those macros are
empty if not profiling.

Force some functions that are undesirable to be force-inlined by the
compiler to be not inlined.

Change-Id: Ia7b7e45380d7efb20c9b1a4d52e05db3ef6bbaab

1 files changed, 211 insertions, 0 deletions

diff --git a/lib/rbcodec/codecs/libspc/spc_dsp_generic.c b/lib/rbcodec/codecs/libspc/spc_dsp_generic.c
new file mode 100644
index 0000000000..60e79f8763
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/spc_dsp_generic.c
@@ -0,0 +1,211 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2006-2007 Adam Gashlin (hcs)
+ * Copyright (C) 2004-2007 Shay Green (blargg)
+ * Copyright (C) 2002 Brad Martin
+ *
+ * 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.
+ *
+ ****************************************************************************/
+static inline int apply_gen_envx( struct voice_t* voice, int output )
+{
+    return (output * voice->envx) >> 11;
+}
+
+static inline int apply_gen_volume( struct voice_t* voice, int output,
+                                    int* amp_0, int* amp_1 )
+{
+    *amp_0 = voice->volume [0] * output;
+    *amp_1 = voice->volume [1] * output;
+    return output;
+}
+
+static inline int apply_gen_amp( struct voice_t* voice, int output,
+                                 int* amp_0, int* amp_1)
+{
+    output = apply_gen_envx( voice, output );
+    output = apply_gen_volume( voice, output, amp_0, amp_1 );
+    return output;
+}
+
+#if !SPC_NOINTERP
+
+#ifndef SPC_GAUSSIAN_FAST_INTERP
+static inline int gaussian_fast_interp( int16_t const* samples,
+                                        int32_t position,
+                                        int16_t const* fwd,
+                                        int16_t const* rev )
+{
+    samples += position >> 12;
+    return (fwd [0] * samples [0] +
+            fwd [1] * samples [1] +
+            rev [1] * samples [2] +
+            rev [0] * samples [3]) >> 11;
+}
+#endif /* SPC_GAUSSIAN_FAST_INTERP */
+
+#ifndef SPC_GAUSSIAN_FAST_AMP
+#define gaussian_fast_amp   apply_amp
+#endif /* SPC_GAUSSIAN_FAST_AMP */
+
+#ifndef SPC_GAUSSIAN_SLOW_INTERP
+static inline int gaussian_slow_interp( int16_t const* samples,
+                                        int32_t position,
+                                        int16_t const* fwd,
+                                        int16_t const* rev )
+{
+    int output;
+    samples += position >> 12;
+    output = (fwd [0] * samples [0]) & ~0xFFF;
+    output = (output + fwd [1] * samples [1]) & ~0xFFF;
+    output = (output + rev [1] * samples [2]) >> 12;
+    output = (int16_t) (output * 2);
+    output += ((rev [0] * samples [3]) >> 12) * 2;
+    return CLAMP16( output );
+}
+#endif /* SPC_GAUSSIAN_SLOW_INTERP */
+
+#ifndef SPC_GAUSSIAN_SLOW_AMP
+static inline int gaussian_slow_amp( struct voice_t* voice, int output,
+                                     int *amp_0, int *amp_1 )
+{
+    output = apply_gen_envx( voice, output ) & ~1;
+    output = apply_gen_volume( voice, output, amp_0, amp_1 );
+    return output;
+}
+#endif /* SPC_GAUSSIAN_SLOW_AMP */
+
+#define interp      gaussian_slow_interp
+#define apply_amp   gaussian_slow_amp
+
+#else /* SPC_NOINTERP */
+
+#ifndef SPC_LINEAR_INTERP
+static inline int linear_interp( int16_t const* samples, int32_t position )
+{
+    int32_t fraction = position & 0xfff;
+    int16_t const* pos = (samples + (position >> 12)) + 1;
+    return pos[0] + ((fraction * (pos[1] - pos[0])) >> 12);
+}
+#endif /* SPC_LINEAR_INTERP */
+
+#define interp( samp, pos, fwd, rev ) \
+    linear_interp( (samp), (pos) )
+
+#ifndef SPC_LINEAR_AMP
+#define linear_amp   apply_gen_amp
+#endif /* SPC_LINEAR_AMP */
+
+#define apply_amp    linear_amp
+#endif /* SPC_NOINTERP */
+
+
+#if !SPC_NOECHO
+
+#ifndef SPC_DSP_ECHO_APPLY
+/* Init FIR filter */
+static inline void echo_init( struct Spc_Dsp* this )
+{
+    this->fir.pos = 0;
+    ci->memset( this->fir.buf, 0, sizeof this->fir.buf );
+}
+
+/* Apply FIR filter */
+static inline void echo_apply(struct Spc_Dsp* this,
+        uint8_t* const echo_ptr, int* out_0, int* out_1)
+{
+    int fb_0 = GET_LE16SA( echo_ptr     );
+    int fb_1 = GET_LE16SA( echo_ptr + 2 );
+        
+    /* Keep last 8 samples */
+    int (* const fir_ptr) [2] = this->fir.buf + this->fir.pos;
+    this->fir.pos = (this->fir.pos + 1) & (FIR_BUF_HALF - 1);
+
+    fir_ptr [           0] [0] = fb_0;
+    fir_ptr [           0] [1] = fb_1;
+    /* duplicate at +8 eliminates wrap checking below */
+    fir_ptr [FIR_BUF_HALF] [0] = fb_0;
+    fir_ptr [FIR_BUF_HALF] [1] = fb_1;
+
+    fb_0 *= this->fir.coeff [0];
+    fb_1 *= this->fir.coeff [0];
+
+    #define DO_PT( i ) \
+        fb_0 += fir_ptr [i] [0] * this->fir.coeff [i]; \
+        fb_1 += fir_ptr [i] [1] * this->fir.coeff [i];
+
+    DO_PT( 1 )
+    DO_PT( 2 )
+    DO_PT( 3 )
+    DO_PT( 4 )
+    DO_PT( 5 )
+    DO_PT( 6 )
+    DO_PT( 7 )
+
+    #undef DO_PT
+
+    *out_0 = fb_0;
+    *out_1 = fb_1;
+}
+#endif /* SPC_DSP_ECHO_APPLY */
+
+#ifndef SPC_DSP_ECHO_FEEDBACK
+/* Feedback into echo buffer */
+static inline void echo_feedback( struct Spc_Dsp* this, uint8_t *echo_ptr,
+                                  int echo_0, int echo_1, int fb_0, int fb_1 )
+{
+    int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
+    int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
+    e0 = CLAMP16( e0 );
+    SET_LE16A( echo_ptr    , e0 );
+    e1 = CLAMP16( e1 );
+    SET_LE16A( echo_ptr + 2, e1 );
+}
+#endif /* SPC_DSP_ECHO_FEEDBACK */
+
+#ifndef SPC_DSP_GENERATE_OUTPUT
+/* Generate final output */
+static inline void echo_output( struct Spc_Dsp* this, int global_muting,
+    int global_vol_0, int global_vol_1, int chans_0, int chans_1, 
+    int fb_0, int fb_1, int* out_0, int* out_1 )
+{
+    *out_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
+                >> global_muting;
+    *out_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
+                >> global_muting;
+}
+#endif /* SPC_DSP_GENERATE_OUTPUT */
+
+#define mix_output echo_output
+
+#else /* SPC_NOECHO */
+
+#ifndef SPC_DSP_GENERATE_OUTPUT
+/* Generate final output */
+static inline void noecho_output( struct Spc_Dsp* this, int global_muting,
+    int global_vol_0, int global_vol_1, int chans_0, int chans_1, 
+    int* out_0, int* out_1 )
+{
+    *out_0 = (chans_0 * global_vol_0) >> global_muting;
+    *out_1 = (chans_1 * global_vol_1) >> global_muting;
+    (void)this;
+}
+#endif /* SPC_DSP_GENERATE_OUTPUT */
+
+#define mix_output(this, gm, gv0, gv1, ch0, ch1, fb_0, fb_1, o0, o1) \
+    noecho_output( (this), (gm), (gv0), (gv1), (ch0), (ch1), (o0), (o1) )
+
+#endif /* !SPC_NOECHO */