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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/libopus/src
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/libopus/src')
-rw-r--r--media/libopus/src/analysis.c672
-rw-r--r--media/libopus/src/analysis.h103
-rw-r--r--media/libopus/src/mlp.c145
-rw-r--r--media/libopus/src/mlp.h43
-rw-r--r--media/libopus/src/mlp_data.c109
-rw-r--r--media/libopus/src/opus.c356
-rw-r--r--media/libopus/src/opus_decoder.c981
-rw-r--r--media/libopus/src/opus_encoder.c2536
-rw-r--r--media/libopus/src/opus_multistream.c92
-rw-r--r--media/libopus/src/opus_multistream_decoder.c537
-rw-r--r--media/libopus/src/opus_multistream_encoder.c1351
-rw-r--r--media/libopus/src/opus_private.h134
-rw-r--r--media/libopus/src/repacketizer.c348
-rw-r--r--media/libopus/src/tansig_table.h45
14 files changed, 7452 insertions, 0 deletions
diff --git a/media/libopus/src/analysis.c b/media/libopus/src/analysis.c
new file mode 100644
index 000000000..663431a43
--- /dev/null
+++ b/media/libopus/src/analysis.c
@@ -0,0 +1,672 @@
+/* Copyright (c) 2011 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 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 "config.h"
+#endif
+
+#include "kiss_fft.h"
+#include "celt.h"
+#include "modes.h"
+#include "arch.h"
+#include "quant_bands.h"
+#include <stdio.h>
+#include "analysis.h"
+#include "mlp.h"
+#include "stack_alloc.h"
+
+#ifndef M_PI
+#define M_PI 3.141592653
+#endif
+
+static const float dct_table[128] = {
+ 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f,
+ 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f,
+ 0.351851f, 0.338330f, 0.311806f, 0.273300f, 0.224292f, 0.166664f, 0.102631f, 0.034654f,
+ -0.034654f,-0.102631f,-0.166664f,-0.224292f,-0.273300f,-0.311806f,-0.338330f,-0.351851f,
+ 0.346760f, 0.293969f, 0.196424f, 0.068975f,-0.068975f,-0.196424f,-0.293969f,-0.346760f,
+ -0.346760f,-0.293969f,-0.196424f,-0.068975f, 0.068975f, 0.196424f, 0.293969f, 0.346760f,
+ 0.338330f, 0.224292f, 0.034654f,-0.166664f,-0.311806f,-0.351851f,-0.273300f,-0.102631f,
+ 0.102631f, 0.273300f, 0.351851f, 0.311806f, 0.166664f,-0.034654f,-0.224292f,-0.338330f,
+ 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f,
+ 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f,
+ 0.311806f, 0.034654f,-0.273300f,-0.338330f,-0.102631f, 0.224292f, 0.351851f, 0.166664f,
+ -0.166664f,-0.351851f,-0.224292f, 0.102631f, 0.338330f, 0.273300f,-0.034654f,-0.311806f,
+ 0.293969f,-0.068975f,-0.346760f,-0.196424f, 0.196424f, 0.346760f, 0.068975f,-0.293969f,
+ -0.293969f, 0.068975f, 0.346760f, 0.196424f,-0.196424f,-0.346760f,-0.068975f, 0.293969f,
+ 0.273300f,-0.166664f,-0.338330f, 0.034654f, 0.351851f, 0.102631f,-0.311806f,-0.224292f,
+ 0.224292f, 0.311806f,-0.102631f,-0.351851f,-0.034654f, 0.338330f, 0.166664f,-0.273300f,
+};
+
+static const float analysis_window[240] = {
+ 0.000043f, 0.000171f, 0.000385f, 0.000685f, 0.001071f, 0.001541f, 0.002098f, 0.002739f,
+ 0.003466f, 0.004278f, 0.005174f, 0.006156f, 0.007222f, 0.008373f, 0.009607f, 0.010926f,
+ 0.012329f, 0.013815f, 0.015385f, 0.017037f, 0.018772f, 0.020590f, 0.022490f, 0.024472f,
+ 0.026535f, 0.028679f, 0.030904f, 0.033210f, 0.035595f, 0.038060f, 0.040604f, 0.043227f,
+ 0.045928f, 0.048707f, 0.051564f, 0.054497f, 0.057506f, 0.060591f, 0.063752f, 0.066987f,
+ 0.070297f, 0.073680f, 0.077136f, 0.080665f, 0.084265f, 0.087937f, 0.091679f, 0.095492f,
+ 0.099373f, 0.103323f, 0.107342f, 0.111427f, 0.115579f, 0.119797f, 0.124080f, 0.128428f,
+ 0.132839f, 0.137313f, 0.141849f, 0.146447f, 0.151105f, 0.155823f, 0.160600f, 0.165435f,
+ 0.170327f, 0.175276f, 0.180280f, 0.185340f, 0.190453f, 0.195619f, 0.200838f, 0.206107f,
+ 0.211427f, 0.216797f, 0.222215f, 0.227680f, 0.233193f, 0.238751f, 0.244353f, 0.250000f,
+ 0.255689f, 0.261421f, 0.267193f, 0.273005f, 0.278856f, 0.284744f, 0.290670f, 0.296632f,
+ 0.302628f, 0.308658f, 0.314721f, 0.320816f, 0.326941f, 0.333097f, 0.339280f, 0.345492f,
+ 0.351729f, 0.357992f, 0.364280f, 0.370590f, 0.376923f, 0.383277f, 0.389651f, 0.396044f,
+ 0.402455f, 0.408882f, 0.415325f, 0.421783f, 0.428254f, 0.434737f, 0.441231f, 0.447736f,
+ 0.454249f, 0.460770f, 0.467298f, 0.473832f, 0.480370f, 0.486912f, 0.493455f, 0.500000f,
+ 0.506545f, 0.513088f, 0.519630f, 0.526168f, 0.532702f, 0.539230f, 0.545751f, 0.552264f,
+ 0.558769f, 0.565263f, 0.571746f, 0.578217f, 0.584675f, 0.591118f, 0.597545f, 0.603956f,
+ 0.610349f, 0.616723f, 0.623077f, 0.629410f, 0.635720f, 0.642008f, 0.648271f, 0.654508f,
+ 0.660720f, 0.666903f, 0.673059f, 0.679184f, 0.685279f, 0.691342f, 0.697372f, 0.703368f,
+ 0.709330f, 0.715256f, 0.721144f, 0.726995f, 0.732807f, 0.738579f, 0.744311f, 0.750000f,
+ 0.755647f, 0.761249f, 0.766807f, 0.772320f, 0.777785f, 0.783203f, 0.788573f, 0.793893f,
+ 0.799162f, 0.804381f, 0.809547f, 0.814660f, 0.819720f, 0.824724f, 0.829673f, 0.834565f,
+ 0.839400f, 0.844177f, 0.848895f, 0.853553f, 0.858151f, 0.862687f, 0.867161f, 0.871572f,
+ 0.875920f, 0.880203f, 0.884421f, 0.888573f, 0.892658f, 0.896677f, 0.900627f, 0.904508f,
+ 0.908321f, 0.912063f, 0.915735f, 0.919335f, 0.922864f, 0.926320f, 0.929703f, 0.933013f,
+ 0.936248f, 0.939409f, 0.942494f, 0.945503f, 0.948436f, 0.951293f, 0.954072f, 0.956773f,
+ 0.959396f, 0.961940f, 0.964405f, 0.966790f, 0.969096f, 0.971321f, 0.973465f, 0.975528f,
+ 0.977510f, 0.979410f, 0.981228f, 0.982963f, 0.984615f, 0.986185f, 0.987671f, 0.989074f,
+ 0.990393f, 0.991627f, 0.992778f, 0.993844f, 0.994826f, 0.995722f, 0.996534f, 0.997261f,
+ 0.997902f, 0.998459f, 0.998929f, 0.999315f, 0.999615f, 0.999829f, 0.999957f, 1.000000f,
+};
+
+static const int tbands[NB_TBANDS+1] = {
+ 2, 4, 6, 8, 10, 12, 14, 16, 20, 24, 28, 32, 40, 48, 56, 68, 80, 96, 120
+};
+
+static const int extra_bands[NB_TOT_BANDS+1] = {
+ 1, 2, 4, 6, 8, 10, 12, 14, 16, 20, 24, 28, 32, 40, 48, 56, 68, 80, 96, 120, 160, 200
+};
+
+/*static const float tweight[NB_TBANDS+1] = {
+ .3, .4, .5, .6, .7, .8, .9, 1., 1., 1., 1., 1., 1., 1., .8, .7, .6, .5
+};*/
+
+#define NB_TONAL_SKIP_BANDS 9
+
+#define cA 0.43157974f
+#define cB 0.67848403f
+#define cC 0.08595542f
+#define cE ((float)M_PI/2)
+static OPUS_INLINE float fast_atan2f(float y, float x) {
+ float x2, y2;
+ /* Should avoid underflow on the values we'll get */
+ if (ABS16(x)+ABS16(y)<1e-9f)
+ {
+ x*=1e12f;
+ y*=1e12f;
+ }
+ x2 = x*x;
+ y2 = y*y;
+ if(x2<y2){
+ float den = (y2 + cB*x2) * (y2 + cC*x2);
+ if (den!=0)
+ return -x*y*(y2 + cA*x2) / den + (y<0 ? -cE : cE);
+ else
+ return (y<0 ? -cE : cE);
+ }else{
+ float den = (x2 + cB*y2) * (x2 + cC*y2);
+ if (den!=0)
+ return x*y*(x2 + cA*y2) / den + (y<0 ? -cE : cE) - (x*y<0 ? -cE : cE);
+ else
+ return (y<0 ? -cE : cE) - (x*y<0 ? -cE : cE);
+ }
+}
+
+void tonality_analysis_init(TonalityAnalysisState *tonal)
+{
+ /* Initialize reusable fields. */
+ tonal->arch = opus_select_arch();
+ /* Clear remaining fields. */
+ tonality_analysis_reset(tonal);
+}
+
+void tonality_analysis_reset(TonalityAnalysisState *tonal)
+{
+ /* Clear non-reusable fields. */
+ char *start = (char*)&tonal->TONALITY_ANALYSIS_RESET_START;
+ OPUS_CLEAR(start, sizeof(TonalityAnalysisState) - (start - (char*)tonal));
+}
+
+void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int len)
+{
+ int pos;
+ int curr_lookahead;
+ float psum;
+ int i;
+
+ pos = tonal->read_pos;
+ curr_lookahead = tonal->write_pos-tonal->read_pos;
+ if (curr_lookahead<0)
+ curr_lookahead += DETECT_SIZE;
+
+ if (len > 480 && pos != tonal->write_pos)
+ {
+ pos++;
+ if (pos==DETECT_SIZE)
+ pos=0;
+ }
+ if (pos == tonal->write_pos)
+ pos--;
+ if (pos<0)
+ pos = DETECT_SIZE-1;
+ OPUS_COPY(info_out, &tonal->info[pos], 1);
+ tonal->read_subframe += len/120;
+ while (tonal->read_subframe>=4)
+ {
+ tonal->read_subframe -= 4;
+ tonal->read_pos++;
+ }
+ if (tonal->read_pos>=DETECT_SIZE)
+ tonal->read_pos-=DETECT_SIZE;
+
+ /* Compensate for the delay in the features themselves.
+ FIXME: Need a better estimate the 10 I just made up */
+ curr_lookahead = IMAX(curr_lookahead-10, 0);
+
+ psum=0;
+ /* Summing the probability of transition patterns that involve music at
+ time (DETECT_SIZE-curr_lookahead-1) */
+ for (i=0;i<DETECT_SIZE-curr_lookahead;i++)
+ psum += tonal->pmusic[i];
+ for (;i<DETECT_SIZE;i++)
+ psum += tonal->pspeech[i];
+ psum = psum*tonal->music_confidence + (1-psum)*tonal->speech_confidence;
+ /*printf("%f %f %f\n", psum, info_out->music_prob, info_out->tonality);*/
+
+ info_out->music_prob = psum;
+}
+
+static void tonality_analysis(TonalityAnalysisState *tonal, const CELTMode *celt_mode, const void *x, int len, int offset, int c1, int c2, int C, int lsb_depth, downmix_func downmix)
+{
+ int i, b;
+ const kiss_fft_state *kfft;
+ VARDECL(kiss_fft_cpx, in);
+ VARDECL(kiss_fft_cpx, out);
+ int N = 480, N2=240;
+ float * OPUS_RESTRICT A = tonal->angle;
+ float * OPUS_RESTRICT dA = tonal->d_angle;
+ float * OPUS_RESTRICT d2A = tonal->d2_angle;
+ VARDECL(float, tonality);
+ VARDECL(float, noisiness);
+ float band_tonality[NB_TBANDS];
+ float logE[NB_TBANDS];
+ float BFCC[8];
+ float features[25];
+ float frame_tonality;
+ float max_frame_tonality;
+ /*float tw_sum=0;*/
+ float frame_noisiness;
+ const float pi4 = (float)(M_PI*M_PI*M_PI*M_PI);
+ float slope=0;
+ float frame_stationarity;
+ float relativeE;
+ float frame_probs[2];
+ float alpha, alphaE, alphaE2;
+ float frame_loudness;
+ float bandwidth_mask;
+ int bandwidth=0;
+ float maxE = 0;
+ float noise_floor;
+ int remaining;
+ AnalysisInfo *info;
+ SAVE_STACK;
+
+ tonal->last_transition++;
+ alpha = 1.f/IMIN(20, 1+tonal->count);
+ alphaE = 1.f/IMIN(50, 1+tonal->count);
+ alphaE2 = 1.f/IMIN(1000, 1+tonal->count);
+
+ if (tonal->count<4)
+ tonal->music_prob = .5;
+ kfft = celt_mode->mdct.kfft[0];
+ if (tonal->count==0)
+ tonal->mem_fill = 240;
+ downmix(x, &tonal->inmem[tonal->mem_fill], IMIN(len, ANALYSIS_BUF_SIZE-tonal->mem_fill), offset, c1, c2, C);
+ if (tonal->mem_fill+len < ANALYSIS_BUF_SIZE)
+ {
+ tonal->mem_fill += len;
+ /* Don't have enough to update the analysis */
+ RESTORE_STACK;
+ return;
+ }
+ info = &tonal->info[tonal->write_pos++];
+ if (tonal->write_pos>=DETECT_SIZE)
+ tonal->write_pos-=DETECT_SIZE;
+
+ ALLOC(in, 480, kiss_fft_cpx);
+ ALLOC(out, 480, kiss_fft_cpx);
+ ALLOC(tonality, 240, float);
+ ALLOC(noisiness, 240, float);
+ for (i=0;i<N2;i++)
+ {
+ float w = analysis_window[i];
+ in[i].r = (kiss_fft_scalar)(w*tonal->inmem[i]);
+ in[i].i = (kiss_fft_scalar)(w*tonal->inmem[N2+i]);
+ in[N-i-1].r = (kiss_fft_scalar)(w*tonal->inmem[N-i-1]);
+ in[N-i-1].i = (kiss_fft_scalar)(w*tonal->inmem[N+N2-i-1]);
+ }
+ OPUS_MOVE(tonal->inmem, tonal->inmem+ANALYSIS_BUF_SIZE-240, 240);
+ remaining = len - (ANALYSIS_BUF_SIZE-tonal->mem_fill);
+ downmix(x, &tonal->inmem[240], remaining, offset+ANALYSIS_BUF_SIZE-tonal->mem_fill, c1, c2, C);
+ tonal->mem_fill = 240 + remaining;
+ opus_fft(kfft, in, out, tonal->arch);
+#ifndef FIXED_POINT
+ /* If there's any NaN on the input, the entire output will be NaN, so we only need to check one value. */
+ if (celt_isnan(out[0].r))
+ {
+ info->valid = 0;
+ RESTORE_STACK;
+ return;
+ }
+#endif
+
+ for (i=1;i<N2;i++)
+ {
+ float X1r, X2r, X1i, X2i;
+ float angle, d_angle, d2_angle;
+ float angle2, d_angle2, d2_angle2;
+ float mod1, mod2, avg_mod;
+ X1r = (float)out[i].r+out[N-i].r;
+ X1i = (float)out[i].i-out[N-i].i;
+ X2r = (float)out[i].i+out[N-i].i;
+ X2i = (float)out[N-i].r-out[i].r;
+
+ angle = (float)(.5f/M_PI)*fast_atan2f(X1i, X1r);
+ d_angle = angle - A[i];
+ d2_angle = d_angle - dA[i];
+
+ angle2 = (float)(.5f/M_PI)*fast_atan2f(X2i, X2r);
+ d_angle2 = angle2 - angle;
+ d2_angle2 = d_angle2 - d_angle;
+
+ mod1 = d2_angle - (float)floor(.5+d2_angle);
+ noisiness[i] = ABS16(mod1);
+ mod1 *= mod1;
+ mod1 *= mod1;
+
+ mod2 = d2_angle2 - (float)floor(.5+d2_angle2);
+ noisiness[i] += ABS16(mod2);
+ mod2 *= mod2;
+ mod2 *= mod2;
+
+ avg_mod = .25f*(d2A[i]+2.f*mod1+mod2);
+ tonality[i] = 1.f/(1.f+40.f*16.f*pi4*avg_mod)-.015f;
+
+ A[i] = angle2;
+ dA[i] = d_angle2;
+ d2A[i] = mod2;
+ }
+
+ frame_tonality = 0;
+ max_frame_tonality = 0;
+ /*tw_sum = 0;*/
+ info->activity = 0;
+ frame_noisiness = 0;
+ frame_stationarity = 0;
+ if (!tonal->count)
+ {
+ for (b=0;b<NB_TBANDS;b++)
+ {
+ tonal->lowE[b] = 1e10;
+ tonal->highE[b] = -1e10;
+ }
+ }
+ relativeE = 0;
+ frame_loudness = 0;
+ for (b=0;b<NB_TBANDS;b++)
+ {
+ float E=0, tE=0, nE=0;
+ float L1, L2;
+ float stationarity;
+ for (i=tbands[b];i<tbands[b+1];i++)
+ {
+ float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r
+ + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i;
+#ifdef FIXED_POINT
+ /* FIXME: It's probably best to change the BFCC filter initial state instead */
+ binE *= 5.55e-17f;
+#endif
+ E += binE;
+ tE += binE*tonality[i];
+ nE += binE*2.f*(.5f-noisiness[i]);
+ }
+#ifndef FIXED_POINT
+ /* Check for extreme band energies that could cause NaNs later. */
+ if (!(E<1e9f) || celt_isnan(E))
+ {
+ info->valid = 0;
+ RESTORE_STACK;
+ return;
+ }
+#endif
+
+ tonal->E[tonal->E_count][b] = E;
+ frame_noisiness += nE/(1e-15f+E);
+
+ frame_loudness += (float)sqrt(E+1e-10f);
+ logE[b] = (float)log(E+1e-10f);
+ tonal->lowE[b] = MIN32(logE[b], tonal->lowE[b]+.01f);
+ tonal->highE[b] = MAX32(logE[b], tonal->highE[b]-.1f);
+ if (tonal->highE[b] < tonal->lowE[b]+1.f)
+ {
+ tonal->highE[b]+=.5f;
+ tonal->lowE[b]-=.5f;
+ }
+ relativeE += (logE[b]-tonal->lowE[b])/(1e-15f+tonal->highE[b]-tonal->lowE[b]);
+
+ L1=L2=0;
+ for (i=0;i<NB_FRAMES;i++)
+ {
+ L1 += (float)sqrt(tonal->E[i][b]);
+ L2 += tonal->E[i][b];
+ }
+
+ stationarity = MIN16(0.99f,L1/(float)sqrt(1e-15+NB_FRAMES*L2));
+ stationarity *= stationarity;
+ stationarity *= stationarity;
+ frame_stationarity += stationarity;
+ /*band_tonality[b] = tE/(1e-15+E)*/;
+ band_tonality[b] = MAX16(tE/(1e-15f+E), stationarity*tonal->prev_band_tonality[b]);
+#if 0
+ if (b>=NB_TONAL_SKIP_BANDS)
+ {
+ frame_tonality += tweight[b]*band_tonality[b];
+ tw_sum += tweight[b];
+ }
+#else
+ frame_tonality += band_tonality[b];
+ if (b>=NB_TBANDS-NB_TONAL_SKIP_BANDS)
+ frame_tonality -= band_tonality[b-NB_TBANDS+NB_TONAL_SKIP_BANDS];
+#endif
+ max_frame_tonality = MAX16(max_frame_tonality, (1.f+.03f*(b-NB_TBANDS))*frame_tonality);
+ slope += band_tonality[b]*(b-8);
+ /*printf("%f %f ", band_tonality[b], stationarity);*/
+ tonal->prev_band_tonality[b] = band_tonality[b];
+ }
+
+ bandwidth_mask = 0;
+ bandwidth = 0;
+ maxE = 0;
+ noise_floor = 5.7e-4f/(1<<(IMAX(0,lsb_depth-8)));
+#ifdef FIXED_POINT
+ noise_floor *= 1<<(15+SIG_SHIFT);
+#endif
+ noise_floor *= noise_floor;
+ for (b=0;b<NB_TOT_BANDS;b++)
+ {
+ float E=0;
+ int band_start, band_end;
+ /* Keep a margin of 300 Hz for aliasing */
+ band_start = extra_bands[b];
+ band_end = extra_bands[b+1];
+ for (i=band_start;i<band_end;i++)
+ {
+ float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r
+ + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i;
+ E += binE;
+ }
+ maxE = MAX32(maxE, E);
+ tonal->meanE[b] = MAX32((1-alphaE2)*tonal->meanE[b], E);
+ E = MAX32(E, tonal->meanE[b]);
+ /* Use a simple follower with 13 dB/Bark slope for spreading function */
+ bandwidth_mask = MAX32(.05f*bandwidth_mask, E);
+ /* Consider the band "active" only if all these conditions are met:
+ 1) less than 10 dB below the simple follower
+ 2) less than 90 dB below the peak band (maximal masking possible considering
+ both the ATH and the loudness-dependent slope of the spreading function)
+ 3) above the PCM quantization noise floor
+ */
+ if (E>.1*bandwidth_mask && E*1e9f > maxE && E > noise_floor*(band_end-band_start))
+ bandwidth = b;
+ }
+ if (tonal->count<=2)
+ bandwidth = 20;
+ frame_loudness = 20*(float)log10(frame_loudness);
+ tonal->Etracker = MAX32(tonal->Etracker-.03f, frame_loudness);
+ tonal->lowECount *= (1-alphaE);
+ if (frame_loudness < tonal->Etracker-30)
+ tonal->lowECount += alphaE;
+
+ for (i=0;i<8;i++)
+ {
+ float sum=0;
+ for (b=0;b<16;b++)
+ sum += dct_table[i*16+b]*logE[b];
+ BFCC[i] = sum;
+ }
+
+ frame_stationarity /= NB_TBANDS;
+ relativeE /= NB_TBANDS;
+ if (tonal->count<10)
+ relativeE = .5;
+ frame_noisiness /= NB_TBANDS;
+#if 1
+ info->activity = frame_noisiness + (1-frame_noisiness)*relativeE;
+#else
+ info->activity = .5*(1+frame_noisiness-frame_stationarity);
+#endif
+ frame_tonality = (max_frame_tonality/(NB_TBANDS-NB_TONAL_SKIP_BANDS));
+ frame_tonality = MAX16(frame_tonality, tonal->prev_tonality*.8f);
+ tonal->prev_tonality = frame_tonality;
+
+ slope /= 8*8;
+ info->tonality_slope = slope;
+
+ tonal->E_count = (tonal->E_count+1)%NB_FRAMES;
+ tonal->count++;
+ info->tonality = frame_tonality;
+
+ for (i=0;i<4;i++)
+ features[i] = -0.12299f*(BFCC[i]+tonal->mem[i+24]) + 0.49195f*(tonal->mem[i]+tonal->mem[i+16]) + 0.69693f*tonal->mem[i+8] - 1.4349f*tonal->cmean[i];
+
+ for (i=0;i<4;i++)
+ tonal->cmean[i] = (1-alpha)*tonal->cmean[i] + alpha*BFCC[i];
+
+ for (i=0;i<4;i++)
+ features[4+i] = 0.63246f*(BFCC[i]-tonal->mem[i+24]) + 0.31623f*(tonal->mem[i]-tonal->mem[i+16]);
+ for (i=0;i<3;i++)
+ features[8+i] = 0.53452f*(BFCC[i]+tonal->mem[i+24]) - 0.26726f*(tonal->mem[i]+tonal->mem[i+16]) -0.53452f*tonal->mem[i+8];
+
+ if (tonal->count > 5)
+ {
+ for (i=0;i<9;i++)
+ tonal->std[i] = (1-alpha)*tonal->std[i] + alpha*features[i]*features[i];
+ }
+
+ for (i=0;i<8;i++)
+ {
+ tonal->mem[i+24] = tonal->mem[i+16];
+ tonal->mem[i+16] = tonal->mem[i+8];
+ tonal->mem[i+8] = tonal->mem[i];
+ tonal->mem[i] = BFCC[i];
+ }
+ for (i=0;i<9;i++)
+ features[11+i] = (float)sqrt(tonal->std[i]);
+ features[20] = info->tonality;
+ features[21] = info->activity;
+ features[22] = frame_stationarity;
+ features[23] = info->tonality_slope;
+ features[24] = tonal->lowECount;
+
+#ifndef DISABLE_FLOAT_API
+ mlp_process(&net, features, frame_probs);
+ frame_probs[0] = .5f*(frame_probs[0]+1);
+ /* Curve fitting between the MLP probability and the actual probability */
+ frame_probs[0] = .01f + 1.21f*frame_probs[0]*frame_probs[0] - .23f*(float)pow(frame_probs[0], 10);
+ /* Probability of active audio (as opposed to silence) */
+ frame_probs[1] = .5f*frame_probs[1]+.5f;
+ /* Consider that silence has a 50-50 probability. */
+ frame_probs[0] = frame_probs[1]*frame_probs[0] + (1-frame_probs[1])*.5f;
+
+ /*printf("%f %f ", frame_probs[0], frame_probs[1]);*/
+ {
+ /* Probability of state transition */
+ float tau;
+ /* Represents independence of the MLP probabilities, where
+ beta=1 means fully independent. */
+ float beta;
+ /* Denormalized probability of speech (p0) and music (p1) after update */
+ float p0, p1;
+ /* Probabilities for "all speech" and "all music" */
+ float s0, m0;
+ /* Probability sum for renormalisation */
+ float psum;
+ /* Instantaneous probability of speech and music, with beta pre-applied. */
+ float speech0;
+ float music0;
+ float p, q;
+
+ /* One transition every 3 minutes of active audio */
+ tau = .00005f*frame_probs[1];
+ /* Adapt beta based on how "unexpected" the new prob is */
+ p = MAX16(.05f,MIN16(.95f,frame_probs[0]));
+ q = MAX16(.05f,MIN16(.95f,tonal->music_prob));
+ beta = .01f+.05f*ABS16(p-q)/(p*(1-q)+q*(1-p));
+ /* p0 and p1 are the probabilities of speech and music at this frame
+ using only information from previous frame and applying the
+ state transition model */
+ p0 = (1-tonal->music_prob)*(1-tau) + tonal->music_prob *tau;
+ p1 = tonal->music_prob *(1-tau) + (1-tonal->music_prob)*tau;
+ /* We apply the current probability with exponent beta to work around
+ the fact that the probability estimates aren't independent. */
+ p0 *= (float)pow(1-frame_probs[0], beta);
+ p1 *= (float)pow(frame_probs[0], beta);
+ /* Normalise the probabilities to get the Marokv probability of music. */
+ tonal->music_prob = p1/(p0+p1);
+ info->music_prob = tonal->music_prob;
+
+ /* This chunk of code deals with delayed decision. */
+ psum=1e-20f;
+ /* Instantaneous probability of speech and music, with beta pre-applied. */
+ speech0 = (float)pow(1-frame_probs[0], beta);
+ music0 = (float)pow(frame_probs[0], beta);
+ if (tonal->count==1)
+ {
+ tonal->pspeech[0]=.5;
+ tonal->pmusic [0]=.5;
+ }
+ /* Updated probability of having only speech (s0) or only music (m0),
+ before considering the new observation. */
+ s0 = tonal->pspeech[0] + tonal->pspeech[1];
+ m0 = tonal->pmusic [0] + tonal->pmusic [1];
+ /* Updates s0 and m0 with instantaneous probability. */
+ tonal->pspeech[0] = s0*(1-tau)*speech0;
+ tonal->pmusic [0] = m0*(1-tau)*music0;
+ /* Propagate the transition probabilities */
+ for (i=1;i<DETECT_SIZE-1;i++)
+ {
+ tonal->pspeech[i] = tonal->pspeech[i+1]*speech0;
+ tonal->pmusic [i] = tonal->pmusic [i+1]*music0;
+ }
+ /* Probability that the latest frame is speech, when all the previous ones were music. */
+ tonal->pspeech[DETECT_SIZE-1] = m0*tau*speech0;
+ /* Probability that the latest frame is music, when all the previous ones were speech. */
+ tonal->pmusic [DETECT_SIZE-1] = s0*tau*music0;
+
+ /* Renormalise probabilities to 1 */
+ for (i=0;i<DETECT_SIZE;i++)
+ psum += tonal->pspeech[i] + tonal->pmusic[i];
+ psum = 1.f/psum;
+ for (i=0;i<DETECT_SIZE;i++)
+ {
+ tonal->pspeech[i] *= psum;
+ tonal->pmusic [i] *= psum;
+ }
+ psum = tonal->pmusic[0];
+ for (i=1;i<DETECT_SIZE;i++)
+ psum += tonal->pspeech[i];
+
+ /* Estimate our confidence in the speech/music decisions */
+ if (frame_probs[1]>.75)
+ {
+ if (tonal->music_prob>.9)
+ {
+ float adapt;
+ adapt = 1.f/(++tonal->music_confidence_count);
+ tonal->music_confidence_count = IMIN(tonal->music_confidence_count, 500);
+ tonal->music_confidence += adapt*MAX16(-.2f,frame_probs[0]-tonal->music_confidence);
+ }
+ if (tonal->music_prob<.1)
+ {
+ float adapt;
+ adapt = 1.f/(++tonal->speech_confidence_count);
+ tonal->speech_confidence_count = IMIN(tonal->speech_confidence_count, 500);
+ tonal->speech_confidence += adapt*MIN16(.2f,frame_probs[0]-tonal->speech_confidence);
+ }
+ } else {
+ if (tonal->music_confidence_count==0)
+ tonal->music_confidence = .9f;
+ if (tonal->speech_confidence_count==0)
+ tonal->speech_confidence = .1f;
+ }
+ }
+ if (tonal->last_music != (tonal->music_prob>.5f))
+ tonal->last_transition=0;
+ tonal->last_music = tonal->music_prob>.5f;
+#else
+ info->music_prob = 0;
+#endif
+ /*for (i=0;i<25;i++)
+ printf("%f ", features[i]);
+ printf("\n");*/
+
+ info->bandwidth = bandwidth;
+ /*printf("%d %d\n", info->bandwidth, info->opus_bandwidth);*/
+ info->noisiness = frame_noisiness;
+ info->valid = 1;
+ RESTORE_STACK;
+}
+
+void run_analysis(TonalityAnalysisState *analysis, const CELTMode *celt_mode, const void *analysis_pcm,
+ int analysis_frame_size, int frame_size, int c1, int c2, int C, opus_int32 Fs,
+ int lsb_depth, downmix_func downmix, AnalysisInfo *analysis_info)
+{
+ int offset;
+ int pcm_len;
+
+ if (analysis_pcm != NULL)
+ {
+ /* Avoid overflow/wrap-around of the analysis buffer */
+ analysis_frame_size = IMIN((DETECT_SIZE-5)*Fs/100, analysis_frame_size);
+
+ pcm_len = analysis_frame_size - analysis->analysis_offset;
+ offset = analysis->analysis_offset;
+ do {
+ tonality_analysis(analysis, celt_mode, analysis_pcm, IMIN(480, pcm_len), offset, c1, c2, C, lsb_depth, downmix);
+ offset += 480;
+ pcm_len -= 480;
+ } while (pcm_len>0);
+ analysis->analysis_offset = analysis_frame_size;
+
+ analysis->analysis_offset -= frame_size;
+ }
+
+ analysis_info->valid = 0;
+ tonality_get_info(analysis, analysis_info, frame_size);
+}
diff --git a/media/libopus/src/analysis.h b/media/libopus/src/analysis.h
new file mode 100644
index 000000000..9eae56a52
--- /dev/null
+++ b/media/libopus/src/analysis.h
@@ -0,0 +1,103 @@
+/* Copyright (c) 2011 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 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 ANALYSIS_H
+#define ANALYSIS_H
+
+#include "celt.h"
+#include "opus_private.h"
+
+#define NB_FRAMES 8
+#define NB_TBANDS 18
+#define NB_TOT_BANDS 21
+#define ANALYSIS_BUF_SIZE 720 /* 15 ms at 48 kHz */
+
+#define DETECT_SIZE 200
+
+typedef struct {
+ int arch;
+#define TONALITY_ANALYSIS_RESET_START angle
+ float angle[240];
+ float d_angle[240];
+ float d2_angle[240];
+ opus_val32 inmem[ANALYSIS_BUF_SIZE];
+ int mem_fill; /* number of usable samples in the buffer */
+ float prev_band_tonality[NB_TBANDS];
+ float prev_tonality;
+ float E[NB_FRAMES][NB_TBANDS];
+ float lowE[NB_TBANDS];
+ float highE[NB_TBANDS];
+ float meanE[NB_TOT_BANDS];
+ float mem[32];
+ float cmean[8];
+ float std[9];
+ float music_prob;
+ float Etracker;
+ float lowECount;
+ int E_count;
+ int last_music;
+ int last_transition;
+ int count;
+ float subframe_mem[3];
+ int analysis_offset;
+ /** Probability of having speech for time i to DETECT_SIZE-1 (and music before).
+ pspeech[0] is the probability that all frames in the window are speech. */
+ float pspeech[DETECT_SIZE];
+ /** Probability of having music for time i to DETECT_SIZE-1 (and speech before).
+ pmusic[0] is the probability that all frames in the window are music. */
+ float pmusic[DETECT_SIZE];
+ float speech_confidence;
+ float music_confidence;
+ int speech_confidence_count;
+ int music_confidence_count;
+ int write_pos;
+ int read_pos;
+ int read_subframe;
+ AnalysisInfo info[DETECT_SIZE];
+} TonalityAnalysisState;
+
+/** Initialize a TonalityAnalysisState struct.
+ *
+ * This performs some possibly slow initialization steps which should
+ * not be repeated every analysis step. No allocated memory is retained
+ * by the state struct, so no cleanup call is required.
+ */
+void tonality_analysis_init(TonalityAnalysisState *analysis);
+
+/** Reset a TonalityAnalysisState stuct.
+ *
+ * Call this when there's a discontinuity in the data.
+ */
+void tonality_analysis_reset(TonalityAnalysisState *analysis);
+
+void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int len);
+
+void run_analysis(TonalityAnalysisState *analysis, const CELTMode *celt_mode, const void *analysis_pcm,
+ int analysis_frame_size, int frame_size, int c1, int c2, int C, opus_int32 Fs,
+ int lsb_depth, downmix_func downmix, AnalysisInfo *analysis_info);
+
+#endif
diff --git a/media/libopus/src/mlp.c b/media/libopus/src/mlp.c
new file mode 100644
index 000000000..ff9e50df4
--- /dev/null
+++ b/media/libopus/src/mlp.c
@@ -0,0 +1,145 @@
+/* Copyright (c) 2008-2011 Octasic Inc.
+ 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 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 "config.h"
+#endif
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#include <math.h>
+#include "mlp.h"
+#include "arch.h"
+#include "tansig_table.h"
+#define MAX_NEURONS 100
+
+#if 0
+static OPUS_INLINE opus_val16 tansig_approx(opus_val32 _x) /* Q19 */
+{
+ int i;
+ opus_val16 xx; /* Q11 */
+ /*double x, y;*/
+ opus_val16 dy, yy; /* Q14 */
+ /*x = 1.9073e-06*_x;*/
+ if (_x>=QCONST32(8,19))
+ return QCONST32(1.,14);
+ if (_x<=-QCONST32(8,19))
+ return -QCONST32(1.,14);
+ xx = EXTRACT16(SHR32(_x, 8));
+ /*i = lrint(25*x);*/
+ i = SHR32(ADD32(1024,MULT16_16(25, xx)),11);
+ /*x -= .04*i;*/
+ xx -= EXTRACT16(SHR32(MULT16_16(20972,i),8));
+ /*x = xx*(1./2048);*/
+ /*y = tansig_table[250+i];*/
+ yy = tansig_table[250+i];
+ /*y = yy*(1./16384);*/
+ dy = 16384-MULT16_16_Q14(yy,yy);
+ yy = yy + MULT16_16_Q14(MULT16_16_Q11(xx,dy),(16384 - MULT16_16_Q11(yy,xx)));
+ return yy;
+}
+#else
+/*extern const float tansig_table[501];*/
+static OPUS_INLINE float tansig_approx(float x)
+{
+ int i;
+ float y, dy;
+ float sign=1;
+ /* Tests are reversed to catch NaNs */
+ if (!(x<8))
+ return 1;
+ if (!(x>-8))
+ return -1;
+#ifndef FIXED_POINT
+ /* Another check in case of -ffast-math */
+ if (celt_isnan(x))
+ return 0;
+#endif
+ if (x<0)
+ {
+ x=-x;
+ sign=-1;
+ }
+ i = (int)floor(.5f+25*x);
+ x -= .04f*i;
+ y = tansig_table[i];
+ dy = 1-y*y;
+ y = y + x*dy*(1 - y*x);
+ return sign*y;
+}
+#endif
+
+#if 0
+void mlp_process(const MLP *m, const opus_val16 *in, opus_val16 *out)
+{
+ int j;
+ opus_val16 hidden[MAX_NEURONS];
+ const opus_val16 *W = m->weights;
+ /* Copy to tmp_in */
+ for (j=0;j<m->topo[1];j++)
+ {
+ int k;
+ opus_val32 sum = SHL32(EXTEND32(*W++),8);
+ for (k=0;k<m->topo[0];k++)
+ sum = MAC16_16(sum, in[k],*W++);
+ hidden[j] = tansig_approx(sum);
+ }
+ for (j=0;j<m->topo[2];j++)
+ {
+ int k;
+ opus_val32 sum = SHL32(EXTEND32(*W++),14);
+ for (k=0;k<m->topo[1];k++)
+ sum = MAC16_16(sum, hidden[k], *W++);
+ out[j] = tansig_approx(EXTRACT16(PSHR32(sum,17)));
+ }
+}
+#else
+void mlp_process(const MLP *m, const float *in, float *out)
+{
+ int j;
+ float hidden[MAX_NEURONS];
+ const float *W = m->weights;
+ /* Copy to tmp_in */
+ for (j=0;j<m->topo[1];j++)
+ {
+ int k;
+ float sum = *W++;
+ for (k=0;k<m->topo[0];k++)
+ sum = sum + in[k]**W++;
+ hidden[j] = tansig_approx(sum);
+ }
+ for (j=0;j<m->topo[2];j++)
+ {
+ int k;
+ float sum = *W++;
+ for (k=0;k<m->topo[1];k++)
+ sum = sum + hidden[k]**W++;
+ out[j] = tansig_approx(sum);
+ }
+}
+#endif
diff --git a/media/libopus/src/mlp.h b/media/libopus/src/mlp.h
new file mode 100644
index 000000000..618e246e2
--- /dev/null
+++ b/media/libopus/src/mlp.h
@@ -0,0 +1,43 @@
+/* Copyright (c) 2008-2011 Octasic Inc.
+ 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 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 _MLP_H_
+#define _MLP_H_
+
+#include "arch.h"
+
+typedef struct {
+ int layers;
+ const int *topo;
+ const float *weights;
+} MLP;
+
+extern const MLP net;
+
+void mlp_process(const MLP *m, const float *in, float *out);
+
+#endif /* _MLP_H_ */
diff --git a/media/libopus/src/mlp_data.c b/media/libopus/src/mlp_data.c
new file mode 100644
index 000000000..c2fda4e2e
--- /dev/null
+++ b/media/libopus/src/mlp_data.c
@@ -0,0 +1,109 @@
+/* The contents of this file was automatically generated by mlp_train.c
+ It contains multi-layer perceptron (MLP) weights. */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "mlp.h"
+
+/* RMS error was 0.138320, seed was 1361535663 */
+
+static const float weights[422] = {
+
+/* hidden layer */
+-0.0941125f, -0.302976f, -0.603555f, -0.19393f, -0.185983f,
+-0.601617f, -0.0465317f, -0.114563f, -0.103599f, -0.618938f,
+-0.317859f, -0.169949f, -0.0702885f, 0.148065f, 0.409524f,
+0.548432f, 0.367649f, -0.494393f, 0.764306f, -1.83957f,
+0.170849f, 12.786f, -1.08848f, -1.27284f, -16.2606f,
+24.1773f, -5.57454f, -0.17276f, -0.163388f, -0.224421f,
+-0.0948944f, -0.0728695f, -0.26557f, -0.100283f, -0.0515459f,
+-0.146142f, -0.120674f, -0.180655f, 0.12857f, 0.442138f,
+-0.493735f, 0.167767f, 0.206699f, -0.197567f, 0.417999f,
+1.50364f, -0.773341f, -10.0401f, 0.401872f, 2.97966f,
+15.2165f, -1.88905f, -1.19254f, 0.0285397f, -0.00405139f,
+0.0707565f, 0.00825699f, -0.0927269f, -0.010393f, -0.00428882f,
+-0.00489743f, -0.0709731f, -0.00255992f, 0.0395619f, 0.226424f,
+0.0325231f, 0.162175f, -0.100118f, 0.485789f, 0.12697f,
+0.285937f, 0.0155637f, 0.10546f, 3.05558f, 1.15059f,
+-1.00904f, -1.83088f, 3.31766f, -3.42516f, -0.119135f,
+-0.0405654f, 0.00690068f, 0.0179877f, -0.0382487f, 0.00597941f,
+-0.0183611f, 0.00190395f, -0.144322f, -0.0435671f, 0.000990594f,
+0.221087f, 0.142405f, 0.484066f, 0.404395f, 0.511955f,
+-0.237255f, 0.241742f, 0.35045f, -0.699428f, 10.3993f,
+2.6507f, -2.43459f, -4.18838f, 1.05928f, 1.71067f,
+0.00667811f, -0.0721335f, -0.0397346f, 0.0362704f, -0.11496f,
+-0.0235776f, 0.0082161f, -0.0141741f, -0.0329699f, -0.0354253f,
+0.00277404f, -0.290654f, -1.14767f, -0.319157f, -0.686544f,
+0.36897f, 0.478899f, 0.182579f, -0.411069f, 0.881104f,
+-4.60683f, 1.4697f, 0.335845f, -1.81905f, -30.1699f,
+5.55225f, 0.0019508f, -0.123576f, -0.0727332f, -0.0641597f,
+-0.0534458f, -0.108166f, -0.0937368f, -0.0697883f, -0.0275475f,
+-0.192309f, -0.110074f, 0.285375f, -0.405597f, 0.0926724f,
+-0.287881f, -0.851193f, -0.099493f, -0.233764f, -1.2852f,
+1.13611f, 3.12168f, -0.0699f, -1.86216f, 2.65292f,
+-7.31036f, 2.44776f, -0.00111802f, -0.0632786f, -0.0376296f,
+-0.149851f, 0.142963f, 0.184368f, 0.123433f, 0.0756158f,
+0.117312f, 0.0933395f, 0.0692163f, 0.0842592f, 0.0704683f,
+0.0589963f, 0.0942205f, -0.448862f, 0.0262677f, 0.270352f,
+-0.262317f, 0.172586f, 2.00227f, -0.159216f, 0.038422f,
+10.2073f, 4.15536f, -2.3407f, -0.0550265f, 0.00964792f,
+-0.141336f, 0.0274501f, 0.0343921f, -0.0487428f, 0.0950172f,
+-0.00775017f, -0.0372492f, -0.00548121f, -0.0663695f, 0.0960506f,
+-0.200008f, -0.0412827f, 0.58728f, 0.0515787f, 0.337254f,
+0.855024f, 0.668371f, -0.114904f, -3.62962f, -0.467477f,
+-0.215472f, 2.61537f, 0.406117f, -1.36373f, 0.0425394f,
+0.12208f, 0.0934502f, 0.123055f, 0.0340935f, -0.142466f,
+0.035037f, -0.0490666f, 0.0733208f, 0.0576672f, 0.123984f,
+-0.0517194f, -0.253018f, 0.590565f, 0.145849f, 0.315185f,
+0.221534f, -0.149081f, 0.216161f, -0.349575f, 24.5664f,
+-0.994196f, 0.614289f, -18.7905f, -2.83277f, -0.716801f,
+-0.347201f, 0.479515f, -0.246027f, 0.0758683f, 0.137293f,
+-0.17781f, 0.118751f, -0.00108329f, -0.237334f, 0.355732f,
+-0.12991f, -0.0547627f, -0.318576f, -0.325524f, 0.180494f,
+-0.0625604f, 0.141219f, 0.344064f, 0.37658f, -0.591772f,
+5.8427f, -0.38075f, 0.221894f, -1.41934f, -1.87943e+06f,
+1.34114f, 0.0283355f, -0.0447856f, -0.0211466f, -0.0256927f,
+0.0139618f, 0.0207934f, -0.0107666f, 0.0110969f, 0.0586069f,
+-0.0253545f, -0.0328433f, 0.11872f, -0.216943f, 0.145748f,
+0.119808f, -0.0915211f, -0.120647f, -0.0787719f, -0.143644f,
+-0.595116f, -1.152f, -1.25335f, -1.17092f, 4.34023f,
+-975268.f, -1.37033f, -0.0401123f, 0.210602f, -0.136656f,
+0.135962f, -0.0523293f, 0.0444604f, 0.0143928f, 0.00412666f,
+-0.0193003f, 0.218452f, -0.110204f, -2.02563f, 0.918238f,
+-2.45362f, 1.19542f, -0.061362f, -1.92243f, 0.308111f,
+0.49764f, 0.912356f, 0.209272f, -2.34525f, 2.19326f,
+-6.47121f, 1.69771f, -0.725123f, 0.0118929f, 0.0377944f,
+0.0554003f, 0.0226452f, -0.0704421f, -0.0300309f, 0.0122978f,
+-0.0041782f, -0.0686612f, 0.0313115f, 0.039111f, 0.364111f,
+-0.0945548f, 0.0229876f, -0.17414f, 0.329795f, 0.114714f,
+0.30022f, 0.106997f, 0.132355f, 5.79932f, 0.908058f,
+-0.905324f, -3.3561f, 0.190647f, 0.184211f, -0.673648f,
+0.231807f, -0.0586222f, 0.230752f, -0.438277f, 0.245857f,
+-0.17215f, 0.0876383f, -0.720512f, 0.162515f, 0.0170571f,
+0.101781f, 0.388477f, 1.32931f, 1.08548f, -0.936301f,
+-2.36958f, -6.71988f, -3.44376f, 2.13818f, 14.2318f,
+4.91459f, -3.09052f, -9.69191f, -0.768234f, 1.79604f,
+0.0549653f, 0.163399f, 0.0797025f, 0.0343933f, -0.0555876f,
+-0.00505673f, 0.0187258f, 0.0326628f, 0.0231486f, 0.15573f,
+0.0476223f, -0.254824f, 1.60155f, -0.801221f, 2.55496f,
+0.737629f, -1.36249f, -0.695463f, -2.44301f, -1.73188f,
+3.95279f, 1.89068f, 0.486087f, -11.3343f, 3.9416e+06f,
+
+/* output layer */
+-0.381439f, 0.12115f, -0.906927f, 2.93878f, 1.6388f,
+0.882811f, 0.874344f, 1.21726f, -0.874545f, 0.321706f,
+0.785055f, 0.946558f, -0.575066f, -3.46553f, 0.884905f,
+0.0924047f, -9.90712f, 0.391338f, 0.160103f, -2.04954f,
+4.1455f, 0.0684029f, -0.144761f, -0.285282f, 0.379244f,
+-1.1584f, -0.0277241f, -9.85f, -4.82386f, 3.71333f,
+3.87308f, 3.52558f};
+
+static const int topo[3] = {25, 15, 2};
+
+const MLP net = {
+ 3,
+ topo,
+ weights
+};
diff --git a/media/libopus/src/opus.c b/media/libopus/src/opus.c
new file mode 100644
index 000000000..f76f125cf
--- /dev/null
+++ b/media/libopus/src/opus.c
@@ -0,0 +1,356 @@
+/* Copyright (c) 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.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "opus.h"
+#include "opus_private.h"
+
+#ifndef DISABLE_FLOAT_API
+OPUS_EXPORT void opus_pcm_soft_clip(float *_x, int N, int C, float *declip_mem)
+{
+ int c;
+ int i;
+ float *x;
+
+ if (C<1 || N<1 || !_x || !declip_mem) return;
+
+ /* First thing: saturate everything to +/- 2 which is the highest level our
+ non-linearity can handle. At the point where the signal reaches +/-2,
+ the derivative will be zero anyway, so this doesn't introduce any
+ discontinuity in the derivative. */
+ for (i=0;i<N*C;i++)
+ _x[i] = MAX16(-2.f, MIN16(2.f, _x[i]));
+ for (c=0;c<C;c++)
+ {
+ float a;
+ float x0;
+ int curr;
+
+ x = _x+c;
+ a = declip_mem[c];
+ /* Continue applying the non-linearity from the previous frame to avoid
+ any discontinuity. */
+ for (i=0;i<N;i++)
+ {
+ if (x[i*C]*a>=0)
+ break;
+ x[i*C] = x[i*C]+a*x[i*C]*x[i*C];
+ }
+
+ curr=0;
+ x0 = x[0];
+ while(1)
+ {
+ int start, end;
+ float maxval;
+ int special=0;
+ int peak_pos;
+ for (i=curr;i<N;i++)
+ {
+ if (x[i*C]>1 || x[i*C]<-1)
+ break;
+ }
+ if (i==N)
+ {
+ a=0;
+ break;
+ }
+ peak_pos = i;
+ start=end=i;
+ maxval=ABS16(x[i*C]);
+ /* Look for first zero crossing before clipping */
+ while (start>0 && x[i*C]*x[(start-1)*C]>=0)
+ start--;
+ /* Look for first zero crossing after clipping */
+ while (end<N && x[i*C]*x[end*C]>=0)
+ {
+ /* Look for other peaks until the next zero-crossing. */
+ if (ABS16(x[end*C])>maxval)
+ {
+ maxval = ABS16(x[end*C]);
+ peak_pos = end;
+ }
+ end++;
+ }
+ /* Detect the special case where we clip before the first zero crossing */
+ special = (start==0 && x[i*C]*x[0]>=0);
+
+ /* Compute a such that maxval + a*maxval^2 = 1 */
+ a=(maxval-1)/(maxval*maxval);
+ /* Slightly boost "a" by 2^-22. This is just enough to ensure -ffast-math
+ does not cause output values larger than +/-1, but small enough not
+ to matter even for 24-bit output. */
+ a += a*2.4e-7;
+ if (x[i*C]>0)
+ a = -a;
+ /* Apply soft clipping */
+ for (i=start;i<end;i++)
+ x[i*C] = x[i*C]+a*x[i*C]*x[i*C];
+
+ if (special && peak_pos>=2)
+ {
+ /* Add a linear ramp from the first sample to the signal peak.
+ This avoids a discontinuity at the beginning of the frame. */
+ float delta;
+ float offset = x0-x[0];
+ delta = offset / peak_pos;
+ for (i=curr;i<peak_pos;i++)
+ {
+ offset -= delta;
+ x[i*C] += offset;
+ x[i*C] = MAX16(-1.f, MIN16(1.f, x[i*C]));
+ }
+ }
+ curr = end;
+ if (curr==N)
+ break;
+ }
+ declip_mem[c] = a;
+ }
+}
+#endif
+
+int encode_size(int size, unsigned char *data)
+{
+ if (size < 252)
+ {
+ data[0] = size;
+ return 1;
+ } else {
+ data[0] = 252+(size&0x3);
+ data[1] = (size-(int)data[0])>>2;
+ return 2;
+ }
+}
+
+static int parse_size(const unsigned char *data, opus_int32 len, opus_int16 *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;
+ }
+}
+
+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_parse_impl(const unsigned char *data, opus_int32 len,
+ int self_delimited, unsigned char *out_toc,
+ const unsigned char *frames[48], opus_int16 size[48],
+ int *payload_offset, opus_int32 *packet_offset)
+{
+ int i, bytes;
+ int count;
+ int cbr;
+ unsigned char ch, toc;
+ int framesize;
+ opus_int32 last_size;
+ opus_int32 pad = 0;
+ const unsigned char *data0 = data;
+
+ if (size==NULL || len<0)
+ return OPUS_BAD_ARG;
+ if (len==0)
+ return OPUS_INVALID_PACKET;
+
+ 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;
+ last_size = len/2;
+ /* If last_size doesn't fit in size[0], we'll catch it later */
+ size[0] = (opus_int16)last_size;
+ }
+ 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 p;
+ do {
+ int tmp;
+ if (len<=0)
+ return OPUS_INVALID_PACKET;
+ p = *data++;
+ len--;
+ tmp = p==255 ? 254: p;
+ len -= tmp;
+ pad += tmp;
+ } while (p==255);
+ }
+ 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] = (opus_int16)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 (bytes+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] = (opus_int16)last_size;
+ }
+
+ if (payload_offset)
+ *payload_offset = (int)(data-data0);
+
+ for (i=0;i<count;i++)
+ {
+ if (frames)
+ frames[i] = data;
+ data += size[i];
+ }
+
+ if (packet_offset)
+ *packet_offset = pad+(opus_int32)(data-data0);
+
+ if (out_toc)
+ *out_toc = toc;
+
+ return count;
+}
+
+int opus_packet_parse(const unsigned char *data, opus_int32 len,
+ unsigned char *out_toc, const unsigned char *frames[48],
+ opus_int16 size[48], int *payload_offset)
+{
+ return opus_packet_parse_impl(data, len, 0, out_toc,
+ frames, size, payload_offset, NULL);
+}
+
diff --git a/media/libopus/src/opus_decoder.c b/media/libopus/src/opus_decoder.c
new file mode 100644
index 000000000..080bec507
--- /dev/null
+++ b/media/libopus/src/opus_decoder.c
@@ -0,0 +1,981 @@
+/* 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 "config.h"
+#endif
+
+#ifndef OPUS_BUILD
+# error "OPUS_BUILD _MUST_ be defined to build Opus. This probably means you need other defines as well, as in a config.h. See the included build files for details."
+#endif
+
+#if defined(__GNUC__) && (__GNUC__ >= 2) && !defined(__OPTIMIZE__) && !defined(OPUS_WILL_BE_SLOW)
+# pragma message "You appear to be compiling without optimization, if so opus will be very slow."
+#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"
+#include "cpu_support.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;
+ int arch;
+
+ /* 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;
+ int last_packet_duration;
+#ifndef FIXED_POINT
+ opus_val16 softclip_mem[2];
+#endif
+
+ opus_uint32 rangeFinal;
+};
+
+
+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;
+ st->arch = opus_select_arch();
+ 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;
+ int pcm_silk_size;
+ VARDECL(opus_int16, pcm_silk);
+ int pcm_transition_silk_size;
+ VARDECL(opus_val16, pcm_transition_silk);
+ int pcm_transition_celt_size;
+ VARDECL(opus_val16, pcm_transition_celt);
+ opus_val16 *pcm_transition=NULL;
+ int redundant_audio_size;
+ 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;
+ int celt_accum;
+ 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;
+ mode = st->prev_mode;
+
+ if (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;
+ }
+
+ /* Avoids trying to run the PLC on sizes other than 2.5 (CELT), 5 (CELT),
+ 10, or 20 (e.g. 12.5 or 30 ms). */
+ if (audiosize > F20)
+ {
+ do {
+ int ret = opus_decode_frame(st, NULL, 0, pcm, IMIN(audiosize, F20), 0);
+ if (ret<0)
+ {
+ RESTORE_STACK;
+ return ret;
+ }
+ pcm += ret*st->channels;
+ audiosize -= ret;
+ } while (audiosize > 0);
+ RESTORE_STACK;
+ return frame_size;
+ } else if (audiosize < F20)
+ {
+ if (audiosize > F10)
+ audiosize = F10;
+ else if (mode != MODE_SILK_ONLY && audiosize > F5 && audiosize < F10)
+ audiosize = F5;
+ }
+ }
+
+ /* In fixed-point, we can tell CELT to do the accumulation on top of the
+ SILK PCM buffer. This saves some stack space. */
+#ifdef FIXED_POINT
+ celt_accum = (mode != MODE_CELT_ONLY) && (frame_size >= F10);
+#else
+ celt_accum = 0;
+#endif
+
+ pcm_transition_silk_size = ALLOC_NONE;
+ pcm_transition_celt_size = ALLOC_NONE;
+ 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;
+ /* Decide where to allocate the stack memory for pcm_transition */
+ if (mode == MODE_CELT_ONLY)
+ pcm_transition_celt_size = F5*st->channels;
+ else
+ pcm_transition_silk_size = F5*st->channels;
+ }
+ ALLOC(pcm_transition_celt, pcm_transition_celt_size, opus_val16);
+ if (transition && mode == MODE_CELT_ONLY)
+ {
+ pcm_transition = pcm_transition_celt;
+ 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;
+ }
+
+ /* Don't allocate any memory when in CELT-only mode */
+ pcm_silk_size = (mode != MODE_CELT_ONLY && !celt_accum) ? IMAX(F10, frame_size)*st->channels : ALLOC_NONE;
+ ALLOC(pcm_silk, pcm_silk_size, opus_int16);
+
+ /* SILK processing */
+ if (mode != MODE_CELT_ONLY)
+ {
+ int lost_flag, decoded_samples;
+ opus_int16 *pcm_ptr;
+#ifdef FIXED_POINT
+ if (celt_accum)
+ pcm_ptr = pcm;
+ else
+#endif
+ 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, st->arch );
+ 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_INTERNAL_ERROR;
+ }
+ }
+ 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;
+ pcm_transition_silk_size=ALLOC_NONE;
+ }
+
+ ALLOC(pcm_transition_silk, pcm_transition_silk_size, opus_val16);
+
+ if (transition && mode != MODE_CELT_ONLY)
+ {
+ pcm_transition = pcm_transition_silk;
+ opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0);
+ }
+
+ /* Only allocation memory for redundancy if/when needed */
+ redundant_audio_size = redundancy ? F5*st->channels : ALLOC_NONE;
+ ALLOC(redundant_audio, redundant_audio_size, opus_val16);
+
+ /* 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, 0);
+ 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, celt_accum);
+ } else {
+ unsigned char silence[2] = {0xFF, 0xFF};
+ if (!celt_accum)
+ {
+ 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, celt_accum);
+ }
+ }
+
+ if (mode != MODE_CELT_ONLY && !celt_accum)
+ {
+#ifdef FIXED_POINT
+ for (i=0;i<frame_size*st->channels;i++)
+ pcm[i] = SAT16(ADD32(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, 0);
+ 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;
+
+ if (celt_ret>=0)
+ {
+ if (OPUS_CHECK_ARRAY(pcm, audiosize*st->channels))
+ OPUS_PRINT_INT(audiosize);
+ }
+
+ RESTORE_STACK;
+ return celt_ret < 0 ? celt_ret : audiosize;
+
+}
+
+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, opus_int32 *packet_offset, int soft_clip)
+{
+ int i, nb_samples;
+ int count, offset;
+ unsigned char toc;
+ int packet_frame_size, packet_bandwidth, packet_mode, packet_stream_channels;
+ /* 48 x 2.5 ms = 120 ms */
+ opus_int16 size[48];
+ if (decode_fec<0 || decode_fec>1)
+ return OPUS_BAD_ARG;
+ /* For FEC/PLC, frame_size has to be to have a multiple of 2.5 ms */
+ if ((decode_fec || len==0 || data==NULL) && frame_size%(st->Fs/400)!=0)
+ return OPUS_BAD_ARG;
+ if (len==0 || data==NULL)
+ {
+ int pcm_count=0;
+ do {
+ int ret;
+ ret = opus_decode_frame(st, NULL, 0, pcm+pcm_count*st->channels, frame_size-pcm_count, 0);
+ if (ret<0)
+ return ret;
+ pcm_count += ret;
+ } while (pcm_count < frame_size);
+ celt_assert(pcm_count == frame_size);
+ if (OPUS_CHECK_ARRAY(pcm, pcm_count*st->channels))
+ OPUS_PRINT_INT(pcm_count);
+ st->last_packet_duration = pcm_count;
+ return pcm_count;
+ } else if (len<0)
+ return OPUS_BAD_ARG;
+
+ packet_mode = opus_packet_get_mode(data);
+ packet_bandwidth = opus_packet_get_bandwidth(data);
+ packet_frame_size = opus_packet_get_samples_per_frame(data, st->Fs);
+ packet_stream_channels = opus_packet_get_nb_channels(data);
+
+ count = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL,
+ size, &offset, packet_offset);
+ if (count<0)
+ return count;
+
+ data += offset;
+
+ if (decode_fec)
+ {
+ int duration_copy;
+ int ret;
+ /* If no FEC can be present, run the PLC (recursive call) */
+ if (frame_size < packet_frame_size || packet_mode == MODE_CELT_ONLY || st->mode == MODE_CELT_ONLY)
+ return opus_decode_native(st, NULL, 0, pcm, frame_size, 0, 0, NULL, soft_clip);
+ /* Otherwise, run the PLC on everything except the size for which we might have FEC */
+ duration_copy = st->last_packet_duration;
+ if (frame_size-packet_frame_size!=0)
+ {
+ ret = opus_decode_native(st, NULL, 0, pcm, frame_size-packet_frame_size, 0, 0, NULL, soft_clip);
+ if (ret<0)
+ {
+ st->last_packet_duration = duration_copy;
+ return ret;
+ }
+ celt_assert(ret==frame_size-packet_frame_size);
+ }
+ /* Complete with FEC */
+ st->mode = packet_mode;
+ st->bandwidth = packet_bandwidth;
+ st->frame_size = packet_frame_size;
+ st->stream_channels = packet_stream_channels;
+ ret = opus_decode_frame(st, data, size[0], pcm+st->channels*(frame_size-packet_frame_size),
+ packet_frame_size, 1);
+ if (ret<0)
+ return ret;
+ else {
+ if (OPUS_CHECK_ARRAY(pcm, frame_size*st->channels))
+ OPUS_PRINT_INT(frame_size);
+ st->last_packet_duration = frame_size;
+ return frame_size;
+ }
+ }
+
+ if (count*packet_frame_size > frame_size)
+ return OPUS_BUFFER_TOO_SMALL;
+
+ /* Update the state as the last step to avoid updating it on an invalid packet */
+ st->mode = packet_mode;
+ st->bandwidth = packet_bandwidth;
+ st->frame_size = packet_frame_size;
+ st->stream_channels = packet_stream_channels;
+
+ nb_samples=0;
+ for (i=0;i<count;i++)
+ {
+ int ret;
+ ret = opus_decode_frame(st, data, size[i], pcm+nb_samples*st->channels, frame_size-nb_samples, 0);
+ if (ret<0)
+ return ret;
+ celt_assert(ret==packet_frame_size);
+ data += size[i];
+ nb_samples += ret;
+ }
+ st->last_packet_duration = nb_samples;
+ if (OPUS_CHECK_ARRAY(pcm, nb_samples*st->channels))
+ OPUS_PRINT_INT(nb_samples);
+#ifndef FIXED_POINT
+ if (soft_clip)
+ opus_pcm_soft_clip(pcm, nb_samples, st->channels, st->softclip_mem);
+ else
+ st->softclip_mem[0]=st->softclip_mem[1]=0;
+#endif
+ 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)
+{
+ if(frame_size<=0)
+ return OPUS_BAD_ARG;
+ return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0);
+}
+
+#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;
+ int nb_samples;
+ ALLOC_STACK;
+
+ if(frame_size<=0)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+ if (data != NULL && len > 0 && !decode_fec)
+ {
+ nb_samples = opus_decoder_get_nb_samples(st, data, len);
+ if (nb_samples>0)
+ frame_size = IMIN(frame_size, nb_samples);
+ else
+ return OPUS_INVALID_PACKET;
+ }
+ ALLOC(out, frame_size*st->channels, opus_int16);
+
+ ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 0);
+ 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;
+ int nb_samples;
+ ALLOC_STACK;
+
+ if(frame_size<=0)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+
+ if (data != NULL && len > 0 && !decode_fec)
+ {
+ nb_samples = opus_decoder_get_nb_samples(st, data, len);
+ if (nb_samples>0)
+ frame_size = IMIN(frame_size, nb_samples);
+ else
+ return OPUS_INVALID_PACKET;
+ }
+ ALLOC(out, frame_size*st->channels, float);
+
+ ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 1);
+ 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)
+{
+ if(frame_size<=0)
+ return OPUS_BAD_ARG;
+ return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0);
+}
+
+#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*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->bandwidth;
+ }
+ break;
+ case OPUS_GET_FINAL_RANGE_REQUEST:
+ {
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *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)
+ {
+ goto bad_arg;
+ }
+ *value = st->Fs;
+ }
+ break;
+ case OPUS_GET_PITCH_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ 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)
+ {
+ goto bad_arg;
+ }
+ *value = st->decode_gain;
+ }
+ break;
+ case OPUS_SET_GAIN_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value<-32768 || value>32767)
+ {
+ goto bad_arg;
+ }
+ st->decode_gain = value;
+ }
+ break;
+ case OPUS_GET_LAST_PACKET_DURATION_REQUEST:
+ {
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->last_packet_duration;
+ }
+ break;
+ default:
+ /*fprintf(stderr, "unknown opus_decoder_ctl() request: %d", request);*/
+ ret = OPUS_UNIMPLEMENTED;
+ break;
+ }
+
+ va_end(ap);
+ return ret;
+bad_arg:
+ va_end(ap);
+ return OPUS_BAD_ARG;
+}
+
+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_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_packet_get_nb_samples(const unsigned char packet[], opus_int32 len,
+ opus_int32 Fs)
+{
+ 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, Fs);
+ /* Can't have more than 120 ms */
+ if (samples*25 > Fs*3)
+ return OPUS_INVALID_PACKET;
+ else
+ return samples;
+}
+
+int opus_decoder_get_nb_samples(const OpusDecoder *dec,
+ const unsigned char packet[], opus_int32 len)
+{
+ return opus_packet_get_nb_samples(packet, len, dec->Fs);
+}
diff --git a/media/libopus/src/opus_encoder.c b/media/libopus/src/opus_encoder.c
new file mode 100644
index 000000000..9a516a884
--- /dev/null
+++ b/media/libopus/src/opus_encoder.c
@@ -0,0 +1,2536 @@
+/* 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.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdarg.h>
+#include "celt.h"
+#include "entenc.h"
+#include "modes.h"
+#include "API.h"
+#include "stack_alloc.h"
+#include "float_cast.h"
+#include "opus.h"
+#include "arch.h"
+#include "pitch.h"
+#include "opus_private.h"
+#include "os_support.h"
+#include "cpu_support.h"
+#include "analysis.h"
+#include "mathops.h"
+#include "tuning_parameters.h"
+#ifdef FIXED_POINT
+#include "fixed/structs_FIX.h"
+#else
+#include "float/structs_FLP.h"
+#endif
+
+#define MAX_ENCODER_BUFFER 480
+
+typedef struct {
+ opus_val32 XX, XY, YY;
+ opus_val16 smoothed_width;
+ opus_val16 max_follower;
+} StereoWidthState;
+
+struct OpusEncoder {
+ int celt_enc_offset;
+ int silk_enc_offset;
+ silk_EncControlStruct silk_mode;
+ int application;
+ int channels;
+ int delay_compensation;
+ int force_channels;
+ int signal_type;
+ int user_bandwidth;
+ int max_bandwidth;
+ int user_forced_mode;
+ int voice_ratio;
+ opus_int32 Fs;
+ int use_vbr;
+ int vbr_constraint;
+ int variable_duration;
+ opus_int32 bitrate_bps;
+ opus_int32 user_bitrate_bps;
+ int lsb_depth;
+ int encoder_buffer;
+ int lfe;
+ int arch;
+#ifndef DISABLE_FLOAT_API
+ TonalityAnalysisState analysis;
+#endif
+
+#define OPUS_ENCODER_RESET_START stream_channels
+ int stream_channels;
+ opus_int16 hybrid_stereo_width_Q14;
+ opus_int32 variable_HP_smth2_Q15;
+ opus_val16 prev_HB_gain;
+ opus_val32 hp_mem[4];
+ int mode;
+ int prev_mode;
+ int prev_channels;
+ int prev_framesize;
+ int bandwidth;
+ int silk_bw_switch;
+ /* Sampling rate (at the API level) */
+ int first;
+ opus_val16 * energy_masking;
+ StereoWidthState width_mem;
+ opus_val16 delay_buffer[MAX_ENCODER_BUFFER*2];
+#ifndef DISABLE_FLOAT_API
+ int detected_bandwidth;
+#endif
+ opus_uint32 rangeFinal;
+};
+
+/* Transition tables for the voice and music. First column is the
+ middle (memoriless) threshold. The second column is the hysteresis
+ (difference with the middle) */
+static const opus_int32 mono_voice_bandwidth_thresholds[8] = {
+ 11000, 1000, /* NB<->MB */
+ 14000, 1000, /* MB<->WB */
+ 17000, 1000, /* WB<->SWB */
+ 21000, 2000, /* SWB<->FB */
+};
+static const opus_int32 mono_music_bandwidth_thresholds[8] = {
+ 12000, 1000, /* NB<->MB */
+ 15000, 1000, /* MB<->WB */
+ 18000, 2000, /* WB<->SWB */
+ 22000, 2000, /* SWB<->FB */
+};
+static const opus_int32 stereo_voice_bandwidth_thresholds[8] = {
+ 11000, 1000, /* NB<->MB */
+ 14000, 1000, /* MB<->WB */
+ 21000, 2000, /* WB<->SWB */
+ 28000, 2000, /* SWB<->FB */
+};
+static const opus_int32 stereo_music_bandwidth_thresholds[8] = {
+ 12000, 1000, /* NB<->MB */
+ 18000, 2000, /* MB<->WB */
+ 21000, 2000, /* WB<->SWB */
+ 30000, 2000, /* SWB<->FB */
+};
+/* Threshold bit-rates for switching between mono and stereo */
+static const opus_int32 stereo_voice_threshold = 30000;
+static const opus_int32 stereo_music_threshold = 30000;
+
+/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
+static const opus_int32 mode_thresholds[2][2] = {
+ /* voice */ /* music */
+ { 64000, 16000}, /* mono */
+ { 36000, 16000}, /* stereo */
+};
+
+int opus_encoder_get_size(int channels)
+{
+ int silkEncSizeBytes, celtEncSizeBytes;
+ int ret;
+ if (channels<1 || channels > 2)
+ return 0;
+ ret = silk_Get_Encoder_Size( &silkEncSizeBytes );
+ if (ret)
+ return 0;
+ silkEncSizeBytes = align(silkEncSizeBytes);
+ celtEncSizeBytes = celt_encoder_get_size(channels);
+ return align(sizeof(OpusEncoder))+silkEncSizeBytes+celtEncSizeBytes;
+}
+
+int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application)
+{
+ void *silk_enc;
+ CELTEncoder *celt_enc;
+ int err;
+ int ret, silkEncSizeBytes;
+
+ if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)||
+ (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO
+ && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY))
+ return OPUS_BAD_ARG;
+
+ OPUS_CLEAR((char*)st, opus_encoder_get_size(channels));
+ /* Create SILK encoder */
+ ret = silk_Get_Encoder_Size( &silkEncSizeBytes );
+ if (ret)
+ return OPUS_BAD_ARG;
+ silkEncSizeBytes = align(silkEncSizeBytes);
+ st->silk_enc_offset = align(sizeof(OpusEncoder));
+ st->celt_enc_offset = st->silk_enc_offset+silkEncSizeBytes;
+ silk_enc = (char*)st+st->silk_enc_offset;
+ celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+
+ st->stream_channels = st->channels = channels;
+
+ st->Fs = Fs;
+
+ st->arch = opus_select_arch();
+
+ ret = silk_InitEncoder( silk_enc, st->arch, &st->silk_mode );
+ if(ret)return OPUS_INTERNAL_ERROR;
+
+ /* default SILK parameters */
+ st->silk_mode.nChannelsAPI = channels;
+ st->silk_mode.nChannelsInternal = channels;
+ st->silk_mode.API_sampleRate = st->Fs;
+ st->silk_mode.maxInternalSampleRate = 16000;
+ st->silk_mode.minInternalSampleRate = 8000;
+ st->silk_mode.desiredInternalSampleRate = 16000;
+ st->silk_mode.payloadSize_ms = 20;
+ st->silk_mode.bitRate = 25000;
+ st->silk_mode.packetLossPercentage = 0;
+ st->silk_mode.complexity = 9;
+ st->silk_mode.useInBandFEC = 0;
+ st->silk_mode.useDTX = 0;
+ st->silk_mode.useCBR = 0;
+ st->silk_mode.reducedDependency = 0;
+
+ /* Create CELT encoder */
+ /* Initialize CELT encoder */
+ err = celt_encoder_init(celt_enc, Fs, channels, st->arch);
+ if(err!=OPUS_OK)return OPUS_INTERNAL_ERROR;
+
+ celt_encoder_ctl(celt_enc, CELT_SET_SIGNALLING(0));
+ celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(st->silk_mode.complexity));
+
+ st->use_vbr = 1;
+ /* Makes constrained VBR the default (safer for real-time use) */
+ st->vbr_constraint = 1;
+ st->user_bitrate_bps = OPUS_AUTO;
+ st->bitrate_bps = 3000+Fs*channels;
+ st->application = application;
+ st->signal_type = OPUS_AUTO;
+ st->user_bandwidth = OPUS_AUTO;
+ st->max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+ st->force_channels = OPUS_AUTO;
+ st->user_forced_mode = OPUS_AUTO;
+ st->voice_ratio = -1;
+ st->encoder_buffer = st->Fs/100;
+ st->lsb_depth = 24;
+ st->variable_duration = OPUS_FRAMESIZE_ARG;
+
+ /* Delay compensation of 4 ms (2.5 ms for SILK's extra look-ahead
+ + 1.5 ms for SILK resamplers and stereo prediction) */
+ st->delay_compensation = st->Fs/250;
+
+ st->hybrid_stereo_width_Q14 = 1 << 14;
+ st->prev_HB_gain = Q15ONE;
+ st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+ st->first = 1;
+ st->mode = MODE_HYBRID;
+ st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+#ifndef DISABLE_FLOAT_API
+ tonality_analysis_init(&st->analysis);
+#endif
+
+ return OPUS_OK;
+}
+
+static unsigned char gen_toc(int mode, int framerate, int bandwidth, int channels)
+{
+ int period;
+ unsigned char toc;
+ period = 0;
+ while (framerate < 400)
+ {
+ framerate <<= 1;
+ period++;
+ }
+ if (mode == MODE_SILK_ONLY)
+ {
+ toc = (bandwidth-OPUS_BANDWIDTH_NARROWBAND)<<5;
+ toc |= (period-2)<<3;
+ } else if (mode == MODE_CELT_ONLY)
+ {
+ int tmp = bandwidth-OPUS_BANDWIDTH_MEDIUMBAND;
+ if (tmp < 0)
+ tmp = 0;
+ toc = 0x80;
+ toc |= tmp << 5;
+ toc |= period<<3;
+ } else /* Hybrid */
+ {
+ toc = 0x60;
+ toc |= (bandwidth-OPUS_BANDWIDTH_SUPERWIDEBAND)<<4;
+ toc |= (period-2)<<3;
+ }
+ toc |= (channels==2)<<2;
+ return toc;
+}
+
+#ifndef FIXED_POINT
+static void silk_biquad_float(
+ const opus_val16 *in, /* I: Input signal */
+ const opus_int32 *B_Q28, /* I: MA coefficients [3] */
+ const opus_int32 *A_Q28, /* I: AR coefficients [2] */
+ opus_val32 *S, /* I/O: State vector [2] */
+ opus_val16 *out, /* O: Output signal */
+ const opus_int32 len, /* I: Signal length (must be even) */
+ int stride
+)
+{
+ /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */
+ opus_int k;
+ opus_val32 vout;
+ opus_val32 inval;
+ opus_val32 A[2], B[3];
+
+ A[0] = (opus_val32)(A_Q28[0] * (1.f/((opus_int32)1<<28)));
+ A[1] = (opus_val32)(A_Q28[1] * (1.f/((opus_int32)1<<28)));
+ B[0] = (opus_val32)(B_Q28[0] * (1.f/((opus_int32)1<<28)));
+ B[1] = (opus_val32)(B_Q28[1] * (1.f/((opus_int32)1<<28)));
+ B[2] = (opus_val32)(B_Q28[2] * (1.f/((opus_int32)1<<28)));
+
+ /* Negate A_Q28 values and split in two parts */
+
+ for( k = 0; k < len; k++ ) {
+ /* S[ 0 ], S[ 1 ]: Q12 */
+ inval = in[ k*stride ];
+ vout = S[ 0 ] + B[0]*inval;
+
+ S[ 0 ] = S[1] - vout*A[0] + B[1]*inval;
+
+ S[ 1 ] = - vout*A[1] + B[2]*inval + VERY_SMALL;
+
+ /* Scale back to Q0 and saturate */
+ out[ k*stride ] = vout;
+ }
+}
+#endif
+
+static void hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{
+ opus_int32 B_Q28[ 3 ], A_Q28[ 2 ];
+ opus_int32 Fc_Q19, r_Q28, r_Q22;
+
+ silk_assert( cutoff_Hz <= silk_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) );
+ Fc_Q19 = silk_DIV32_16( silk_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), Fs/1000 );
+ silk_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 );
+
+ r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - silk_MUL( SILK_FIX_CONST( 0.92, 9 ), Fc_Q19 );
+
+ /* b = r * [ 1; -2; 1 ]; */
+ /* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */
+ B_Q28[ 0 ] = r_Q28;
+ B_Q28[ 1 ] = silk_LSHIFT( -r_Q28, 1 );
+ B_Q28[ 2 ] = r_Q28;
+
+ /* -r * ( 2 - Fc * Fc ); */
+ r_Q22 = silk_RSHIFT( r_Q28, 6 );
+ A_Q28[ 0 ] = silk_SMULWW( r_Q22, silk_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0, 22 ) );
+ A_Q28[ 1 ] = silk_SMULWW( r_Q22, r_Q22 );
+
+#ifdef FIXED_POINT
+ silk_biquad_alt( in, B_Q28, A_Q28, hp_mem, out, len, channels );
+ if( channels == 2 ) {
+ silk_biquad_alt( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels );
+ }
+#else
+ silk_biquad_float( in, B_Q28, A_Q28, hp_mem, out, len, channels );
+ if( channels == 2 ) {
+ silk_biquad_float( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels );
+ }
+#endif
+}
+
+#ifdef FIXED_POINT
+static void dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{
+ int c, i;
+ int shift;
+
+ /* Approximates -round(log2(4.*cutoff_Hz/Fs)) */
+ shift=celt_ilog2(Fs/(cutoff_Hz*3));
+ for (c=0;c<channels;c++)
+ {
+ for (i=0;i<len;i++)
+ {
+ opus_val32 x, tmp, y;
+ x = SHL32(EXTEND32(in[channels*i+c]), 15);
+ /* First stage */
+ tmp = x-hp_mem[2*c];
+ hp_mem[2*c] = hp_mem[2*c] + PSHR32(x - hp_mem[2*c], shift);
+ /* Second stage */
+ y = tmp - hp_mem[2*c+1];
+ hp_mem[2*c+1] = hp_mem[2*c+1] + PSHR32(tmp - hp_mem[2*c+1], shift);
+ out[channels*i+c] = EXTRACT16(SATURATE(PSHR32(y, 15), 32767));
+ }
+ }
+}
+
+#else
+static void dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{
+ int c, i;
+ float coef;
+
+ coef = 4.0f*cutoff_Hz/Fs;
+ for (c=0;c<channels;c++)
+ {
+ for (i=0;i<len;i++)
+ {
+ opus_val32 x, tmp, y;
+ x = in[channels*i+c];
+ /* First stage */
+ tmp = x-hp_mem[2*c];
+ hp_mem[2*c] = hp_mem[2*c] + coef*(x - hp_mem[2*c]) + VERY_SMALL;
+ /* Second stage */
+ y = tmp - hp_mem[2*c+1];
+ hp_mem[2*c+1] = hp_mem[2*c+1] + coef*(tmp - hp_mem[2*c+1]) + VERY_SMALL;
+ out[channels*i+c] = y;
+ }
+ }
+}
+#endif
+
+static void stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2,
+ int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs)
+{
+ int i;
+ int overlap;
+ int inc;
+ inc = 48000/Fs;
+ overlap=overlap48/inc;
+ g1 = Q15ONE-g1;
+ g2 = Q15ONE-g2;
+ for (i=0;i<overlap;i++)
+ {
+ opus_val32 diff;
+ opus_val16 g, w;
+ w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+ g = SHR32(MAC16_16(MULT16_16(w,g2),
+ Q15ONE-w, g1), 15);
+ diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1]));
+ diff = MULT16_16_Q15(g, diff);
+ out[i*channels] = out[i*channels] - diff;
+ out[i*channels+1] = out[i*channels+1] + diff;
+ }
+ for (;i<frame_size;i++)
+ {
+ opus_val32 diff;
+ diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1]));
+ diff = MULT16_16_Q15(g2, diff);
+ out[i*channels] = out[i*channels] - diff;
+ out[i*channels+1] = out[i*channels+1] + diff;
+ }
+}
+
+static void gain_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2,
+ int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs)
+{
+ int i;
+ int inc;
+ int overlap;
+ int c;
+ inc = 48000/Fs;
+ overlap=overlap48/inc;
+ if (channels==1)
+ {
+ for (i=0;i<overlap;i++)
+ {
+ opus_val16 g, w;
+ w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+ g = SHR32(MAC16_16(MULT16_16(w,g2),
+ Q15ONE-w, g1), 15);
+ out[i] = MULT16_16_Q15(g, in[i]);
+ }
+ } else {
+ for (i=0;i<overlap;i++)
+ {
+ opus_val16 g, w;
+ w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+ g = SHR32(MAC16_16(MULT16_16(w,g2),
+ Q15ONE-w, g1), 15);
+ out[i*2] = MULT16_16_Q15(g, in[i*2]);
+ out[i*2+1] = MULT16_16_Q15(g, in[i*2+1]);
+ }
+ }
+ c=0;do {
+ for (i=overlap;i<frame_size;i++)
+ {
+ out[i*channels+c] = MULT16_16_Q15(g2, in[i*channels+c]);
+ }
+ }
+ while (++c<channels);
+}
+
+OpusEncoder *opus_encoder_create(opus_int32 Fs, int channels, int application, int *error)
+{
+ int ret;
+ OpusEncoder *st;
+ if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)||
+ (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO
+ && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY))
+ {
+ if (error)
+ *error = OPUS_BAD_ARG;
+ return NULL;
+ }
+ st = (OpusEncoder *)opus_alloc(opus_encoder_get_size(channels));
+ if (st == NULL)
+ {
+ if (error)
+ *error = OPUS_ALLOC_FAIL;
+ return NULL;
+ }
+ ret = opus_encoder_init(st, Fs, channels, application);
+ if (error)
+ *error = ret;
+ if (ret != OPUS_OK)
+ {
+ opus_free(st);
+ st = NULL;
+ }
+ return st;
+}
+
+static opus_int32 user_bitrate_to_bitrate(OpusEncoder *st, int frame_size, int max_data_bytes)
+{
+ if(!frame_size)frame_size=st->Fs/400;
+ if (st->user_bitrate_bps==OPUS_AUTO)
+ return 60*st->Fs/frame_size + st->Fs*st->channels;
+ else if (st->user_bitrate_bps==OPUS_BITRATE_MAX)
+ return max_data_bytes*8*st->Fs/frame_size;
+ else
+ return st->user_bitrate_bps;
+}
+
+#ifndef DISABLE_FLOAT_API
+/* Don't use more than 60 ms for the frame size analysis */
+#define MAX_DYNAMIC_FRAMESIZE 24
+/* Estimates how much the bitrate will be boosted based on the sub-frame energy */
+static float transient_boost(const float *E, const float *E_1, int LM, int maxM)
+{
+ int i;
+ int M;
+ float sumE=0, sumE_1=0;
+ float metric;
+
+ M = IMIN(maxM, (1<<LM)+1);
+ for (i=0;i<M;i++)
+ {
+ sumE += E[i];
+ sumE_1 += E_1[i];
+ }
+ metric = sumE*sumE_1/(M*M);
+ /*if (LM==3)
+ printf("%f\n", metric);*/
+ /*return metric>10 ? 1 : 0;*/
+ /*return MAX16(0,1-exp(-.25*(metric-2.)));*/
+ return MIN16(1,(float)sqrt(MAX16(0,.05f*(metric-2))));
+}
+
+/* Viterbi decoding trying to find the best frame size combination using look-ahead
+
+ State numbering:
+ 0: unused
+ 1: 2.5 ms
+ 2: 5 ms (#1)
+ 3: 5 ms (#2)
+ 4: 10 ms (#1)
+ 5: 10 ms (#2)
+ 6: 10 ms (#3)
+ 7: 10 ms (#4)
+ 8: 20 ms (#1)
+ 9: 20 ms (#2)
+ 10: 20 ms (#3)
+ 11: 20 ms (#4)
+ 12: 20 ms (#5)
+ 13: 20 ms (#6)
+ 14: 20 ms (#7)
+ 15: 20 ms (#8)
+*/
+static int transient_viterbi(const float *E, const float *E_1, int N, int frame_cost, int rate)
+{
+ int i;
+ float cost[MAX_DYNAMIC_FRAMESIZE][16];
+ int states[MAX_DYNAMIC_FRAMESIZE][16];
+ float best_cost;
+ int best_state;
+ float factor;
+ /* Take into account that we damp VBR in the 32 kb/s to 64 kb/s range. */
+ if (rate<80)
+ factor=0;
+ else if (rate>160)
+ factor=1;
+ else
+ factor = (rate-80.f)/80.f;
+ /* Makes variable framesize less aggressive at lower bitrates, but I can't
+ find any valid theoretical justification for this (other than it seems
+ to help) */
+ for (i=0;i<16;i++)
+ {
+ /* Impossible state */
+ states[0][i] = -1;
+ cost[0][i] = 1e10;
+ }
+ for (i=0;i<4;i++)
+ {
+ cost[0][1<<i] = (frame_cost + rate*(1<<i))*(1+factor*transient_boost(E, E_1, i, N+1));
+ states[0][1<<i] = i;
+ }
+ for (i=1;i<N;i++)
+ {
+ int j;
+
+ /* Follow continuations */
+ for (j=2;j<16;j++)
+ {
+ cost[i][j] = cost[i-1][j-1];
+ states[i][j] = j-1;
+ }
+
+ /* New frames */
+ for(j=0;j<4;j++)
+ {
+ int k;
+ float min_cost;
+ float curr_cost;
+ states[i][1<<j] = 1;
+ min_cost = cost[i-1][1];
+ for(k=1;k<4;k++)
+ {
+ float tmp = cost[i-1][(1<<(k+1))-1];
+ if (tmp < min_cost)
+ {
+ states[i][1<<j] = (1<<(k+1))-1;
+ min_cost = tmp;
+ }
+ }
+ curr_cost = (frame_cost + rate*(1<<j))*(1+factor*transient_boost(E+i, E_1+i, j, N-i+1));
+ cost[i][1<<j] = min_cost;
+ /* If part of the frame is outside the analysis window, only count part of the cost */
+ if (N-i < (1<<j))
+ cost[i][1<<j] += curr_cost*(float)(N-i)/(1<<j);
+ else
+ cost[i][1<<j] += curr_cost;
+ }
+ }
+
+ best_state=1;
+ best_cost = cost[N-1][1];
+ /* Find best end state (doesn't force a frame to end at N-1) */
+ for (i=2;i<16;i++)
+ {
+ if (cost[N-1][i]<best_cost)
+ {
+ best_cost = cost[N-1][i];
+ best_state = i;
+ }
+ }
+
+ /* Follow transitions back */
+ for (i=N-1;i>=0;i--)
+ {
+ /*printf("%d ", best_state);*/
+ best_state = states[i][best_state];
+ }
+ /*printf("%d\n", best_state);*/
+ return best_state;
+}
+
+static int optimize_framesize(const void *x, int len, int C, opus_int32 Fs,
+ int bitrate, opus_val16 tonality, float *mem, int buffering,
+ downmix_func downmix)
+{
+ int N;
+ int i;
+ float e[MAX_DYNAMIC_FRAMESIZE+4];
+ float e_1[MAX_DYNAMIC_FRAMESIZE+3];
+ opus_val32 memx;
+ int bestLM=0;
+ int subframe;
+ int pos;
+ int offset;
+ VARDECL(opus_val32, sub);
+
+ subframe = Fs/400;
+ ALLOC(sub, subframe, opus_val32);
+ e[0]=mem[0];
+ e_1[0]=1.f/(EPSILON+mem[0]);
+ if (buffering)
+ {
+ /* Consider the CELT delay when not in restricted-lowdelay */
+ /* We assume the buffering is between 2.5 and 5 ms */
+ offset = 2*subframe - buffering;
+ celt_assert(offset>=0 && offset <= subframe);
+ len -= offset;
+ e[1]=mem[1];
+ e_1[1]=1.f/(EPSILON+mem[1]);
+ e[2]=mem[2];
+ e_1[2]=1.f/(EPSILON+mem[2]);
+ pos = 3;
+ } else {
+ pos=1;
+ offset=0;
+ }
+ N=IMIN(len/subframe, MAX_DYNAMIC_FRAMESIZE);
+ /* Just silencing a warning, it's really initialized later */
+ memx = 0;
+ for (i=0;i<N;i++)
+ {
+ float tmp;
+ opus_val32 tmpx;
+ int j;
+ tmp=EPSILON;
+
+ downmix(x, sub, subframe, i*subframe+offset, 0, -2, C);
+ if (i==0)
+ memx = sub[0];
+ for (j=0;j<subframe;j++)
+ {
+ tmpx = sub[j];
+ tmp += (tmpx-memx)*(float)(tmpx-memx);
+ memx = tmpx;
+ }
+ e[i+pos] = tmp;
+ e_1[i+pos] = 1.f/tmp;
+ }
+ /* Hack to get 20 ms working with APPLICATION_AUDIO
+ The real problem is that the corresponding memory needs to use 1.5 ms
+ from this frame and 1 ms from the next frame */
+ e[i+pos] = e[i+pos-1];
+ if (buffering)
+ N=IMIN(MAX_DYNAMIC_FRAMESIZE, N+2);
+ bestLM = transient_viterbi(e, e_1, N, (int)((1.f+.5f*tonality)*(60*C+40)), bitrate/400);
+ mem[0] = e[1<<bestLM];
+ if (buffering)
+ {
+ mem[1] = e[(1<<bestLM)+1];
+ mem[2] = e[(1<<bestLM)+2];
+ }
+ return bestLM;
+}
+
+#endif
+
+#ifndef DISABLE_FLOAT_API
+#ifdef FIXED_POINT
+#define PCM2VAL(x) FLOAT2INT16(x)
+#else
+#define PCM2VAL(x) SCALEIN(x)
+#endif
+void downmix_float(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C)
+{
+ const float *x;
+ opus_val32 scale;
+ int j;
+ x = (const float *)_x;
+ for (j=0;j<subframe;j++)
+ sub[j] = PCM2VAL(x[(j+offset)*C+c1]);
+ if (c2>-1)
+ {
+ for (j=0;j<subframe;j++)
+ sub[j] += PCM2VAL(x[(j+offset)*C+c2]);
+ } else if (c2==-2)
+ {
+ int c;
+ for (c=1;c<C;c++)
+ {
+ for (j=0;j<subframe;j++)
+ sub[j] += PCM2VAL(x[(j+offset)*C+c]);
+ }
+ }
+#ifdef FIXED_POINT
+ scale = (1<<SIG_SHIFT);
+#else
+ scale = 1.f;
+#endif
+ if (C==-2)
+ scale /= C;
+ else
+ scale /= 2;
+ for (j=0;j<subframe;j++)
+ sub[j] *= scale;
+}
+#endif
+
+void downmix_int(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C)
+{
+ const opus_int16 *x;
+ opus_val32 scale;
+ int j;
+ x = (const opus_int16 *)_x;
+ for (j=0;j<subframe;j++)
+ sub[j] = x[(j+offset)*C+c1];
+ if (c2>-1)
+ {
+ for (j=0;j<subframe;j++)
+ sub[j] += x[(j+offset)*C+c2];
+ } else if (c2==-2)
+ {
+ int c;
+ for (c=1;c<C;c++)
+ {
+ for (j=0;j<subframe;j++)
+ sub[j] += x[(j+offset)*C+c];
+ }
+ }
+#ifdef FIXED_POINT
+ scale = (1<<SIG_SHIFT);
+#else
+ scale = 1.f/32768;
+#endif
+ if (C==-2)
+ scale /= C;
+ else
+ scale /= 2;
+ for (j=0;j<subframe;j++)
+ sub[j] *= scale;
+}
+
+opus_int32 frame_size_select(opus_int32 frame_size, int variable_duration, opus_int32 Fs)
+{
+ int new_size;
+ if (frame_size<Fs/400)
+ return -1;
+ if (variable_duration == OPUS_FRAMESIZE_ARG)
+ new_size = frame_size;
+ else if (variable_duration == OPUS_FRAMESIZE_VARIABLE)
+ new_size = Fs/50;
+ else if (variable_duration >= OPUS_FRAMESIZE_2_5_MS && variable_duration <= OPUS_FRAMESIZE_60_MS)
+ new_size = IMIN(3*Fs/50, (Fs/400)<<(variable_duration-OPUS_FRAMESIZE_2_5_MS));
+ else
+ return -1;
+ if (new_size>frame_size)
+ return -1;
+ if (400*new_size!=Fs && 200*new_size!=Fs && 100*new_size!=Fs &&
+ 50*new_size!=Fs && 25*new_size!=Fs && 50*new_size!=3*Fs)
+ return -1;
+ return new_size;
+}
+
+opus_int32 compute_frame_size(const void *analysis_pcm, int frame_size,
+ int variable_duration, int C, opus_int32 Fs, int bitrate_bps,
+ int delay_compensation, downmix_func downmix
+#ifndef DISABLE_FLOAT_API
+ , float *subframe_mem
+#endif
+ )
+{
+#ifndef DISABLE_FLOAT_API
+ if (variable_duration == OPUS_FRAMESIZE_VARIABLE && frame_size >= Fs/200)
+ {
+ int LM = 3;
+ LM = optimize_framesize(analysis_pcm, frame_size, C, Fs, bitrate_bps,
+ 0, subframe_mem, delay_compensation, downmix);
+ while ((Fs/400<<LM)>frame_size)
+ LM--;
+ frame_size = (Fs/400<<LM);
+ } else
+#else
+ (void)analysis_pcm;
+ (void)C;
+ (void)bitrate_bps;
+ (void)delay_compensation;
+ (void)downmix;
+#endif
+ {
+ frame_size = frame_size_select(frame_size, variable_duration, Fs);
+ }
+ if (frame_size<0)
+ return -1;
+ return frame_size;
+}
+
+opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int32 Fs, StereoWidthState *mem)
+{
+ opus_val32 xx, xy, yy;
+ opus_val16 sqrt_xx, sqrt_yy;
+ opus_val16 qrrt_xx, qrrt_yy;
+ int frame_rate;
+ int i;
+ opus_val16 short_alpha;
+
+ frame_rate = Fs/frame_size;
+ short_alpha = Q15ONE - MULT16_16(25, Q15ONE)/IMAX(50,frame_rate);
+ xx=xy=yy=0;
+ /* Unroll by 4. The frame size is always a multiple of 4 *except* for
+ 2.5 ms frames at 12 kHz. Since this setting is very rare (and very
+ stupid), we just discard the last two samples. */
+ for (i=0;i<frame_size-3;i+=4)
+ {
+ opus_val32 pxx=0;
+ opus_val32 pxy=0;
+ opus_val32 pyy=0;
+ opus_val16 x, y;
+ x = pcm[2*i];
+ y = pcm[2*i+1];
+ pxx = SHR32(MULT16_16(x,x),2);
+ pxy = SHR32(MULT16_16(x,y),2);
+ pyy = SHR32(MULT16_16(y,y),2);
+ x = pcm[2*i+2];
+ y = pcm[2*i+3];
+ pxx += SHR32(MULT16_16(x,x),2);
+ pxy += SHR32(MULT16_16(x,y),2);
+ pyy += SHR32(MULT16_16(y,y),2);
+ x = pcm[2*i+4];
+ y = pcm[2*i+5];
+ pxx += SHR32(MULT16_16(x,x),2);
+ pxy += SHR32(MULT16_16(x,y),2);
+ pyy += SHR32(MULT16_16(y,y),2);
+ x = pcm[2*i+6];
+ y = pcm[2*i+7];
+ pxx += SHR32(MULT16_16(x,x),2);
+ pxy += SHR32(MULT16_16(x,y),2);
+ pyy += SHR32(MULT16_16(y,y),2);
+
+ xx += SHR32(pxx, 10);
+ xy += SHR32(pxy, 10);
+ yy += SHR32(pyy, 10);
+ }
+ mem->XX += MULT16_32_Q15(short_alpha, xx-mem->XX);
+ mem->XY += MULT16_32_Q15(short_alpha, xy-mem->XY);
+ mem->YY += MULT16_32_Q15(short_alpha, yy-mem->YY);
+ mem->XX = MAX32(0, mem->XX);
+ mem->XY = MAX32(0, mem->XY);
+ mem->YY = MAX32(0, mem->YY);
+ if (MAX32(mem->XX, mem->YY)>QCONST16(8e-4f, 18))
+ {
+ opus_val16 corr;
+ opus_val16 ldiff;
+ opus_val16 width;
+ sqrt_xx = celt_sqrt(mem->XX);
+ sqrt_yy = celt_sqrt(mem->YY);
+ qrrt_xx = celt_sqrt(sqrt_xx);
+ qrrt_yy = celt_sqrt(sqrt_yy);
+ /* Inter-channel correlation */
+ mem->XY = MIN32(mem->XY, sqrt_xx*sqrt_yy);
+ corr = SHR32(frac_div32(mem->XY,EPSILON+MULT16_16(sqrt_xx,sqrt_yy)),16);
+ /* Approximate loudness difference */
+ ldiff = MULT16_16(Q15ONE, ABS16(qrrt_xx-qrrt_yy))/(EPSILON+qrrt_xx+qrrt_yy);
+ width = MULT16_16_Q15(celt_sqrt(QCONST32(1.f,30)-MULT16_16(corr,corr)), ldiff);
+ /* Smoothing over one second */
+ mem->smoothed_width += (width-mem->smoothed_width)/frame_rate;
+ /* Peak follower */
+ mem->max_follower = MAX16(mem->max_follower-QCONST16(.02f,15)/frame_rate, mem->smoothed_width);
+ }
+ /*printf("%f %f %f %f %f ", corr/(float)Q15ONE, ldiff/(float)Q15ONE, width/(float)Q15ONE, mem->smoothed_width/(float)Q15ONE, mem->max_follower/(float)Q15ONE);*/
+ return EXTRACT16(MIN32(Q15ONE, MULT16_16(20, mem->max_follower)));
+}
+
+opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
+ unsigned char *data, opus_int32 out_data_bytes, int lsb_depth,
+ const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2,
+ int analysis_channels, downmix_func downmix, int float_api)
+{
+ void *silk_enc;
+ CELTEncoder *celt_enc;
+ int i;
+ int ret=0;
+ opus_int32 nBytes;
+ ec_enc enc;
+ int bytes_target;
+ int prefill=0;
+ int start_band = 0;
+ int redundancy = 0;
+ int redundancy_bytes = 0; /* Number of bytes to use for redundancy frame */
+ int celt_to_silk = 0;
+ VARDECL(opus_val16, pcm_buf);
+ int nb_compr_bytes;
+ int to_celt = 0;
+ opus_uint32 redundant_rng = 0;
+ int cutoff_Hz, hp_freq_smth1;
+ int voice_est; /* Probability of voice in Q7 */
+ opus_int32 equiv_rate;
+ int delay_compensation;
+ int frame_rate;
+ opus_int32 max_rate; /* Max bitrate we're allowed to use */
+ int curr_bandwidth;
+ opus_val16 HB_gain;
+ opus_int32 max_data_bytes; /* Max number of bytes we're allowed to use */
+ int total_buffer;
+ opus_val16 stereo_width;
+ const CELTMode *celt_mode;
+#ifndef DISABLE_FLOAT_API
+ AnalysisInfo analysis_info;
+ int analysis_read_pos_bak=-1;
+ int analysis_read_subframe_bak=-1;
+#endif
+ VARDECL(opus_val16, tmp_prefill);
+
+ ALLOC_STACK;
+
+ max_data_bytes = IMIN(1276, out_data_bytes);
+
+ st->rangeFinal = 0;
+ if ((!st->variable_duration && 400*frame_size != st->Fs && 200*frame_size != st->Fs && 100*frame_size != st->Fs &&
+ 50*frame_size != st->Fs && 25*frame_size != st->Fs && 50*frame_size != 3*st->Fs)
+ || (400*frame_size < st->Fs)
+ || max_data_bytes<=0
+ )
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+ silk_enc = (char*)st+st->silk_enc_offset;
+ celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ delay_compensation = 0;
+ else
+ delay_compensation = st->delay_compensation;
+
+ lsb_depth = IMIN(lsb_depth, st->lsb_depth);
+
+ celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode));
+#ifndef DISABLE_FLOAT_API
+ analysis_info.valid = 0;
+#ifdef FIXED_POINT
+ if (st->silk_mode.complexity >= 10 && st->Fs==48000)
+#else
+ if (st->silk_mode.complexity >= 7 && st->Fs==48000)
+#endif
+ {
+ analysis_read_pos_bak = st->analysis.read_pos;
+ analysis_read_subframe_bak = st->analysis.read_subframe;
+ run_analysis(&st->analysis, celt_mode, analysis_pcm, analysis_size, frame_size,
+ c1, c2, analysis_channels, st->Fs,
+ lsb_depth, downmix, &analysis_info);
+ }
+#else
+ (void)analysis_pcm;
+ (void)analysis_size;
+#endif
+
+ st->voice_ratio = -1;
+
+#ifndef DISABLE_FLOAT_API
+ st->detected_bandwidth = 0;
+ if (analysis_info.valid)
+ {
+ int analysis_bandwidth;
+ if (st->signal_type == OPUS_AUTO)
+ st->voice_ratio = (int)floor(.5+100*(1-analysis_info.music_prob));
+
+ analysis_bandwidth = analysis_info.bandwidth;
+ if (analysis_bandwidth<=12)
+ st->detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+ else if (analysis_bandwidth<=14)
+ st->detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+ else if (analysis_bandwidth<=16)
+ st->detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ else if (analysis_bandwidth<=18)
+ st->detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+ else
+ st->detected_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+ }
+#endif
+
+ if (st->channels==2 && st->force_channels!=1)
+ stereo_width = compute_stereo_width(pcm, frame_size, st->Fs, &st->width_mem);
+ else
+ stereo_width = 0;
+ total_buffer = delay_compensation;
+ st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes);
+
+ frame_rate = st->Fs/frame_size;
+ if (!st->use_vbr)
+ {
+ int cbrBytes;
+ /* Multiply by 3 to make sure the division is exact. */
+ int frame_rate3 = 3*st->Fs/frame_size;
+ /* We need to make sure that "int" values always fit in 16 bits. */
+ cbrBytes = IMIN( (3*st->bitrate_bps/8 + frame_rate3/2)/frame_rate3, max_data_bytes);
+ st->bitrate_bps = cbrBytes*(opus_int32)frame_rate3*8/3;
+ max_data_bytes = cbrBytes;
+ }
+ if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8
+ || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400)))
+ {
+ /*If the space is too low to do something useful, emit 'PLC' frames.*/
+ int tocmode = st->mode;
+ int bw = st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth;
+ if (tocmode==0)
+ tocmode = MODE_SILK_ONLY;
+ if (frame_rate>100)
+ tocmode = MODE_CELT_ONLY;
+ if (frame_rate < 50)
+ tocmode = MODE_SILK_ONLY;
+ if(tocmode==MODE_SILK_ONLY&&bw>OPUS_BANDWIDTH_WIDEBAND)
+ bw=OPUS_BANDWIDTH_WIDEBAND;
+ else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND)
+ bw=OPUS_BANDWIDTH_NARROWBAND;
+ else if (tocmode==MODE_HYBRID&&bw<=OPUS_BANDWIDTH_SUPERWIDEBAND)
+ bw=OPUS_BANDWIDTH_SUPERWIDEBAND;
+ data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels);
+ ret = 1;
+ if (!st->use_vbr)
+ {
+ ret = opus_packet_pad(data, ret, max_data_bytes);
+ if (ret == OPUS_OK)
+ ret = max_data_bytes;
+ }
+ RESTORE_STACK;
+ return ret;
+ }
+ max_rate = frame_rate*max_data_bytes*8;
+
+ /* Equivalent 20-ms rate for mode/channel/bandwidth decisions */
+ equiv_rate = st->bitrate_bps - (40*st->channels+20)*(st->Fs/frame_size - 50);
+
+ if (st->signal_type == OPUS_SIGNAL_VOICE)
+ voice_est = 127;
+ else if (st->signal_type == OPUS_SIGNAL_MUSIC)
+ voice_est = 0;
+ else if (st->voice_ratio >= 0)
+ {
+ voice_est = st->voice_ratio*327>>8;
+ /* For AUDIO, never be more than 90% confident of having speech */
+ if (st->application == OPUS_APPLICATION_AUDIO)
+ voice_est = IMIN(voice_est, 115);
+ } else if (st->application == OPUS_APPLICATION_VOIP)
+ voice_est = 115;
+ else
+ voice_est = 48;
+
+ if (st->force_channels!=OPUS_AUTO && st->channels == 2)
+ {
+ st->stream_channels = st->force_channels;
+ } else {
+#ifdef FUZZING
+ /* Random mono/stereo decision */
+ if (st->channels == 2 && (rand()&0x1F)==0)
+ st->stream_channels = 3-st->stream_channels;
+#else
+ /* Rate-dependent mono-stereo decision */
+ if (st->channels == 2)
+ {
+ opus_int32 stereo_threshold;
+ stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14);
+ if (st->stream_channels == 2)
+ stereo_threshold -= 1000;
+ else
+ stereo_threshold += 1000;
+ st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1;
+ } else {
+ st->stream_channels = st->channels;
+ }
+#endif
+ }
+ equiv_rate = st->bitrate_bps - (40*st->stream_channels+20)*(st->Fs/frame_size - 50);
+
+ /* Mode selection depending on application and signal type */
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ {
+ st->mode = MODE_CELT_ONLY;
+ } else if (st->user_forced_mode == OPUS_AUTO)
+ {
+#ifdef FUZZING
+ /* Random mode switching */
+ if ((rand()&0xF)==0)
+ {
+ if ((rand()&0x1)==0)
+ st->mode = MODE_CELT_ONLY;
+ else
+ st->mode = MODE_SILK_ONLY;
+ } else {
+ if (st->prev_mode==MODE_CELT_ONLY)
+ st->mode = MODE_CELT_ONLY;
+ else
+ st->mode = MODE_SILK_ONLY;
+ }
+#else
+ opus_int32 mode_voice, mode_music;
+ opus_int32 threshold;
+
+ /* Interpolate based on stereo width */
+ mode_voice = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[0][0])
+ + MULT16_32_Q15(stereo_width,mode_thresholds[1][0]));
+ mode_music = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[1][1])
+ + MULT16_32_Q15(stereo_width,mode_thresholds[1][1]));
+ /* Interpolate based on speech/music probability */
+ threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14);
+ /* Bias towards SILK for VoIP because of some useful features */
+ if (st->application == OPUS_APPLICATION_VOIP)
+ threshold += 8000;
+
+ /*printf("%f %d\n", stereo_width/(float)Q15ONE, threshold);*/
+ /* Hysteresis */
+ if (st->prev_mode == MODE_CELT_ONLY)
+ threshold -= 4000;
+ else if (st->prev_mode>0)
+ threshold += 4000;
+
+ st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY;
+
+ /* When FEC is enabled and there's enough packet loss, use SILK */
+ if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4)
+ st->mode = MODE_SILK_ONLY;
+ /* When encoding voice and DTX is enabled, set the encoder to SILK mode (at least for now) */
+ if (st->silk_mode.useDTX && voice_est > 100)
+ st->mode = MODE_SILK_ONLY;
+#endif
+ } else {
+ st->mode = st->user_forced_mode;
+ }
+
+ /* Override the chosen mode to make sure we meet the requested frame size */
+ if (st->mode != MODE_CELT_ONLY && frame_size < st->Fs/100)
+ st->mode = MODE_CELT_ONLY;
+ if (st->lfe)
+ st->mode = MODE_CELT_ONLY;
+ /* If max_data_bytes represents less than 8 kb/s, switch to CELT-only mode */
+ if (max_data_bytes < (frame_rate > 50 ? 12000 : 8000)*frame_size / (st->Fs * 8))
+ st->mode = MODE_CELT_ONLY;
+
+ if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0
+ && st->mode != MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY)
+ {
+ /* Delay stereo->mono transition by two frames so that SILK can do a smooth downmix */
+ st->silk_mode.toMono = 1;
+ st->stream_channels = 2;
+ } else {
+ st->silk_mode.toMono = 0;
+ }
+
+ if (st->prev_mode > 0 &&
+ ((st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) ||
+ (st->mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY)))
+ {
+ redundancy = 1;
+ celt_to_silk = (st->mode != MODE_CELT_ONLY);
+ if (!celt_to_silk)
+ {
+ /* Switch to SILK/hybrid if frame size is 10 ms or more*/
+ if (frame_size >= st->Fs/100)
+ {
+ st->mode = st->prev_mode;
+ to_celt = 1;
+ } else {
+ redundancy=0;
+ }
+ }
+ }
+ /* For the first frame at a new SILK bandwidth */
+ if (st->silk_bw_switch)
+ {
+ redundancy = 1;
+ celt_to_silk = 1;
+ st->silk_bw_switch = 0;
+ prefill=1;
+ }
+
+ if (redundancy)
+ {
+ /* Fair share of the max size allowed */
+ redundancy_bytes = IMIN(257, max_data_bytes*(opus_int32)(st->Fs/200)/(frame_size+st->Fs/200));
+ /* For VBR, target the actual bitrate (subject to the limit above) */
+ if (st->use_vbr)
+ redundancy_bytes = IMIN(redundancy_bytes, st->bitrate_bps/1600);
+ }
+
+ if (st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY)
+ {
+ silk_EncControlStruct dummy;
+ silk_InitEncoder( silk_enc, st->arch, &dummy);
+ prefill=1;
+ }
+
+ /* Automatic (rate-dependent) bandwidth selection */
+ if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch)
+ {
+ const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds;
+ opus_int32 bandwidth_thresholds[8];
+ int bandwidth = OPUS_BANDWIDTH_FULLBAND;
+ opus_int32 equiv_rate2;
+
+ equiv_rate2 = equiv_rate;
+ if (st->mode != MODE_CELT_ONLY)
+ {
+ /* Adjust the threshold +/- 10% depending on complexity */
+ equiv_rate2 = equiv_rate2 * (45+st->silk_mode.complexity)/50;
+ /* CBR is less efficient by ~1 kb/s */
+ if (!st->use_vbr)
+ equiv_rate2 -= 1000;
+ }
+ if (st->channels==2 && st->force_channels!=1)
+ {
+ voice_bandwidth_thresholds = stereo_voice_bandwidth_thresholds;
+ music_bandwidth_thresholds = stereo_music_bandwidth_thresholds;
+ } else {
+ voice_bandwidth_thresholds = mono_voice_bandwidth_thresholds;
+ music_bandwidth_thresholds = mono_music_bandwidth_thresholds;
+ }
+ /* Interpolate bandwidth thresholds depending on voice estimation */
+ for (i=0;i<8;i++)
+ {
+ bandwidth_thresholds[i] = music_bandwidth_thresholds[i]
+ + ((voice_est*voice_est*(voice_bandwidth_thresholds[i]-music_bandwidth_thresholds[i]))>>14);
+ }
+ do {
+ int threshold, hysteresis;
+ threshold = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)];
+ hysteresis = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)+1];
+ if (!st->first)
+ {
+ if (st->bandwidth >= bandwidth)
+ threshold -= hysteresis;
+ else
+ threshold += hysteresis;
+ }
+ if (equiv_rate2 >= threshold)
+ break;
+ } while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND);
+ st->bandwidth = bandwidth;
+ /* Prevents any transition to SWB/FB until the SILK layer has fully
+ switched to WB mode and turned the variable LP filter off */
+ if (!st->first && st->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+ st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ }
+
+ if (st->bandwidth>st->max_bandwidth)
+ st->bandwidth = st->max_bandwidth;
+
+ if (st->user_bandwidth != OPUS_AUTO)
+ st->bandwidth = st->user_bandwidth;
+
+ /* This prevents us from using hybrid at unsafe CBR/max rates */
+ if (st->mode != MODE_CELT_ONLY && max_rate < 15000)
+ {
+ st->bandwidth = IMIN(st->bandwidth, OPUS_BANDWIDTH_WIDEBAND);
+ }
+
+ /* Prevents Opus from wasting bits on frequencies that are above
+ the Nyquist rate of the input signal */
+ if (st->Fs <= 24000 && st->bandwidth > OPUS_BANDWIDTH_SUPERWIDEBAND)
+ st->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+ if (st->Fs <= 16000 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+ st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ if (st->Fs <= 12000 && st->bandwidth > OPUS_BANDWIDTH_MEDIUMBAND)
+ st->bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+ if (st->Fs <= 8000 && st->bandwidth > OPUS_BANDWIDTH_NARROWBAND)
+ st->bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+#ifndef DISABLE_FLOAT_API
+ /* Use detected bandwidth to reduce the encoded bandwidth. */
+ if (st->detected_bandwidth && st->user_bandwidth == OPUS_AUTO)
+ {
+ int min_detected_bandwidth;
+ /* Makes bandwidth detection more conservative just in case the detector
+ gets it wrong when we could have coded a high bandwidth transparently.
+ When operating in SILK/hybrid mode, we don't go below wideband to avoid
+ more complicated switches that require redundancy. */
+ if (equiv_rate <= 18000*st->stream_channels && st->mode == MODE_CELT_ONLY)
+ min_detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+ else if (equiv_rate <= 24000*st->stream_channels && st->mode == MODE_CELT_ONLY)
+ min_detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+ else if (equiv_rate <= 30000*st->stream_channels)
+ min_detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ else if (equiv_rate <= 44000*st->stream_channels)
+ min_detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+ else
+ min_detected_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+ st->detected_bandwidth = IMAX(st->detected_bandwidth, min_detected_bandwidth);
+ st->bandwidth = IMIN(st->bandwidth, st->detected_bandwidth);
+ }
+#endif
+ celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(lsb_depth));
+
+ /* CELT mode doesn't support mediumband, use wideband instead */
+ if (st->mode == MODE_CELT_ONLY && st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+ st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ if (st->lfe)
+ st->bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+
+ /* Can't support higher than wideband for >20 ms frames */
+ if (frame_size > st->Fs/50 && (st->mode == MODE_CELT_ONLY || st->bandwidth > OPUS_BANDWIDTH_WIDEBAND))
+ {
+ VARDECL(unsigned char, tmp_data);
+ int nb_frames;
+ int bak_mode, bak_bandwidth, bak_channels, bak_to_mono;
+ VARDECL(OpusRepacketizer, rp);
+ opus_int32 bytes_per_frame;
+ opus_int32 repacketize_len;
+
+#ifndef DISABLE_FLOAT_API
+ if (analysis_read_pos_bak!= -1)
+ {
+ st->analysis.read_pos = analysis_read_pos_bak;
+ st->analysis.read_subframe = analysis_read_subframe_bak;
+ }
+#endif
+
+ nb_frames = frame_size > st->Fs/25 ? 3 : 2;
+ bytes_per_frame = IMIN(1276,(out_data_bytes-3)/nb_frames);
+
+ ALLOC(tmp_data, nb_frames*bytes_per_frame, unsigned char);
+
+ ALLOC(rp, 1, OpusRepacketizer);
+ opus_repacketizer_init(rp);
+
+ bak_mode = st->user_forced_mode;
+ bak_bandwidth = st->user_bandwidth;
+ bak_channels = st->force_channels;
+
+ st->user_forced_mode = st->mode;
+ st->user_bandwidth = st->bandwidth;
+ st->force_channels = st->stream_channels;
+ bak_to_mono = st->silk_mode.toMono;
+
+ if (bak_to_mono)
+ st->force_channels = 1;
+ else
+ st->prev_channels = st->stream_channels;
+ for (i=0;i<nb_frames;i++)
+ {
+ int tmp_len;
+ st->silk_mode.toMono = 0;
+ /* When switching from SILK/Hybrid to CELT, only ask for a switch at the last frame */
+ if (to_celt && i==nb_frames-1)
+ st->user_forced_mode = MODE_CELT_ONLY;
+ tmp_len = opus_encode_native(st, pcm+i*(st->channels*st->Fs/50), st->Fs/50,
+ tmp_data+i*bytes_per_frame, bytes_per_frame, lsb_depth,
+ NULL, 0, c1, c2, analysis_channels, downmix, float_api);
+ if (tmp_len<0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len);
+ if (ret<0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ }
+ if (st->use_vbr)
+ repacketize_len = out_data_bytes;
+ else
+ repacketize_len = IMIN(3*st->bitrate_bps/(3*8*50/nb_frames), out_data_bytes);
+ ret = opus_repacketizer_out_range_impl(rp, 0, nb_frames, data, repacketize_len, 0, !st->use_vbr);
+ if (ret<0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ st->user_forced_mode = bak_mode;
+ st->user_bandwidth = bak_bandwidth;
+ st->force_channels = bak_channels;
+ st->silk_mode.toMono = bak_to_mono;
+ RESTORE_STACK;
+ return ret;
+ }
+ curr_bandwidth = st->bandwidth;
+
+ /* Chooses the appropriate mode for speech
+ *NEVER* switch to/from CELT-only mode here as this will invalidate some assumptions */
+ if (st->mode == MODE_SILK_ONLY && curr_bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+ st->mode = MODE_HYBRID;
+ if (st->mode == MODE_HYBRID && curr_bandwidth <= OPUS_BANDWIDTH_WIDEBAND)
+ st->mode = MODE_SILK_ONLY;
+
+ /* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, curr_bandwidth); */
+ bytes_target = IMIN(max_data_bytes-redundancy_bytes, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1;
+
+ data += 1;
+
+ ec_enc_init(&enc, data, max_data_bytes-1);
+
+ ALLOC(pcm_buf, (total_buffer+frame_size)*st->channels, opus_val16);
+ OPUS_COPY(pcm_buf, &st->delay_buffer[(st->encoder_buffer-total_buffer)*st->channels], total_buffer*st->channels);
+
+ if (st->mode == MODE_CELT_ONLY)
+ hp_freq_smth1 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+ else
+ hp_freq_smth1 = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.variable_HP_smth1_Q15;
+
+ st->variable_HP_smth2_Q15 = silk_SMLAWB( st->variable_HP_smth2_Q15,
+ hp_freq_smth1 - st->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) );
+
+ /* convert from log scale to Hertz */
+ cutoff_Hz = silk_log2lin( silk_RSHIFT( st->variable_HP_smth2_Q15, 8 ) );
+
+ if (st->application == OPUS_APPLICATION_VOIP)
+ {
+ hp_cutoff(pcm, cutoff_Hz, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
+ } else {
+ dc_reject(pcm, 3, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
+ }
+#ifndef FIXED_POINT
+ if (float_api)
+ {
+ opus_val32 sum;
+ sum = celt_inner_prod(&pcm_buf[total_buffer*st->channels], &pcm_buf[total_buffer*st->channels], frame_size*st->channels, st->arch);
+ /* This should filter out both NaNs and ridiculous signals that could
+ cause NaNs further down. */
+ if (!(sum < 1e9f) || celt_isnan(sum))
+ {
+ OPUS_CLEAR(&pcm_buf[total_buffer*st->channels], frame_size*st->channels);
+ st->hp_mem[0] = st->hp_mem[1] = st->hp_mem[2] = st->hp_mem[3] = 0;
+ }
+ }
+#endif
+
+
+ /* SILK processing */
+ HB_gain = Q15ONE;
+ if (st->mode != MODE_CELT_ONLY)
+ {
+ opus_int32 total_bitRate, celt_rate;
+#ifdef FIXED_POINT
+ const opus_int16 *pcm_silk;
+#else
+ VARDECL(opus_int16, pcm_silk);
+ ALLOC(pcm_silk, st->channels*frame_size, opus_int16);
+#endif
+
+ /* Distribute bits between SILK and CELT */
+ total_bitRate = 8 * bytes_target * frame_rate;
+ if( st->mode == MODE_HYBRID ) {
+ int HB_gain_ref;
+ /* Base rate for SILK */
+ st->silk_mode.bitRate = st->stream_channels * ( 5000 + 1000 * ( st->Fs == 100 * frame_size ) );
+ if( curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND ) {
+ /* SILK gets 2/3 of the remaining bits */
+ st->silk_mode.bitRate += ( total_bitRate - st->silk_mode.bitRate ) * 2 / 3;
+ } else { /* FULLBAND */
+ /* SILK gets 3/5 of the remaining bits */
+ st->silk_mode.bitRate += ( total_bitRate - st->silk_mode.bitRate ) * 3 / 5;
+ }
+ /* Don't let SILK use more than 80% */
+ if( st->silk_mode.bitRate > total_bitRate * 4/5 ) {
+ st->silk_mode.bitRate = total_bitRate * 4/5;
+ }
+ if (!st->energy_masking)
+ {
+ /* Increasingly attenuate high band when it gets allocated fewer bits */
+ celt_rate = total_bitRate - st->silk_mode.bitRate;
+ HB_gain_ref = (curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND) ? 3000 : 3600;
+ HB_gain = SHL32((opus_val32)celt_rate, 9) / SHR32((opus_val32)celt_rate + st->stream_channels * HB_gain_ref, 6);
+ HB_gain = HB_gain < (opus_val32)Q15ONE*6/7 ? HB_gain + Q15ONE/7 : Q15ONE;
+ }
+ } else {
+ /* SILK gets all bits */
+ st->silk_mode.bitRate = total_bitRate;
+ }
+
+ /* Surround masking for SILK */
+ if (st->energy_masking && st->use_vbr && !st->lfe)
+ {
+ opus_val32 mask_sum=0;
+ opus_val16 masking_depth;
+ opus_int32 rate_offset;
+ int c;
+ int end = 17;
+ opus_int16 srate = 16000;
+ if (st->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
+ {
+ end = 13;
+ srate = 8000;
+ } else if (st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+ {
+ end = 15;
+ srate = 12000;
+ }
+ for (c=0;c<st->channels;c++)
+ {
+ for(i=0;i<end;i++)
+ {
+ opus_val16 mask;
+ mask = MAX16(MIN16(st->energy_masking[21*c+i],
+ QCONST16(.5f, DB_SHIFT)), -QCONST16(2.0f, DB_SHIFT));
+ if (mask > 0)
+ mask = HALF16(mask);
+ mask_sum += mask;
+ }
+ }
+ /* Conservative rate reduction, we cut the masking in half */
+ masking_depth = mask_sum / end*st->channels;
+ masking_depth += QCONST16(.2f, DB_SHIFT);
+ rate_offset = (opus_int32)PSHR32(MULT16_16(srate, masking_depth), DB_SHIFT);
+ rate_offset = MAX32(rate_offset, -2*st->silk_mode.bitRate/3);
+ /* Split the rate change between the SILK and CELT part for hybrid. */
+ if (st->bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND || st->bandwidth==OPUS_BANDWIDTH_FULLBAND)
+ st->silk_mode.bitRate += 3*rate_offset/5;
+ else
+ st->silk_mode.bitRate += rate_offset;
+ bytes_target += rate_offset * frame_size / (8 * st->Fs);
+ }
+
+ st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs;
+ st->silk_mode.nChannelsAPI = st->channels;
+ st->silk_mode.nChannelsInternal = st->stream_channels;
+ if (curr_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+ st->silk_mode.desiredInternalSampleRate = 8000;
+ } else if (curr_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+ st->silk_mode.desiredInternalSampleRate = 12000;
+ } else {
+ silk_assert( st->mode == MODE_HYBRID || curr_bandwidth == OPUS_BANDWIDTH_WIDEBAND );
+ st->silk_mode.desiredInternalSampleRate = 16000;
+ }
+ if( st->mode == MODE_HYBRID ) {
+ /* Don't allow bandwidth reduction at lowest bitrates in hybrid mode */
+ st->silk_mode.minInternalSampleRate = 16000;
+ } else {
+ st->silk_mode.minInternalSampleRate = 8000;
+ }
+
+ if (st->mode == MODE_SILK_ONLY)
+ {
+ opus_int32 effective_max_rate = max_rate;
+ st->silk_mode.maxInternalSampleRate = 16000;
+ if (frame_rate > 50)
+ effective_max_rate = effective_max_rate*2/3;
+ if (effective_max_rate < 13000)
+ {
+ st->silk_mode.maxInternalSampleRate = 12000;
+ st->silk_mode.desiredInternalSampleRate = IMIN(12000, st->silk_mode.desiredInternalSampleRate);
+ }
+ if (effective_max_rate < 9600)
+ {
+ st->silk_mode.maxInternalSampleRate = 8000;
+ st->silk_mode.desiredInternalSampleRate = IMIN(8000, st->silk_mode.desiredInternalSampleRate);
+ }
+ } else {
+ st->silk_mode.maxInternalSampleRate = 16000;
+ }
+
+ st->silk_mode.useCBR = !st->use_vbr;
+
+ /* Call SILK encoder for the low band */
+ nBytes = IMIN(1275, max_data_bytes-1-redundancy_bytes);
+
+ st->silk_mode.maxBits = nBytes*8;
+ /* Only allow up to 90% of the bits for hybrid mode*/
+ if (st->mode == MODE_HYBRID)
+ st->silk_mode.maxBits = (opus_int32)st->silk_mode.maxBits*9/10;
+ if (st->silk_mode.useCBR)
+ {
+ st->silk_mode.maxBits = (st->silk_mode.bitRate * frame_size / (st->Fs * 8))*8;
+ /* Reduce the initial target to make it easier to reach the CBR rate */
+ st->silk_mode.bitRate = IMAX(1, st->silk_mode.bitRate-2000);
+ }
+
+ if (prefill)
+ {
+ opus_int32 zero=0;
+ int prefill_offset;
+ /* Use a smooth onset for the SILK prefill to avoid the encoder trying to encode
+ a discontinuity. The exact location is what we need to avoid leaving any "gap"
+ in the audio when mixing with the redundant CELT frame. Here we can afford to
+ overwrite st->delay_buffer because the only thing that uses it before it gets
+ rewritten is tmp_prefill[] and even then only the part after the ramp really
+ gets used (rather than sent to the encoder and discarded) */
+ prefill_offset = st->channels*(st->encoder_buffer-st->delay_compensation-st->Fs/400);
+ gain_fade(st->delay_buffer+prefill_offset, st->delay_buffer+prefill_offset,
+ 0, Q15ONE, celt_mode->overlap, st->Fs/400, st->channels, celt_mode->window, st->Fs);
+ OPUS_CLEAR(st->delay_buffer, prefill_offset);
+#ifdef FIXED_POINT
+ pcm_silk = st->delay_buffer;
+#else
+ for (i=0;i<st->encoder_buffer*st->channels;i++)
+ pcm_silk[i] = FLOAT2INT16(st->delay_buffer[i]);
+#endif
+ silk_Encode( silk_enc, &st->silk_mode, pcm_silk, st->encoder_buffer, NULL, &zero, 1 );
+ }
+
+#ifdef FIXED_POINT
+ pcm_silk = pcm_buf+total_buffer*st->channels;
+#else
+ for (i=0;i<frame_size*st->channels;i++)
+ pcm_silk[i] = FLOAT2INT16(pcm_buf[total_buffer*st->channels + i]);
+#endif
+ ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 );
+ if( ret ) {
+ /*fprintf (stderr, "SILK encode error: %d\n", ret);*/
+ /* Handle error */
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ if (nBytes==0)
+ {
+ st->rangeFinal = 0;
+ data[-1] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels);
+ RESTORE_STACK;
+ return 1;
+ }
+ /* Extract SILK internal bandwidth for signaling in first byte */
+ if( st->mode == MODE_SILK_ONLY ) {
+ if( st->silk_mode.internalSampleRate == 8000 ) {
+ curr_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+ } else if( st->silk_mode.internalSampleRate == 12000 ) {
+ curr_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+ } else if( st->silk_mode.internalSampleRate == 16000 ) {
+ curr_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+ }
+ } else {
+ silk_assert( st->silk_mode.internalSampleRate == 16000 );
+ }
+
+ st->silk_mode.opusCanSwitch = st->silk_mode.switchReady;
+ /* FIXME: How do we allocate the redundancy for CBR? */
+ if (st->silk_mode.opusCanSwitch)
+ {
+ redundancy = 1;
+ celt_to_silk = 0;
+ st->silk_bw_switch = 1;
+ }
+ }
+
+ /* CELT processing */
+ {
+ int endband=21;
+
+ switch(curr_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_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband));
+ celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels));
+ }
+ celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX));
+ if (st->mode != MODE_SILK_ONLY)
+ {
+ opus_val32 celt_pred=2;
+ celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
+ /* We may still decide to disable prediction later */
+ if (st->silk_mode.reducedDependency)
+ celt_pred = 0;
+ celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(celt_pred));
+
+ if (st->mode == MODE_HYBRID)
+ {
+ int len;
+
+ len = (ec_tell(&enc)+7)>>3;
+ if (redundancy)
+ len += st->mode == MODE_HYBRID ? 3 : 1;
+ if( st->use_vbr ) {
+ nb_compr_bytes = len + bytes_target - (st->silk_mode.bitRate * frame_size) / (8 * st->Fs);
+ } else {
+ /* check if SILK used up too much */
+ nb_compr_bytes = len > bytes_target ? len : bytes_target;
+ }
+ } else {
+ if (st->use_vbr)
+ {
+ opus_int32 bonus=0;
+#ifndef DISABLE_FLOAT_API
+ if (st->variable_duration==OPUS_FRAMESIZE_VARIABLE && frame_size != st->Fs/50)
+ {
+ bonus = (60*st->stream_channels+40)*(st->Fs/frame_size-50);
+ if (analysis_info.valid)
+ bonus = (opus_int32)(bonus*(1.f+.5f*analysis_info.tonality));
+ }
+#endif
+ celt_encoder_ctl(celt_enc, OPUS_SET_VBR(1));
+ celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(st->vbr_constraint));
+ celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps+bonus));
+ nb_compr_bytes = max_data_bytes-1-redundancy_bytes;
+ } else {
+ nb_compr_bytes = bytes_target;
+ }
+ }
+
+ } else {
+ nb_compr_bytes = 0;
+ }
+
+ ALLOC(tmp_prefill, st->channels*st->Fs/400, opus_val16);
+ if (st->mode != MODE_SILK_ONLY && st->mode != st->prev_mode && st->prev_mode > 0)
+ {
+ OPUS_COPY(tmp_prefill, &st->delay_buffer[(st->encoder_buffer-total_buffer-st->Fs/400)*st->channels], st->channels*st->Fs/400);
+ }
+
+ if (st->channels*(st->encoder_buffer-(frame_size+total_buffer)) > 0)
+ {
+ OPUS_MOVE(st->delay_buffer, &st->delay_buffer[st->channels*frame_size], st->channels*(st->encoder_buffer-frame_size-total_buffer));
+ OPUS_COPY(&st->delay_buffer[st->channels*(st->encoder_buffer-frame_size-total_buffer)],
+ &pcm_buf[0],
+ (frame_size+total_buffer)*st->channels);
+ } else {
+ OPUS_COPY(st->delay_buffer, &pcm_buf[(frame_size+total_buffer-st->encoder_buffer)*st->channels], st->encoder_buffer*st->channels);
+ }
+ /* gain_fade() and stereo_fade() need to be after the buffer copying
+ because we don't want any of this to affect the SILK part */
+ if( st->prev_HB_gain < Q15ONE || HB_gain < Q15ONE ) {
+ gain_fade(pcm_buf, pcm_buf,
+ st->prev_HB_gain, HB_gain, celt_mode->overlap, frame_size, st->channels, celt_mode->window, st->Fs);
+ }
+ st->prev_HB_gain = HB_gain;
+ if (st->mode != MODE_HYBRID || st->stream_channels==1)
+ st->silk_mode.stereoWidth_Q14 = IMIN((1<<14),2*IMAX(0,equiv_rate-30000));
+ if( !st->energy_masking && st->channels == 2 ) {
+ /* Apply stereo width reduction (at low bitrates) */
+ if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) {
+ opus_val16 g1, g2;
+ g1 = st->hybrid_stereo_width_Q14;
+ g2 = (opus_val16)(st->silk_mode.stereoWidth_Q14);
+#ifdef FIXED_POINT
+ g1 = g1==16384 ? Q15ONE : SHL16(g1,1);
+ g2 = g2==16384 ? Q15ONE : SHL16(g2,1);
+#else
+ g1 *= (1.f/16384);
+ g2 *= (1.f/16384);
+#endif
+ stereo_fade(pcm_buf, pcm_buf, g1, g2, celt_mode->overlap,
+ frame_size, st->channels, celt_mode->window, st->Fs);
+ st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14;
+ }
+ }
+
+ if ( st->mode != MODE_CELT_ONLY && ec_tell(&enc)+17+20*(st->mode == MODE_HYBRID) <= 8*(max_data_bytes-1))
+ {
+ /* For SILK mode, the redundancy is inferred from the length */
+ if (st->mode == MODE_HYBRID && (redundancy || ec_tell(&enc)+37 <= 8*nb_compr_bytes))
+ ec_enc_bit_logp(&enc, redundancy, 12);
+ if (redundancy)
+ {
+ int max_redundancy;
+ ec_enc_bit_logp(&enc, celt_to_silk, 1);
+ if (st->mode == MODE_HYBRID)
+ max_redundancy = (max_data_bytes-1)-nb_compr_bytes;
+ else
+ max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+7)>>3);
+ /* Target the same bit-rate for redundancy as for the rest,
+ up to a max of 257 bytes */
+ redundancy_bytes = IMIN(max_redundancy, st->bitrate_bps/1600);
+ redundancy_bytes = IMIN(257, IMAX(2, redundancy_bytes));
+ if (st->mode == MODE_HYBRID)
+ ec_enc_uint(&enc, redundancy_bytes-2, 256);
+ }
+ } else {
+ redundancy = 0;
+ }
+
+ if (!redundancy)
+ {
+ st->silk_bw_switch = 0;
+ redundancy_bytes = 0;
+ }
+ if (st->mode != MODE_CELT_ONLY)start_band=17;
+
+ if (st->mode == MODE_SILK_ONLY)
+ {
+ ret = (ec_tell(&enc)+7)>>3;
+ ec_enc_done(&enc);
+ nb_compr_bytes = ret;
+ } else {
+ nb_compr_bytes = IMIN((max_data_bytes-1)-redundancy_bytes, nb_compr_bytes);
+ ec_enc_shrink(&enc, nb_compr_bytes);
+ }
+
+#ifndef DISABLE_FLOAT_API
+ if (redundancy || st->mode != MODE_SILK_ONLY)
+ celt_encoder_ctl(celt_enc, CELT_SET_ANALYSIS(&analysis_info));
+#endif
+
+ /* 5 ms redundant frame for CELT->SILK */
+ if (redundancy && celt_to_silk)
+ {
+ int err;
+ celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0));
+ celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
+ err = celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL);
+ if (err < 0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng));
+ celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+ }
+
+ celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(start_band));
+
+ if (st->mode != MODE_SILK_ONLY)
+ {
+ if (st->mode != st->prev_mode && st->prev_mode > 0)
+ {
+ unsigned char dummy[2];
+ celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+
+ /* Prefilling */
+ celt_encode_with_ec(celt_enc, tmp_prefill, st->Fs/400, dummy, 2, NULL);
+ celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0));
+ }
+ /* If false, we already busted the budget and we'll end up with a "PLC packet" */
+ if (ec_tell(&enc) <= 8*nb_compr_bytes)
+ {
+ ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc);
+ if (ret < 0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ }
+ }
+
+ /* 5 ms redundant frame for SILK->CELT */
+ if (redundancy && !celt_to_silk)
+ {
+ int err;
+ unsigned char dummy[2];
+ int N2, N4;
+ N2 = st->Fs/200;
+ N4 = st->Fs/400;
+
+ celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+ celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0));
+ celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0));
+
+ /* NOTE: We could speed this up slightly (at the expense of code size) by just adding a function that prefills the buffer */
+ celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2-N4), N4, dummy, 2, NULL);
+
+ err = celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL);
+ if (err < 0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng));
+ }
+
+
+
+ /* Signalling the mode in the first byte */
+ data--;
+ data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels);
+
+ st->rangeFinal = enc.rng ^ redundant_rng;
+
+ if (to_celt)
+ st->prev_mode = MODE_CELT_ONLY;
+ else
+ st->prev_mode = st->mode;
+ st->prev_channels = st->stream_channels;
+ st->prev_framesize = frame_size;
+
+ st->first = 0;
+
+ /* In the unlikely case that the SILK encoder busted its target, tell
+ the decoder to call the PLC */
+ if (ec_tell(&enc) > (max_data_bytes-1)*8)
+ {
+ if (max_data_bytes < 2)
+ {
+ RESTORE_STACK;
+ return OPUS_BUFFER_TOO_SMALL;
+ }
+ data[1] = 0;
+ ret = 1;
+ st->rangeFinal = 0;
+ } else if (st->mode==MODE_SILK_ONLY&&!redundancy)
+ {
+ /*When in LPC only mode it's perfectly
+ reasonable to strip off trailing zero bytes as
+ the required range decoder behavior is to
+ fill these in. This can't be done when the MDCT
+ modes are used because the decoder needs to know
+ the actual length for allocation purposes.*/
+ while(ret>2&&data[ret]==0)ret--;
+ }
+ /* Count ToC and redundancy */
+ ret += 1+redundancy_bytes;
+ if (!st->use_vbr)
+ {
+ if (opus_packet_pad(data, ret, max_data_bytes) != OPUS_OK)
+
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ ret = max_data_bytes;
+ }
+ RESTORE_STACK;
+ return ret;
+}
+
+#ifdef FIXED_POINT
+
+#ifndef DISABLE_FLOAT_API
+opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size,
+ unsigned char *data, opus_int32 max_data_bytes)
+{
+ int i, ret;
+ int frame_size;
+ int delay_compensation;
+ VARDECL(opus_int16, in);
+ ALLOC_STACK;
+
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ delay_compensation = 0;
+ else
+ delay_compensation = st->delay_compensation;
+ frame_size = compute_frame_size(pcm, analysis_frame_size,
+ st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+ delay_compensation, downmix_float, st->analysis.subframe_mem);
+
+ ALLOC(in, frame_size*st->channels, opus_int16);
+
+ for (i=0;i<frame_size*st->channels;i++)
+ in[i] = FLOAT2INT16(pcm[i]);
+ ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16,
+ pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1);
+ RESTORE_STACK;
+ return ret;
+}
+#endif
+
+opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size,
+ unsigned char *data, opus_int32 out_data_bytes)
+{
+ int frame_size;
+ int delay_compensation;
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ delay_compensation = 0;
+ else
+ delay_compensation = st->delay_compensation;
+ frame_size = compute_frame_size(pcm, analysis_frame_size,
+ st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+ delay_compensation, downmix_int
+#ifndef DISABLE_FLOAT_API
+ , st->analysis.subframe_mem
+#endif
+ );
+ return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 16,
+ pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0);
+}
+
+#else
+opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size,
+ unsigned char *data, opus_int32 max_data_bytes)
+{
+ int i, ret;
+ int frame_size;
+ int delay_compensation;
+ VARDECL(float, in);
+ ALLOC_STACK;
+
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ delay_compensation = 0;
+ else
+ delay_compensation = st->delay_compensation;
+ frame_size = compute_frame_size(pcm, analysis_frame_size,
+ st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+ delay_compensation, downmix_int, st->analysis.subframe_mem);
+
+ ALLOC(in, frame_size*st->channels, float);
+
+ for (i=0;i<frame_size*st->channels;i++)
+ in[i] = (1.0f/32768)*pcm[i];
+ ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16,
+ pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0);
+ RESTORE_STACK;
+ return ret;
+}
+opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size,
+ unsigned char *data, opus_int32 out_data_bytes)
+{
+ int frame_size;
+ int delay_compensation;
+ if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ delay_compensation = 0;
+ else
+ delay_compensation = st->delay_compensation;
+ frame_size = compute_frame_size(pcm, analysis_frame_size,
+ st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+ delay_compensation, downmix_float, st->analysis.subframe_mem);
+ return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 24,
+ pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1);
+}
+#endif
+
+
+int opus_encoder_ctl(OpusEncoder *st, int request, ...)
+{
+ int ret;
+ CELTEncoder *celt_enc;
+ va_list ap;
+
+ ret = OPUS_OK;
+ va_start(ap, request);
+
+ celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+
+ switch (request)
+ {
+ case OPUS_SET_APPLICATION_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if ( (value != OPUS_APPLICATION_VOIP && value != OPUS_APPLICATION_AUDIO
+ && value != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ || (!st->first && st->application != value))
+ {
+ ret = OPUS_BAD_ARG;
+ break;
+ }
+ st->application = value;
+ }
+ break;
+ case OPUS_GET_APPLICATION_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->application;
+ }
+ break;
+ case OPUS_SET_BITRATE_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value != OPUS_AUTO && value != OPUS_BITRATE_MAX)
+ {
+ if (value <= 0)
+ goto bad_arg;
+ else if (value <= 500)
+ value = 500;
+ else if (value > (opus_int32)300000*st->channels)
+ value = (opus_int32)300000*st->channels;
+ }
+ st->user_bitrate_bps = value;
+ }
+ break;
+ case OPUS_GET_BITRATE_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = user_bitrate_to_bitrate(st, st->prev_framesize, 1276);
+ }
+ break;
+ case OPUS_SET_FORCE_CHANNELS_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if((value<1 || value>st->channels) && value != OPUS_AUTO)
+ {
+ goto bad_arg;
+ }
+ st->force_channels = value;
+ }
+ break;
+ case OPUS_GET_FORCE_CHANNELS_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->force_channels;
+ }
+ break;
+ case OPUS_SET_MAX_BANDWIDTH_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND)
+ {
+ goto bad_arg;
+ }
+ st->max_bandwidth = value;
+ if (st->max_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+ st->silk_mode.maxInternalSampleRate = 8000;
+ } else if (st->max_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+ st->silk_mode.maxInternalSampleRate = 12000;
+ } else {
+ st->silk_mode.maxInternalSampleRate = 16000;
+ }
+ }
+ break;
+ case OPUS_GET_MAX_BANDWIDTH_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->max_bandwidth;
+ }
+ break;
+ case OPUS_SET_BANDWIDTH_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if ((value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) && value != OPUS_AUTO)
+ {
+ goto bad_arg;
+ }
+ st->user_bandwidth = value;
+ if (st->user_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+ st->silk_mode.maxInternalSampleRate = 8000;
+ } else if (st->user_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+ st->silk_mode.maxInternalSampleRate = 12000;
+ } else {
+ st->silk_mode.maxInternalSampleRate = 16000;
+ }
+ }
+ break;
+ case OPUS_GET_BANDWIDTH_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->bandwidth;
+ }
+ break;
+ case OPUS_SET_DTX_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value<0 || value>1)
+ {
+ goto bad_arg;
+ }
+ st->silk_mode.useDTX = value;
+ }
+ break;
+ case OPUS_GET_DTX_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->silk_mode.useDTX;
+ }
+ break;
+ case OPUS_SET_COMPLEXITY_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value<0 || value>10)
+ {
+ goto bad_arg;
+ }
+ st->silk_mode.complexity = value;
+ celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(value));
+ }
+ break;
+ case OPUS_GET_COMPLEXITY_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->silk_mode.complexity;
+ }
+ break;
+ case OPUS_SET_INBAND_FEC_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value<0 || value>1)
+ {
+ goto bad_arg;
+ }
+ st->silk_mode.useInBandFEC = value;
+ }
+ break;
+ case OPUS_GET_INBAND_FEC_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->silk_mode.useInBandFEC;
+ }
+ break;
+ case OPUS_SET_PACKET_LOSS_PERC_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value < 0 || value > 100)
+ {
+ goto bad_arg;
+ }
+ st->silk_mode.packetLossPercentage = value;
+ celt_encoder_ctl(celt_enc, OPUS_SET_PACKET_LOSS_PERC(value));
+ }
+ break;
+ case OPUS_GET_PACKET_LOSS_PERC_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->silk_mode.packetLossPercentage;
+ }
+ break;
+ case OPUS_SET_VBR_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value<0 || value>1)
+ {
+ goto bad_arg;
+ }
+ st->use_vbr = value;
+ st->silk_mode.useCBR = 1-value;
+ }
+ break;
+ case OPUS_GET_VBR_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->use_vbr;
+ }
+ break;
+ case OPUS_SET_VOICE_RATIO_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value<-1 || value>100)
+ {
+ goto bad_arg;
+ }
+ st->voice_ratio = value;
+ }
+ break;
+ case OPUS_GET_VOICE_RATIO_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->voice_ratio;
+ }
+ break;
+ case OPUS_SET_VBR_CONSTRAINT_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value<0 || value>1)
+ {
+ goto bad_arg;
+ }
+ st->vbr_constraint = value;
+ }
+ break;
+ case OPUS_GET_VBR_CONSTRAINT_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->vbr_constraint;
+ }
+ break;
+ case OPUS_SET_SIGNAL_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if(value!=OPUS_AUTO && value!=OPUS_SIGNAL_VOICE && value!=OPUS_SIGNAL_MUSIC)
+ {
+ goto bad_arg;
+ }
+ st->signal_type = value;
+ }
+ break;
+ case OPUS_GET_SIGNAL_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->signal_type;
+ }
+ break;
+ case OPUS_GET_LOOKAHEAD_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->Fs/400;
+ if (st->application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ *value += st->delay_compensation;
+ }
+ break;
+ case OPUS_GET_SAMPLE_RATE_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->Fs;
+ }
+ break;
+ case OPUS_GET_FINAL_RANGE_REQUEST:
+ {
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->rangeFinal;
+ }
+ 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*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->lsb_depth;
+ }
+ break;
+ case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value != OPUS_FRAMESIZE_ARG && value != OPUS_FRAMESIZE_2_5_MS &&
+ value != OPUS_FRAMESIZE_5_MS && value != OPUS_FRAMESIZE_10_MS &&
+ value != OPUS_FRAMESIZE_20_MS && value != OPUS_FRAMESIZE_40_MS &&
+ value != OPUS_FRAMESIZE_60_MS && value != OPUS_FRAMESIZE_VARIABLE)
+ {
+ goto bad_arg;
+ }
+ st->variable_duration = value;
+ celt_encoder_ctl(celt_enc, OPUS_SET_EXPERT_FRAME_DURATION(value));
+ }
+ break;
+ case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->variable_duration;
+ }
+ break;
+ case OPUS_SET_PREDICTION_DISABLED_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value > 1 || value < 0)
+ goto bad_arg;
+ st->silk_mode.reducedDependency = value;
+ }
+ break;
+ case OPUS_GET_PREDICTION_DISABLED_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ goto bad_arg;
+ *value = st->silk_mode.reducedDependency;
+ }
+ break;
+ case OPUS_RESET_STATE:
+ {
+ void *silk_enc;
+ silk_EncControlStruct dummy;
+ char *start;
+ silk_enc = (char*)st+st->silk_enc_offset;
+#ifndef DISABLE_FLOAT_API
+ tonality_analysis_reset(&st->analysis);
+#endif
+
+ start = (char*)&st->OPUS_ENCODER_RESET_START;
+ OPUS_CLEAR(start, sizeof(OpusEncoder) - (start - (char*)st));
+
+ celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+ silk_InitEncoder( silk_enc, st->arch, &dummy );
+ st->stream_channels = st->channels;
+ st->hybrid_stereo_width_Q14 = 1 << 14;
+ st->prev_HB_gain = Q15ONE;
+ st->first = 1;
+ st->mode = MODE_HYBRID;
+ st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+ st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+ }
+ break;
+ case OPUS_SET_FORCE_MODE_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if ((value < MODE_SILK_ONLY || value > MODE_CELT_ONLY) && value != OPUS_AUTO)
+ {
+ goto bad_arg;
+ }
+ st->user_forced_mode = value;
+ }
+ break;
+ case OPUS_SET_LFE_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ st->lfe = value;
+ ret = celt_encoder_ctl(celt_enc, OPUS_SET_LFE(value));
+ }
+ break;
+ case OPUS_SET_ENERGY_MASK_REQUEST:
+ {
+ opus_val16 *value = va_arg(ap, opus_val16*);
+ st->energy_masking = value;
+ ret = celt_encoder_ctl(celt_enc, OPUS_SET_ENERGY_MASK(value));
+ }
+ break;
+
+ case CELT_GET_MODE_REQUEST:
+ {
+ const CELTMode ** value = va_arg(ap, const CELTMode**);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ ret = celt_encoder_ctl(celt_enc, CELT_GET_MODE(value));
+ }
+ break;
+ default:
+ /* fprintf(stderr, "unknown opus_encoder_ctl() request: %d", request);*/
+ ret = OPUS_UNIMPLEMENTED;
+ break;
+ }
+ va_end(ap);
+ return ret;
+bad_arg:
+ va_end(ap);
+ return OPUS_BAD_ARG;
+}
+
+void opus_encoder_destroy(OpusEncoder *st)
+{
+ opus_free(st);
+}
diff --git a/media/libopus/src/opus_multistream.c b/media/libopus/src/opus_multistream.c
new file mode 100644
index 000000000..09c3639b7
--- /dev/null
+++ b/media/libopus/src/opus_multistream.c
@@ -0,0 +1,92 @@
+/* Copyright (c) 2011 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 "config.h"
+#endif
+
+#include "opus_multistream.h"
+#include "opus.h"
+#include "opus_private.h"
+#include "stack_alloc.h"
+#include <stdarg.h>
+#include "float_cast.h"
+#include "os_support.h"
+
+
+int validate_layout(const ChannelLayout *layout)
+{
+ int i, max_channel;
+
+ max_channel = layout->nb_streams+layout->nb_coupled_streams;
+ if (max_channel>255)
+ return 0;
+ for (i=0;i<layout->nb_channels;i++)
+ {
+ if (layout->mapping[i] >= max_channel && layout->mapping[i] != 255)
+ return 0;
+ }
+ return 1;
+}
+
+
+int get_left_channel(const ChannelLayout *layout, int stream_id, int prev)
+{
+ int i;
+ i = (prev<0) ? 0 : prev+1;
+ for (;i<layout->nb_channels;i++)
+ {
+ if (layout->mapping[i]==stream_id*2)
+ return i;
+ }
+ return -1;
+}
+
+int get_right_channel(const ChannelLayout *layout, int stream_id, int prev)
+{
+ int i;
+ i = (prev<0) ? 0 : prev+1;
+ for (;i<layout->nb_channels;i++)
+ {
+ if (layout->mapping[i]==stream_id*2+1)
+ return i;
+ }
+ return -1;
+}
+
+int get_mono_channel(const ChannelLayout *layout, int stream_id, int prev)
+{
+ int i;
+ i = (prev<0) ? 0 : prev+1;
+ for (;i<layout->nb_channels;i++)
+ {
+ if (layout->mapping[i]==stream_id+layout->nb_coupled_streams)
+ return i;
+ }
+ return -1;
+}
+
diff --git a/media/libopus/src/opus_multistream_decoder.c b/media/libopus/src/opus_multistream_decoder.c
new file mode 100644
index 000000000..b95eaa6ea
--- /dev/null
+++ b/media/libopus/src/opus_multistream_decoder.c
@@ -0,0 +1,537 @@
+/* Copyright (c) 2011 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 "config.h"
+#endif
+
+#include "opus_multistream.h"
+#include "opus.h"
+#include "opus_private.h"
+#include "stack_alloc.h"
+#include <stdarg.h>
+#include "float_cast.h"
+#include "os_support.h"
+
+struct OpusMSDecoder {
+ ChannelLayout layout;
+ /* Decoder states go here */
+};
+
+
+
+
+/* DECODER */
+
+opus_int32 opus_multistream_decoder_get_size(int nb_streams, int nb_coupled_streams)
+{
+ int coupled_size;
+ int mono_size;
+
+ if(nb_streams<1||nb_coupled_streams>nb_streams||nb_coupled_streams<0)return 0;
+ coupled_size = opus_decoder_get_size(2);
+ mono_size = opus_decoder_get_size(1);
+ return align(sizeof(OpusMSDecoder))
+ + nb_coupled_streams * align(coupled_size)
+ + (nb_streams-nb_coupled_streams) * align(mono_size);
+}
+
+int opus_multistream_decoder_init(
+ OpusMSDecoder *st,
+ opus_int32 Fs,
+ int channels,
+ int streams,
+ int coupled_streams,
+ const unsigned char *mapping
+)
+{
+ int coupled_size;
+ int mono_size;
+ int i, ret;
+ char *ptr;
+
+ if ((channels>255) || (channels<1) || (coupled_streams>streams) ||
+ (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams))
+ return OPUS_BAD_ARG;
+
+ st->layout.nb_channels = channels;
+ st->layout.nb_streams = streams;
+ st->layout.nb_coupled_streams = coupled_streams;
+
+ for (i=0;i<st->layout.nb_channels;i++)
+ st->layout.mapping[i] = mapping[i];
+ if (!validate_layout(&st->layout))
+ return OPUS_BAD_ARG;
+
+ ptr = (char*)st + align(sizeof(OpusMSDecoder));
+ coupled_size = opus_decoder_get_size(2);
+ mono_size = opus_decoder_get_size(1);
+
+ for (i=0;i<st->layout.nb_coupled_streams;i++)
+ {
+ ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 2);
+ if(ret!=OPUS_OK)return ret;
+ ptr += align(coupled_size);
+ }
+ for (;i<st->layout.nb_streams;i++)
+ {
+ ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 1);
+ if(ret!=OPUS_OK)return ret;
+ ptr += align(mono_size);
+ }
+ return OPUS_OK;
+}
+
+
+OpusMSDecoder *opus_multistream_decoder_create(
+ opus_int32 Fs,
+ int channels,
+ int streams,
+ int coupled_streams,
+ const unsigned char *mapping,
+ int *error
+)
+{
+ int ret;
+ OpusMSDecoder *st;
+ if ((channels>255) || (channels<1) || (coupled_streams>streams) ||
+ (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams))
+ {
+ if (error)
+ *error = OPUS_BAD_ARG;
+ return NULL;
+ }
+ st = (OpusMSDecoder *)opus_alloc(opus_multistream_decoder_get_size(streams, coupled_streams));
+ if (st==NULL)
+ {
+ if (error)
+ *error = OPUS_ALLOC_FAIL;
+ return NULL;
+ }
+ ret = opus_multistream_decoder_init(st, Fs, channels, streams, coupled_streams, mapping);
+ if (error)
+ *error = ret;
+ if (ret != OPUS_OK)
+ {
+ opus_free(st);
+ st = NULL;
+ }
+ return st;
+}
+
+typedef void (*opus_copy_channel_out_func)(
+ void *dst,
+ int dst_stride,
+ int dst_channel,
+ const opus_val16 *src,
+ int src_stride,
+ int frame_size
+);
+
+static int opus_multistream_packet_validate(const unsigned char *data,
+ opus_int32 len, int nb_streams, opus_int32 Fs)
+{
+ int s;
+ int count;
+ unsigned char toc;
+ opus_int16 size[48];
+ int samples=0;
+ opus_int32 packet_offset;
+
+ for (s=0;s<nb_streams;s++)
+ {
+ int tmp_samples;
+ if (len<=0)
+ return OPUS_INVALID_PACKET;
+ count = opus_packet_parse_impl(data, len, s!=nb_streams-1, &toc, NULL,
+ size, NULL, &packet_offset);
+ if (count<0)
+ return count;
+ tmp_samples = opus_packet_get_nb_samples(data, packet_offset, Fs);
+ if (s!=0 && samples != tmp_samples)
+ return OPUS_INVALID_PACKET;
+ samples = tmp_samples;
+ data += packet_offset;
+ len -= packet_offset;
+ }
+ return samples;
+}
+
+static int opus_multistream_decode_native(
+ OpusMSDecoder *st,
+ const unsigned char *data,
+ opus_int32 len,
+ void *pcm,
+ opus_copy_channel_out_func copy_channel_out,
+ int frame_size,
+ int decode_fec,
+ int soft_clip
+)
+{
+ opus_int32 Fs;
+ int coupled_size;
+ int mono_size;
+ int s, c;
+ char *ptr;
+ int do_plc=0;
+ VARDECL(opus_val16, buf);
+ ALLOC_STACK;
+
+ /* Limit frame_size to avoid excessive stack allocations. */
+ opus_multistream_decoder_ctl(st, OPUS_GET_SAMPLE_RATE(&Fs));
+ frame_size = IMIN(frame_size, Fs/25*3);
+ ALLOC(buf, 2*frame_size, opus_val16);
+ ptr = (char*)st + align(sizeof(OpusMSDecoder));
+ coupled_size = opus_decoder_get_size(2);
+ mono_size = opus_decoder_get_size(1);
+
+ if (len==0)
+ do_plc = 1;
+ if (len < 0)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+ if (!do_plc && len < 2*st->layout.nb_streams-1)
+ {
+ RESTORE_STACK;
+ return OPUS_INVALID_PACKET;
+ }
+ if (!do_plc)
+ {
+ int ret = opus_multistream_packet_validate(data, len, st->layout.nb_streams, Fs);
+ if (ret < 0)
+ {
+ RESTORE_STACK;
+ return ret;
+ } else if (ret > frame_size)
+ {
+ RESTORE_STACK;
+ return OPUS_BUFFER_TOO_SMALL;
+ }
+ }
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusDecoder *dec;
+ int packet_offset, ret;
+
+ dec = (OpusDecoder*)ptr;
+ ptr += (s < st->layout.nb_coupled_streams) ? align(coupled_size) : align(mono_size);
+
+ if (!do_plc && len<=0)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ packet_offset = 0;
+ ret = opus_decode_native(dec, data, len, buf, frame_size, decode_fec, s!=st->layout.nb_streams-1, &packet_offset, soft_clip);
+ data += packet_offset;
+ len -= packet_offset;
+ if (ret <= 0)
+ {
+ RESTORE_STACK;
+ return ret;
+ }
+ frame_size = ret;
+ if (s < st->layout.nb_coupled_streams)
+ {
+ int chan, prev;
+ prev = -1;
+ /* Copy "left" audio to the channel(s) where it belongs */
+ while ( (chan = get_left_channel(&st->layout, s, prev)) != -1)
+ {
+ (*copy_channel_out)(pcm, st->layout.nb_channels, chan,
+ buf, 2, frame_size);
+ prev = chan;
+ }
+ prev = -1;
+ /* Copy "right" audio to the channel(s) where it belongs */
+ while ( (chan = get_right_channel(&st->layout, s, prev)) != -1)
+ {
+ (*copy_channel_out)(pcm, st->layout.nb_channels, chan,
+ buf+1, 2, frame_size);
+ prev = chan;
+ }
+ } else {
+ int chan, prev;
+ prev = -1;
+ /* Copy audio to the channel(s) where it belongs */
+ while ( (chan = get_mono_channel(&st->layout, s, prev)) != -1)
+ {
+ (*copy_channel_out)(pcm, st->layout.nb_channels, chan,
+ buf, 1, frame_size);
+ prev = chan;
+ }
+ }
+ }
+ /* Handle muted channels */
+ for (c=0;c<st->layout.nb_channels;c++)
+ {
+ if (st->layout.mapping[c] == 255)
+ {
+ (*copy_channel_out)(pcm, st->layout.nb_channels, c,
+ NULL, 0, frame_size);
+ }
+ }
+ RESTORE_STACK;
+ return frame_size;
+}
+
+#if !defined(DISABLE_FLOAT_API)
+static void opus_copy_channel_out_float(
+ void *dst,
+ int dst_stride,
+ int dst_channel,
+ const opus_val16 *src,
+ int src_stride,
+ int frame_size
+)
+{
+ float *float_dst;
+ opus_int32 i;
+ float_dst = (float*)dst;
+ if (src != NULL)
+ {
+ for (i=0;i<frame_size;i++)
+#if defined(FIXED_POINT)
+ float_dst[i*dst_stride+dst_channel] = (1/32768.f)*src[i*src_stride];
+#else
+ float_dst[i*dst_stride+dst_channel] = src[i*src_stride];
+#endif
+ }
+ else
+ {
+ for (i=0;i<frame_size;i++)
+ float_dst[i*dst_stride+dst_channel] = 0;
+ }
+}
+#endif
+
+static void opus_copy_channel_out_short(
+ void *dst,
+ int dst_stride,
+ int dst_channel,
+ const opus_val16 *src,
+ int src_stride,
+ int frame_size
+)
+{
+ opus_int16 *short_dst;
+ opus_int32 i;
+ short_dst = (opus_int16*)dst;
+ if (src != NULL)
+ {
+ for (i=0;i<frame_size;i++)
+#if defined(FIXED_POINT)
+ short_dst[i*dst_stride+dst_channel] = src[i*src_stride];
+#else
+ short_dst[i*dst_stride+dst_channel] = FLOAT2INT16(src[i*src_stride]);
+#endif
+ }
+ else
+ {
+ for (i=0;i<frame_size;i++)
+ short_dst[i*dst_stride+dst_channel] = 0;
+ }
+}
+
+
+
+#ifdef FIXED_POINT
+int opus_multistream_decode(
+ OpusMSDecoder *st,
+ const unsigned char *data,
+ opus_int32 len,
+ opus_int16 *pcm,
+ int frame_size,
+ int decode_fec
+)
+{
+ return opus_multistream_decode_native(st, data, len,
+ pcm, opus_copy_channel_out_short, frame_size, decode_fec, 0);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_multistream_decode_float(OpusMSDecoder *st, const unsigned char *data,
+ opus_int32 len, float *pcm, int frame_size, int decode_fec)
+{
+ return opus_multistream_decode_native(st, data, len,
+ pcm, opus_copy_channel_out_float, frame_size, decode_fec, 0);
+}
+#endif
+
+#else
+
+int opus_multistream_decode(OpusMSDecoder *st, const unsigned char *data,
+ opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec)
+{
+ return opus_multistream_decode_native(st, data, len,
+ pcm, opus_copy_channel_out_short, frame_size, decode_fec, 1);
+}
+
+int opus_multistream_decode_float(
+ OpusMSDecoder *st,
+ const unsigned char *data,
+ opus_int32 len,
+ float *pcm,
+ int frame_size,
+ int decode_fec
+)
+{
+ return opus_multistream_decode_native(st, data, len,
+ pcm, opus_copy_channel_out_float, frame_size, decode_fec, 0);
+}
+#endif
+
+int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...)
+{
+ va_list ap;
+ int coupled_size, mono_size;
+ char *ptr;
+ int ret = OPUS_OK;
+
+ va_start(ap, request);
+
+ coupled_size = opus_decoder_get_size(2);
+ mono_size = opus_decoder_get_size(1);
+ ptr = (char*)st + align(sizeof(OpusMSDecoder));
+ switch (request)
+ {
+ case OPUS_GET_BANDWIDTH_REQUEST:
+ case OPUS_GET_SAMPLE_RATE_REQUEST:
+ case OPUS_GET_GAIN_REQUEST:
+ case OPUS_GET_LAST_PACKET_DURATION_REQUEST:
+ {
+ OpusDecoder *dec;
+ /* For int32* GET params, just query the first stream */
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ dec = (OpusDecoder*)ptr;
+ ret = opus_decoder_ctl(dec, request, value);
+ }
+ break;
+ case OPUS_GET_FINAL_RANGE_REQUEST:
+ {
+ int s;
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ opus_uint32 tmp;
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = 0;
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusDecoder *dec;
+ dec = (OpusDecoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_decoder_ctl(dec, request, &tmp);
+ if (ret != OPUS_OK) break;
+ *value ^= tmp;
+ }
+ }
+ break;
+ case OPUS_RESET_STATE:
+ {
+ int s;
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusDecoder *dec;
+
+ dec = (OpusDecoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_decoder_ctl(dec, OPUS_RESET_STATE);
+ if (ret != OPUS_OK)
+ break;
+ }
+ }
+ break;
+ case OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST:
+ {
+ int s;
+ opus_int32 stream_id;
+ OpusDecoder **value;
+ stream_id = va_arg(ap, opus_int32);
+ if (stream_id<0 || stream_id >= st->layout.nb_streams)
+ ret = OPUS_BAD_ARG;
+ value = va_arg(ap, OpusDecoder**);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ for (s=0;s<stream_id;s++)
+ {
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ }
+ *value = (OpusDecoder*)ptr;
+ }
+ break;
+ case OPUS_SET_GAIN_REQUEST:
+ {
+ int s;
+ /* This works for int32 params */
+ opus_int32 value = va_arg(ap, opus_int32);
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusDecoder *dec;
+
+ dec = (OpusDecoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_decoder_ctl(dec, request, value);
+ if (ret != OPUS_OK)
+ break;
+ }
+ }
+ break;
+ default:
+ ret = OPUS_UNIMPLEMENTED;
+ break;
+ }
+
+ va_end(ap);
+ return ret;
+bad_arg:
+ va_end(ap);
+ return OPUS_BAD_ARG;
+}
+
+
+void opus_multistream_decoder_destroy(OpusMSDecoder *st)
+{
+ opus_free(st);
+}
diff --git a/media/libopus/src/opus_multistream_encoder.c b/media/libopus/src/opus_multistream_encoder.c
new file mode 100644
index 000000000..e722e31ab
--- /dev/null
+++ b/media/libopus/src/opus_multistream_encoder.c
@@ -0,0 +1,1351 @@
+/* Copyright (c) 2011 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 "config.h"
+#endif
+
+#include "opus_multistream.h"
+#include "opus.h"
+#include "opus_private.h"
+#include "stack_alloc.h"
+#include <stdarg.h>
+#include "float_cast.h"
+#include "os_support.h"
+#include "mathops.h"
+#include "mdct.h"
+#include "modes.h"
+#include "bands.h"
+#include "quant_bands.h"
+#include "pitch.h"
+
+typedef struct {
+ int nb_streams;
+ int nb_coupled_streams;
+ unsigned char mapping[8];
+} VorbisLayout;
+
+/* Index is nb_channel-1*/
+static const VorbisLayout vorbis_mappings[8] = {
+ {1, 0, {0}}, /* 1: mono */
+ {1, 1, {0, 1}}, /* 2: stereo */
+ {2, 1, {0, 2, 1}}, /* 3: 1-d surround */
+ {2, 2, {0, 1, 2, 3}}, /* 4: quadraphonic surround */
+ {3, 2, {0, 4, 1, 2, 3}}, /* 5: 5-channel surround */
+ {4, 2, {0, 4, 1, 2, 3, 5}}, /* 6: 5.1 surround */
+ {4, 3, {0, 4, 1, 2, 3, 5, 6}}, /* 7: 6.1 surround */
+ {5, 3, {0, 6, 1, 2, 3, 4, 5, 7}}, /* 8: 7.1 surround */
+};
+
+typedef void (*opus_copy_channel_in_func)(
+ opus_val16 *dst,
+ int dst_stride,
+ const void *src,
+ int src_stride,
+ int src_channel,
+ int frame_size
+);
+
+typedef enum {
+ MAPPING_TYPE_NONE,
+ MAPPING_TYPE_SURROUND
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ , /* Do not include comma at end of enumerator list */
+ MAPPING_TYPE_AMBISONICS
+#endif
+} MappingType;
+
+struct OpusMSEncoder {
+ ChannelLayout layout;
+ int arch;
+ int lfe_stream;
+ int application;
+ int variable_duration;
+ MappingType mapping_type;
+ opus_int32 bitrate_bps;
+ float subframe_mem[3];
+ /* Encoder states go here */
+ /* then opus_val32 window_mem[channels*120]; */
+ /* then opus_val32 preemph_mem[channels]; */
+};
+
+static opus_val32 *ms_get_preemph_mem(OpusMSEncoder *st)
+{
+ int s;
+ char *ptr;
+ int coupled_size, mono_size;
+
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ }
+ /* void* cast avoids clang -Wcast-align warning */
+ return (opus_val32*)(void*)(ptr+st->layout.nb_channels*120*sizeof(opus_val32));
+}
+
+static opus_val32 *ms_get_window_mem(OpusMSEncoder *st)
+{
+ int s;
+ char *ptr;
+ int coupled_size, mono_size;
+
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ }
+ /* void* cast avoids clang -Wcast-align warning */
+ return (opus_val32*)(void*)ptr;
+}
+
+static int validate_encoder_layout(const ChannelLayout *layout)
+{
+ int s;
+ for (s=0;s<layout->nb_streams;s++)
+ {
+ if (s < layout->nb_coupled_streams)
+ {
+ if (get_left_channel(layout, s, -1)==-1)
+ return 0;
+ if (get_right_channel(layout, s, -1)==-1)
+ return 0;
+ } else {
+ if (get_mono_channel(layout, s, -1)==-1)
+ return 0;
+ }
+ }
+ return 1;
+}
+
+static void channel_pos(int channels, int pos[8])
+{
+ /* Position in the mix: 0 don't mix, 1: left, 2: center, 3:right */
+ if (channels==4)
+ {
+ pos[0]=1;
+ pos[1]=3;
+ pos[2]=1;
+ pos[3]=3;
+ } else if (channels==3||channels==5||channels==6)
+ {
+ pos[0]=1;
+ pos[1]=2;
+ pos[2]=3;
+ pos[3]=1;
+ pos[4]=3;
+ pos[5]=0;
+ } else if (channels==7)
+ {
+ pos[0]=1;
+ pos[1]=2;
+ pos[2]=3;
+ pos[3]=1;
+ pos[4]=3;
+ pos[5]=2;
+ pos[6]=0;
+ } else if (channels==8)
+ {
+ pos[0]=1;
+ pos[1]=2;
+ pos[2]=3;
+ pos[3]=1;
+ pos[4]=3;
+ pos[5]=1;
+ pos[6]=3;
+ pos[7]=0;
+ }
+}
+
+#if 1
+/* Computes a rough approximation of log2(2^a + 2^b) */
+static opus_val16 logSum(opus_val16 a, opus_val16 b)
+{
+ opus_val16 max;
+ opus_val32 diff;
+ opus_val16 frac;
+ static const opus_val16 diff_table[17] = {
+ QCONST16(0.5000000f, DB_SHIFT), QCONST16(0.2924813f, DB_SHIFT), QCONST16(0.1609640f, DB_SHIFT), QCONST16(0.0849625f, DB_SHIFT),
+ QCONST16(0.0437314f, DB_SHIFT), QCONST16(0.0221971f, DB_SHIFT), QCONST16(0.0111839f, DB_SHIFT), QCONST16(0.0056136f, DB_SHIFT),
+ QCONST16(0.0028123f, DB_SHIFT)
+ };
+ int low;
+ if (a>b)
+ {
+ max = a;
+ diff = SUB32(EXTEND32(a),EXTEND32(b));
+ } else {
+ max = b;
+ diff = SUB32(EXTEND32(b),EXTEND32(a));
+ }
+ if (!(diff < QCONST16(8.f, DB_SHIFT))) /* inverted to catch NaNs */
+ return max;
+#ifdef FIXED_POINT
+ low = SHR32(diff, DB_SHIFT-1);
+ frac = SHL16(diff - SHL16(low, DB_SHIFT-1), 16-DB_SHIFT);
+#else
+ low = (int)floor(2*diff);
+ frac = 2*diff - low;
+#endif
+ return max + diff_table[low] + MULT16_16_Q15(frac, SUB16(diff_table[low+1], diff_table[low]));
+}
+#else
+opus_val16 logSum(opus_val16 a, opus_val16 b)
+{
+ return log2(pow(4, a)+ pow(4, b))/2;
+}
+#endif
+
+void surround_analysis(const CELTMode *celt_mode, const void *pcm, opus_val16 *bandLogE, opus_val32 *mem, opus_val32 *preemph_mem,
+ int len, int overlap, int channels, int rate, opus_copy_channel_in_func copy_channel_in, int arch
+)
+{
+ int c;
+ int i;
+ int LM;
+ int pos[8] = {0};
+ int upsample;
+ int frame_size;
+ opus_val16 channel_offset;
+ opus_val32 bandE[21];
+ opus_val16 maskLogE[3][21];
+ VARDECL(opus_val32, in);
+ VARDECL(opus_val16, x);
+ VARDECL(opus_val32, freq);
+ SAVE_STACK;
+
+ upsample = resampling_factor(rate);
+ frame_size = len*upsample;
+
+ /* LM = log2(frame_size / 120) */
+ for (LM=0;LM<celt_mode->maxLM;LM++)
+ if (celt_mode->shortMdctSize<<LM==frame_size)
+ break;
+
+ ALLOC(in, frame_size+overlap, opus_val32);
+ ALLOC(x, len, opus_val16);
+ ALLOC(freq, frame_size, opus_val32);
+
+ channel_pos(channels, pos);
+
+ for (c=0;c<3;c++)
+ for (i=0;i<21;i++)
+ maskLogE[c][i] = -QCONST16(28.f, DB_SHIFT);
+
+ for (c=0;c<channels;c++)
+ {
+ OPUS_COPY(in, mem+c*overlap, overlap);
+ (*copy_channel_in)(x, 1, pcm, channels, c, len);
+ celt_preemphasis(x, in+overlap, frame_size, 1, upsample, celt_mode->preemph, preemph_mem+c, 0);
+#ifndef FIXED_POINT
+ {
+ opus_val32 sum;
+ sum = celt_inner_prod(in, in, frame_size+overlap, 0);
+ /* This should filter out both NaNs and ridiculous signals that could
+ cause NaNs further down. */
+ if (!(sum < 1e9f) || celt_isnan(sum))
+ {
+ OPUS_CLEAR(in, frame_size+overlap);
+ preemph_mem[c] = 0;
+ }
+ }
+#endif
+ clt_mdct_forward(&celt_mode->mdct, in, freq, celt_mode->window,
+ overlap, celt_mode->maxLM-LM, 1, arch);
+ if (upsample != 1)
+ {
+ int bound = len;
+ for (i=0;i<bound;i++)
+ freq[i] *= upsample;
+ for (;i<frame_size;i++)
+ freq[i] = 0;
+ }
+
+ compute_band_energies(celt_mode, freq, bandE, 21, 1, LM);
+ amp2Log2(celt_mode, 21, 21, bandE, bandLogE+21*c, 1);
+ /* Apply spreading function with -6 dB/band going up and -12 dB/band going down. */
+ for (i=1;i<21;i++)
+ bandLogE[21*c+i] = MAX16(bandLogE[21*c+i], bandLogE[21*c+i-1]-QCONST16(1.f, DB_SHIFT));
+ for (i=19;i>=0;i--)
+ bandLogE[21*c+i] = MAX16(bandLogE[21*c+i], bandLogE[21*c+i+1]-QCONST16(2.f, DB_SHIFT));
+ if (pos[c]==1)
+ {
+ for (i=0;i<21;i++)
+ maskLogE[0][i] = logSum(maskLogE[0][i], bandLogE[21*c+i]);
+ } else if (pos[c]==3)
+ {
+ for (i=0;i<21;i++)
+ maskLogE[2][i] = logSum(maskLogE[2][i], bandLogE[21*c+i]);
+ } else if (pos[c]==2)
+ {
+ for (i=0;i<21;i++)
+ {
+ maskLogE[0][i] = logSum(maskLogE[0][i], bandLogE[21*c+i]-QCONST16(.5f, DB_SHIFT));
+ maskLogE[2][i] = logSum(maskLogE[2][i], bandLogE[21*c+i]-QCONST16(.5f, DB_SHIFT));
+ }
+ }
+#if 0
+ for (i=0;i<21;i++)
+ printf("%f ", bandLogE[21*c+i]);
+ float sum=0;
+ for (i=0;i<21;i++)
+ sum += bandLogE[21*c+i];
+ printf("%f ", sum/21);
+#endif
+ OPUS_COPY(mem+c*overlap, in+frame_size, overlap);
+ }
+ for (i=0;i<21;i++)
+ maskLogE[1][i] = MIN32(maskLogE[0][i],maskLogE[2][i]);
+ channel_offset = HALF16(celt_log2(QCONST32(2.f,14)/(channels-1)));
+ for (c=0;c<3;c++)
+ for (i=0;i<21;i++)
+ maskLogE[c][i] += channel_offset;
+#if 0
+ for (c=0;c<3;c++)
+ {
+ for (i=0;i<21;i++)
+ printf("%f ", maskLogE[c][i]);
+ }
+#endif
+ for (c=0;c<channels;c++)
+ {
+ opus_val16 *mask;
+ if (pos[c]!=0)
+ {
+ mask = &maskLogE[pos[c]-1][0];
+ for (i=0;i<21;i++)
+ bandLogE[21*c+i] = bandLogE[21*c+i] - mask[i];
+ } else {
+ for (i=0;i<21;i++)
+ bandLogE[21*c+i] = 0;
+ }
+#if 0
+ for (i=0;i<21;i++)
+ printf("%f ", bandLogE[21*c+i]);
+ printf("\n");
+#endif
+#if 0
+ float sum=0;
+ for (i=0;i<21;i++)
+ sum += bandLogE[21*c+i];
+ printf("%f ", sum/(float)QCONST32(21.f, DB_SHIFT));
+ printf("\n");
+#endif
+ }
+ RESTORE_STACK;
+}
+
+opus_int32 opus_multistream_encoder_get_size(int nb_streams, int nb_coupled_streams)
+{
+ int coupled_size;
+ int mono_size;
+
+ if(nb_streams<1||nb_coupled_streams>nb_streams||nb_coupled_streams<0)return 0;
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+ return align(sizeof(OpusMSEncoder))
+ + nb_coupled_streams * align(coupled_size)
+ + (nb_streams-nb_coupled_streams) * align(mono_size);
+}
+
+opus_int32 opus_multistream_surround_encoder_get_size(int channels, int mapping_family)
+{
+ int nb_streams;
+ int nb_coupled_streams;
+ opus_int32 size;
+
+ if (mapping_family==0)
+ {
+ if (channels==1)
+ {
+ nb_streams=1;
+ nb_coupled_streams=0;
+ } else if (channels==2)
+ {
+ nb_streams=1;
+ nb_coupled_streams=1;
+ } else
+ return 0;
+ } else if (mapping_family==1 && channels<=8 && channels>=1)
+ {
+ nb_streams=vorbis_mappings[channels-1].nb_streams;
+ nb_coupled_streams=vorbis_mappings[channels-1].nb_coupled_streams;
+ } else if (mapping_family==255)
+ {
+ nb_streams=channels;
+ nb_coupled_streams=0;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ } else if (mapping_family==254)
+ {
+ nb_streams=channels;
+ nb_coupled_streams=0;
+#endif
+ } else
+ return 0;
+ size = opus_multistream_encoder_get_size(nb_streams, nb_coupled_streams);
+ if (channels>2)
+ {
+ size += channels*(120*sizeof(opus_val32) + sizeof(opus_val32));
+ }
+ return size;
+}
+
+static int opus_multistream_encoder_init_impl(
+ OpusMSEncoder *st,
+ opus_int32 Fs,
+ int channels,
+ int streams,
+ int coupled_streams,
+ const unsigned char *mapping,
+ int application,
+ MappingType mapping_type
+)
+{
+ int coupled_size;
+ int mono_size;
+ int i, ret;
+ char *ptr;
+
+ if ((channels>255) || (channels<1) || (coupled_streams>streams) ||
+ (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams))
+ return OPUS_BAD_ARG;
+
+ st->arch = opus_select_arch();
+ st->layout.nb_channels = channels;
+ st->layout.nb_streams = streams;
+ st->layout.nb_coupled_streams = coupled_streams;
+ st->subframe_mem[0]=st->subframe_mem[1]=st->subframe_mem[2]=0;
+ if (mapping_type != MAPPING_TYPE_SURROUND)
+ st->lfe_stream = -1;
+ st->bitrate_bps = OPUS_AUTO;
+ st->application = application;
+ st->variable_duration = OPUS_FRAMESIZE_ARG;
+ for (i=0;i<st->layout.nb_channels;i++)
+ st->layout.mapping[i] = mapping[i];
+ if (!validate_layout(&st->layout) || !validate_encoder_layout(&st->layout))
+ return OPUS_BAD_ARG;
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+
+ for (i=0;i<st->layout.nb_coupled_streams;i++)
+ {
+ ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 2, application);
+ if(ret!=OPUS_OK)return ret;
+ if (i==st->lfe_stream)
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_SET_LFE(1));
+ ptr += align(coupled_size);
+ }
+ for (;i<st->layout.nb_streams;i++)
+ {
+ ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 1, application);
+ if (i==st->lfe_stream)
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_SET_LFE(1));
+ if(ret!=OPUS_OK)return ret;
+ ptr += align(mono_size);
+ }
+ if (mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ OPUS_CLEAR(ms_get_preemph_mem(st), channels);
+ OPUS_CLEAR(ms_get_window_mem(st), channels*120);
+ }
+ st->mapping_type = mapping_type;
+ return OPUS_OK;
+}
+
+int opus_multistream_encoder_init(
+ OpusMSEncoder *st,
+ opus_int32 Fs,
+ int channels,
+ int streams,
+ int coupled_streams,
+ const unsigned char *mapping,
+ int application
+)
+{
+ return opus_multistream_encoder_init_impl(st, Fs, channels, streams,
+ coupled_streams, mapping,
+ application, MAPPING_TYPE_NONE);
+}
+
+int opus_multistream_surround_encoder_init(
+ OpusMSEncoder *st,
+ opus_int32 Fs,
+ int channels,
+ int mapping_family,
+ int *streams,
+ int *coupled_streams,
+ unsigned char *mapping,
+ int application
+)
+{
+ MappingType mapping_type;
+
+ if ((channels>255) || (channels<1))
+ return OPUS_BAD_ARG;
+ st->lfe_stream = -1;
+ if (mapping_family==0)
+ {
+ if (channels==1)
+ {
+ *streams=1;
+ *coupled_streams=0;
+ mapping[0]=0;
+ } else if (channels==2)
+ {
+ *streams=1;
+ *coupled_streams=1;
+ mapping[0]=0;
+ mapping[1]=1;
+ } else
+ return OPUS_UNIMPLEMENTED;
+ } else if (mapping_family==1 && channels<=8 && channels>=1)
+ {
+ int i;
+ *streams=vorbis_mappings[channels-1].nb_streams;
+ *coupled_streams=vorbis_mappings[channels-1].nb_coupled_streams;
+ for (i=0;i<channels;i++)
+ mapping[i] = vorbis_mappings[channels-1].mapping[i];
+ if (channels>=6)
+ st->lfe_stream = *streams-1;
+ } else if (mapping_family==255)
+ {
+ int i;
+ *streams=channels;
+ *coupled_streams=0;
+ for(i=0;i<channels;i++)
+ mapping[i] = i;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ } else if (mapping_family==254)
+ {
+ int i;
+ *streams=channels;
+ *coupled_streams=0;
+ for(i=0;i<channels;i++)
+ mapping[i] = i;
+#endif
+ } else
+ return OPUS_UNIMPLEMENTED;
+
+ if (channels>2 && mapping_family==1) {
+ mapping_type = MAPPING_TYPE_SURROUND;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ } else if (mapping_family==254)
+ {
+ mapping_type = MAPPING_TYPE_AMBISONICS;
+#endif
+ } else
+ {
+ mapping_type = MAPPING_TYPE_NONE;
+ }
+ return opus_multistream_encoder_init_impl(st, Fs, channels, *streams,
+ *coupled_streams, mapping,
+ application, mapping_type);
+}
+
+OpusMSEncoder *opus_multistream_encoder_create(
+ opus_int32 Fs,
+ int channels,
+ int streams,
+ int coupled_streams,
+ const unsigned char *mapping,
+ int application,
+ int *error
+)
+{
+ int ret;
+ OpusMSEncoder *st;
+ if ((channels>255) || (channels<1) || (coupled_streams>streams) ||
+ (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams))
+ {
+ if (error)
+ *error = OPUS_BAD_ARG;
+ return NULL;
+ }
+ st = (OpusMSEncoder *)opus_alloc(opus_multistream_encoder_get_size(streams, coupled_streams));
+ if (st==NULL)
+ {
+ if (error)
+ *error = OPUS_ALLOC_FAIL;
+ return NULL;
+ }
+ ret = opus_multistream_encoder_init(st, Fs, channels, streams, coupled_streams, mapping, application);
+ if (ret != OPUS_OK)
+ {
+ opus_free(st);
+ st = NULL;
+ }
+ if (error)
+ *error = ret;
+ return st;
+}
+
+OpusMSEncoder *opus_multistream_surround_encoder_create(
+ opus_int32 Fs,
+ int channels,
+ int mapping_family,
+ int *streams,
+ int *coupled_streams,
+ unsigned char *mapping,
+ int application,
+ int *error
+)
+{
+ int ret;
+ opus_int32 size;
+ OpusMSEncoder *st;
+ if ((channels>255) || (channels<1))
+ {
+ if (error)
+ *error = OPUS_BAD_ARG;
+ return NULL;
+ }
+ size = opus_multistream_surround_encoder_get_size(channels, mapping_family);
+ if (!size)
+ {
+ if (error)
+ *error = OPUS_UNIMPLEMENTED;
+ return NULL;
+ }
+ st = (OpusMSEncoder *)opus_alloc(size);
+ if (st==NULL)
+ {
+ if (error)
+ *error = OPUS_ALLOC_FAIL;
+ return NULL;
+ }
+ ret = opus_multistream_surround_encoder_init(st, Fs, channels, mapping_family, streams, coupled_streams, mapping, application);
+ if (ret != OPUS_OK)
+ {
+ opus_free(st);
+ st = NULL;
+ }
+ if (error)
+ *error = ret;
+ return st;
+}
+
+static void surround_rate_allocation(
+ OpusMSEncoder *st,
+ opus_int32 *rate,
+ int frame_size,
+ opus_int32 Fs
+ )
+{
+ int i;
+ opus_int32 channel_rate;
+ int stream_offset;
+ int lfe_offset;
+ int coupled_ratio; /* Q8 */
+ int lfe_ratio; /* Q8 */
+
+ if (st->bitrate_bps > st->layout.nb_channels*40000)
+ stream_offset = 20000;
+ else
+ stream_offset = st->bitrate_bps/st->layout.nb_channels/2;
+ stream_offset += 60*(Fs/frame_size-50);
+ /* We start by giving each stream (coupled or uncoupled) the same bitrate.
+ This models the main saving of coupled channels over uncoupled. */
+ /* The LFE stream is an exception to the above and gets fewer bits. */
+ lfe_offset = 3500 + 60*(Fs/frame_size-50);
+ /* Coupled streams get twice the mono rate after the first 20 kb/s. */
+ coupled_ratio = 512;
+ /* Should depend on the bitrate, for now we assume LFE gets 1/8 the bits of mono */
+ lfe_ratio = 32;
+
+ /* Compute bitrate allocation between streams */
+ if (st->bitrate_bps==OPUS_AUTO)
+ {
+ channel_rate = Fs+60*Fs/frame_size;
+ } else if (st->bitrate_bps==OPUS_BITRATE_MAX)
+ {
+ channel_rate = 300000;
+ } else {
+ int nb_lfe;
+ int nb_uncoupled;
+ int nb_coupled;
+ int total;
+ nb_lfe = (st->lfe_stream!=-1);
+ nb_coupled = st->layout.nb_coupled_streams;
+ nb_uncoupled = st->layout.nb_streams-nb_coupled-nb_lfe;
+ total = (nb_uncoupled<<8) /* mono */
+ + coupled_ratio*nb_coupled /* stereo */
+ + nb_lfe*lfe_ratio;
+ channel_rate = 256*(st->bitrate_bps-lfe_offset*nb_lfe-stream_offset*(nb_coupled+nb_uncoupled))/total;
+ }
+#ifndef FIXED_POINT
+ if (st->variable_duration==OPUS_FRAMESIZE_VARIABLE && frame_size != Fs/50)
+ {
+ opus_int32 bonus;
+ bonus = 60*(Fs/frame_size-50);
+ channel_rate += bonus;
+ }
+#endif
+
+ for (i=0;i<st->layout.nb_streams;i++)
+ {
+ if (i<st->layout.nb_coupled_streams)
+ rate[i] = stream_offset+(channel_rate*coupled_ratio>>8);
+ else if (i!=st->lfe_stream)
+ rate[i] = stream_offset+channel_rate;
+ else
+ rate[i] = lfe_offset+(channel_rate*lfe_ratio>>8);
+ }
+}
+
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+static void ambisonics_rate_allocation(
+ OpusMSEncoder *st,
+ opus_int32 *rate,
+ int frame_size,
+ opus_int32 Fs
+ )
+{
+ int i;
+ int non_mono_rate;
+ int total_rate;
+
+ /* The mono channel gets (rate_ratio_num / rate_ratio_den) times as many bits
+ * as all other channels */
+ const int rate_ratio_num = 4;
+ const int rate_ratio_den = 3;
+ const int num_channels = st->layout.nb_streams;
+
+ if (st->bitrate_bps==OPUS_AUTO)
+ {
+ total_rate = num_channels * (20000 + st->layout.nb_streams*(Fs+60*Fs/frame_size));
+ } else if (st->bitrate_bps==OPUS_BITRATE_MAX)
+ {
+ total_rate = num_channels * 320000;
+ } else {
+ total_rate = st->bitrate_bps;
+ }
+
+ /* Let y be the non-mono rate and let p, q be integers such that the mono
+ * channel rate is (p/q) * y.
+ * Also let T be the total bitrate to allocate. Then
+ * (n - 1) y + (p/q) y = T
+ * y = (T q) / (qn - q + p)
+ */
+ non_mono_rate =
+ total_rate * rate_ratio_den
+ / (rate_ratio_den*num_channels + rate_ratio_num - rate_ratio_den);
+
+#ifndef FIXED_POINT
+ if (st->variable_duration==OPUS_FRAMESIZE_VARIABLE && frame_size != Fs/50)
+ {
+ opus_int32 bonus = 60*(Fs/frame_size-50);
+ non_mono_rate += bonus;
+ }
+#endif
+
+ rate[0] = total_rate - (num_channels - 1) * non_mono_rate;
+ for (i=1;i<st->layout.nb_streams;i++)
+ {
+ rate[i] = non_mono_rate;
+ }
+}
+#endif /* ENABLE_EXPERIMENTAL_AMBISONICS */
+
+static opus_int32 rate_allocation(
+ OpusMSEncoder *st,
+ opus_int32 *rate,
+ int frame_size
+ )
+{
+ int i;
+ opus_int32 rate_sum=0;
+ opus_int32 Fs;
+ char *ptr;
+
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs));
+
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ if (st->mapping_type == MAPPING_TYPE_AMBISONICS) {
+ ambisonics_rate_allocation(st, rate, frame_size, Fs);
+ } else
+#endif
+ {
+ surround_rate_allocation(st, rate, frame_size, Fs);
+ }
+
+ for (i=0;i<st->layout.nb_streams;i++)
+ {
+ rate[i] = IMAX(rate[i], 500);
+ rate_sum += rate[i];
+ }
+ return rate_sum;
+}
+
+/* Max size in case the encoder decides to return three frames */
+#define MS_FRAME_TMP (3*1275+7)
+static int opus_multistream_encode_native
+(
+ OpusMSEncoder *st,
+ opus_copy_channel_in_func copy_channel_in,
+ const void *pcm,
+ int analysis_frame_size,
+ unsigned char *data,
+ opus_int32 max_data_bytes,
+ int lsb_depth,
+ downmix_func downmix,
+ int float_api
+)
+{
+ opus_int32 Fs;
+ int coupled_size;
+ int mono_size;
+ int s;
+ char *ptr;
+ int tot_size;
+ VARDECL(opus_val16, buf);
+ VARDECL(opus_val16, bandSMR);
+ unsigned char tmp_data[MS_FRAME_TMP];
+ OpusRepacketizer rp;
+ opus_int32 vbr;
+ const CELTMode *celt_mode;
+ opus_int32 bitrates[256];
+ opus_val16 bandLogE[42];
+ opus_val32 *mem = NULL;
+ opus_val32 *preemph_mem=NULL;
+ int frame_size;
+ opus_int32 rate_sum;
+ opus_int32 smallest_packet;
+ ALLOC_STACK;
+
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ preemph_mem = ms_get_preemph_mem(st);
+ mem = ms_get_window_mem(st);
+ }
+
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs));
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_VBR(&vbr));
+ opus_encoder_ctl((OpusEncoder*)ptr, CELT_GET_MODE(&celt_mode));
+
+ {
+ opus_int32 delay_compensation;
+ int channels;
+
+ channels = st->layout.nb_streams + st->layout.nb_coupled_streams;
+ opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_LOOKAHEAD(&delay_compensation));
+ delay_compensation -= Fs/400;
+ frame_size = compute_frame_size(pcm, analysis_frame_size,
+ st->variable_duration, channels, Fs, st->bitrate_bps,
+ delay_compensation, downmix
+#ifndef DISABLE_FLOAT_API
+ , st->subframe_mem
+#endif
+ );
+ }
+
+ if (400*frame_size < Fs)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+ /* Validate frame_size before using it to allocate stack space.
+ This mirrors the checks in opus_encode[_float](). */
+ if (400*frame_size != Fs && 200*frame_size != Fs &&
+ 100*frame_size != Fs && 50*frame_size != Fs &&
+ 25*frame_size != Fs && 50*frame_size != 3*Fs)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+
+ /* Smallest packet the encoder can produce. */
+ smallest_packet = st->layout.nb_streams*2-1;
+ if (max_data_bytes < smallest_packet)
+ {
+ RESTORE_STACK;
+ return OPUS_BUFFER_TOO_SMALL;
+ }
+ ALLOC(buf, 2*frame_size, opus_val16);
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+
+ ALLOC(bandSMR, 21*st->layout.nb_channels, opus_val16);
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ surround_analysis(celt_mode, pcm, bandSMR, mem, preemph_mem, frame_size, 120, st->layout.nb_channels, Fs, copy_channel_in, st->arch);
+ }
+
+ /* Compute bitrate allocation between streams (this could be a lot better) */
+ rate_sum = rate_allocation(st, bitrates, frame_size);
+
+ if (!vbr)
+ {
+ if (st->bitrate_bps == OPUS_AUTO)
+ {
+ max_data_bytes = IMIN(max_data_bytes, 3*rate_sum/(3*8*Fs/frame_size));
+ } else if (st->bitrate_bps != OPUS_BITRATE_MAX)
+ {
+ max_data_bytes = IMIN(max_data_bytes, IMAX(smallest_packet,
+ 3*st->bitrate_bps/(3*8*Fs/frame_size)));
+ }
+ }
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusEncoder *enc;
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrates[s]));
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ opus_int32 equiv_rate;
+ equiv_rate = st->bitrate_bps;
+ if (frame_size*50 < Fs)
+ equiv_rate -= 60*(Fs/frame_size - 50)*st->layout.nb_channels;
+ if (equiv_rate > 10000*st->layout.nb_channels)
+ opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND));
+ else if (equiv_rate > 7000*st->layout.nb_channels)
+ opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_SUPERWIDEBAND));
+ else if (equiv_rate > 5000*st->layout.nb_channels)
+ opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND));
+ else
+ opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND));
+ if (s < st->layout.nb_coupled_streams)
+ {
+ /* To preserve the spatial image, force stereo CELT on coupled streams */
+ opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY));
+ opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(2));
+ }
+ }
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+ else if (st->mapping_type == MAPPING_TYPE_AMBISONICS) {
+ opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY));
+ }
+#endif
+ }
+
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ /* Counting ToC */
+ tot_size = 0;
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusEncoder *enc;
+ int len;
+ int curr_max;
+ int c1, c2;
+ int ret;
+
+ opus_repacketizer_init(&rp);
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ {
+ int i;
+ int left, right;
+ left = get_left_channel(&st->layout, s, -1);
+ right = get_right_channel(&st->layout, s, -1);
+ (*copy_channel_in)(buf, 2,
+ pcm, st->layout.nb_channels, left, frame_size);
+ (*copy_channel_in)(buf+1, 2,
+ pcm, st->layout.nb_channels, right, frame_size);
+ ptr += align(coupled_size);
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ for (i=0;i<21;i++)
+ {
+ bandLogE[i] = bandSMR[21*left+i];
+ bandLogE[21+i] = bandSMR[21*right+i];
+ }
+ }
+ c1 = left;
+ c2 = right;
+ } else {
+ int i;
+ int chan = get_mono_channel(&st->layout, s, -1);
+ (*copy_channel_in)(buf, 1,
+ pcm, st->layout.nb_channels, chan, frame_size);
+ ptr += align(mono_size);
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ for (i=0;i<21;i++)
+ bandLogE[i] = bandSMR[21*chan+i];
+ }
+ c1 = chan;
+ c2 = -1;
+ }
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ opus_encoder_ctl(enc, OPUS_SET_ENERGY_MASK(bandLogE));
+ /* number of bytes left (+Toc) */
+ curr_max = max_data_bytes - tot_size;
+ /* Reserve one byte for the last stream and two for the others */
+ curr_max -= IMAX(0,2*(st->layout.nb_streams-s-1)-1);
+ curr_max = IMIN(curr_max,MS_FRAME_TMP);
+ /* Repacketizer will add one or two bytes for self-delimited frames */
+ if (s != st->layout.nb_streams-1) curr_max -= curr_max>253 ? 2 : 1;
+ if (!vbr && s == st->layout.nb_streams-1)
+ opus_encoder_ctl(enc, OPUS_SET_BITRATE(curr_max*(8*Fs/frame_size)));
+ len = opus_encode_native(enc, buf, frame_size, tmp_data, curr_max, lsb_depth,
+ pcm, analysis_frame_size, c1, c2, st->layout.nb_channels, downmix, float_api);
+ if (len<0)
+ {
+ RESTORE_STACK;
+ return len;
+ }
+ /* We need to use the repacketizer to add the self-delimiting lengths
+ while taking into account the fact that the encoder can now return
+ more than one frame at a time (e.g. 60 ms CELT-only) */
+ ret = opus_repacketizer_cat(&rp, tmp_data, len);
+ /* If the opus_repacketizer_cat() fails, then something's seriously wrong
+ with the encoder. */
+ if (ret != OPUS_OK)
+ {
+ RESTORE_STACK;
+ return OPUS_INTERNAL_ERROR;
+ }
+ len = opus_repacketizer_out_range_impl(&rp, 0, opus_repacketizer_get_nb_frames(&rp),
+ data, max_data_bytes-tot_size, s != st->layout.nb_streams-1, !vbr && s == st->layout.nb_streams-1);
+ data += len;
+ tot_size += len;
+ }
+ /*printf("\n");*/
+ RESTORE_STACK;
+ return tot_size;
+}
+
+#if !defined(DISABLE_FLOAT_API)
+static void opus_copy_channel_in_float(
+ opus_val16 *dst,
+ int dst_stride,
+ const void *src,
+ int src_stride,
+ int src_channel,
+ int frame_size
+)
+{
+ const float *float_src;
+ opus_int32 i;
+ float_src = (const float *)src;
+ for (i=0;i<frame_size;i++)
+#if defined(FIXED_POINT)
+ dst[i*dst_stride] = FLOAT2INT16(float_src[i*src_stride+src_channel]);
+#else
+ dst[i*dst_stride] = float_src[i*src_stride+src_channel];
+#endif
+}
+#endif
+
+static void opus_copy_channel_in_short(
+ opus_val16 *dst,
+ int dst_stride,
+ const void *src,
+ int src_stride,
+ int src_channel,
+ int frame_size
+)
+{
+ const opus_int16 *short_src;
+ opus_int32 i;
+ short_src = (const opus_int16 *)src;
+ for (i=0;i<frame_size;i++)
+#if defined(FIXED_POINT)
+ dst[i*dst_stride] = short_src[i*src_stride+src_channel];
+#else
+ dst[i*dst_stride] = (1/32768.f)*short_src[i*src_stride+src_channel];
+#endif
+}
+
+
+#ifdef FIXED_POINT
+int opus_multistream_encode(
+ OpusMSEncoder *st,
+ const opus_val16 *pcm,
+ int frame_size,
+ unsigned char *data,
+ opus_int32 max_data_bytes
+)
+{
+ return opus_multistream_encode_native(st, opus_copy_channel_in_short,
+ pcm, frame_size, data, max_data_bytes, 16, downmix_int, 0);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_multistream_encode_float(
+ OpusMSEncoder *st,
+ const float *pcm,
+ int frame_size,
+ unsigned char *data,
+ opus_int32 max_data_bytes
+)
+{
+ return opus_multistream_encode_native(st, opus_copy_channel_in_float,
+ pcm, frame_size, data, max_data_bytes, 16, downmix_float, 1);
+}
+#endif
+
+#else
+
+int opus_multistream_encode_float
+(
+ OpusMSEncoder *st,
+ const opus_val16 *pcm,
+ int frame_size,
+ unsigned char *data,
+ opus_int32 max_data_bytes
+)
+{
+ return opus_multistream_encode_native(st, opus_copy_channel_in_float,
+ pcm, frame_size, data, max_data_bytes, 24, downmix_float, 1);
+}
+
+int opus_multistream_encode(
+ OpusMSEncoder *st,
+ const opus_int16 *pcm,
+ int frame_size,
+ unsigned char *data,
+ opus_int32 max_data_bytes
+)
+{
+ return opus_multistream_encode_native(st, opus_copy_channel_in_short,
+ pcm, frame_size, data, max_data_bytes, 16, downmix_int, 0);
+}
+#endif
+
+int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...)
+{
+ va_list ap;
+ int coupled_size, mono_size;
+ char *ptr;
+ int ret = OPUS_OK;
+
+ va_start(ap, request);
+
+ coupled_size = opus_encoder_get_size(2);
+ mono_size = opus_encoder_get_size(1);
+ ptr = (char*)st + align(sizeof(OpusMSEncoder));
+ switch (request)
+ {
+ case OPUS_SET_BITRATE_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value<0 && value!=OPUS_AUTO && value!=OPUS_BITRATE_MAX)
+ {
+ goto bad_arg;
+ }
+ st->bitrate_bps = value;
+ }
+ break;
+ case OPUS_GET_BITRATE_REQUEST:
+ {
+ int s;
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = 0;
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ opus_int32 rate;
+ OpusEncoder *enc;
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ opus_encoder_ctl(enc, request, &rate);
+ *value += rate;
+ }
+ }
+ break;
+ case OPUS_GET_LSB_DEPTH_REQUEST:
+ case OPUS_GET_VBR_REQUEST:
+ case OPUS_GET_APPLICATION_REQUEST:
+ case OPUS_GET_BANDWIDTH_REQUEST:
+ case OPUS_GET_COMPLEXITY_REQUEST:
+ case OPUS_GET_PACKET_LOSS_PERC_REQUEST:
+ case OPUS_GET_DTX_REQUEST:
+ case OPUS_GET_VOICE_RATIO_REQUEST:
+ case OPUS_GET_VBR_CONSTRAINT_REQUEST:
+ case OPUS_GET_SIGNAL_REQUEST:
+ case OPUS_GET_LOOKAHEAD_REQUEST:
+ case OPUS_GET_SAMPLE_RATE_REQUEST:
+ case OPUS_GET_INBAND_FEC_REQUEST:
+ case OPUS_GET_FORCE_CHANNELS_REQUEST:
+ case OPUS_GET_PREDICTION_DISABLED_REQUEST:
+ {
+ OpusEncoder *enc;
+ /* For int32* GET params, just query the first stream */
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ enc = (OpusEncoder*)ptr;
+ ret = opus_encoder_ctl(enc, request, value);
+ }
+ break;
+ case OPUS_GET_FINAL_RANGE_REQUEST:
+ {
+ int s;
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ opus_uint32 tmp;
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value=0;
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusEncoder *enc;
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_encoder_ctl(enc, request, &tmp);
+ if (ret != OPUS_OK) break;
+ *value ^= tmp;
+ }
+ }
+ break;
+ case OPUS_SET_LSB_DEPTH_REQUEST:
+ case OPUS_SET_COMPLEXITY_REQUEST:
+ case OPUS_SET_VBR_REQUEST:
+ case OPUS_SET_VBR_CONSTRAINT_REQUEST:
+ case OPUS_SET_MAX_BANDWIDTH_REQUEST:
+ case OPUS_SET_BANDWIDTH_REQUEST:
+ case OPUS_SET_SIGNAL_REQUEST:
+ case OPUS_SET_APPLICATION_REQUEST:
+ case OPUS_SET_INBAND_FEC_REQUEST:
+ case OPUS_SET_PACKET_LOSS_PERC_REQUEST:
+ case OPUS_SET_DTX_REQUEST:
+ case OPUS_SET_FORCE_MODE_REQUEST:
+ case OPUS_SET_FORCE_CHANNELS_REQUEST:
+ case OPUS_SET_PREDICTION_DISABLED_REQUEST:
+ {
+ int s;
+ /* This works for int32 params */
+ opus_int32 value = va_arg(ap, opus_int32);
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusEncoder *enc;
+
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_encoder_ctl(enc, request, value);
+ if (ret != OPUS_OK)
+ break;
+ }
+ }
+ break;
+ case OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST:
+ {
+ int s;
+ opus_int32 stream_id;
+ OpusEncoder **value;
+ stream_id = va_arg(ap, opus_int32);
+ if (stream_id<0 || stream_id >= st->layout.nb_streams)
+ ret = OPUS_BAD_ARG;
+ value = va_arg(ap, OpusEncoder**);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ for (s=0;s<stream_id;s++)
+ {
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ }
+ *value = (OpusEncoder*)ptr;
+ }
+ break;
+ case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ st->variable_duration = value;
+ }
+ break;
+ case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->variable_duration;
+ }
+ break;
+ case OPUS_RESET_STATE:
+ {
+ int s;
+ st->subframe_mem[0] = st->subframe_mem[1] = st->subframe_mem[2] = 0;
+ if (st->mapping_type == MAPPING_TYPE_SURROUND)
+ {
+ OPUS_CLEAR(ms_get_preemph_mem(st), st->layout.nb_channels);
+ OPUS_CLEAR(ms_get_window_mem(st), st->layout.nb_channels*120);
+ }
+ for (s=0;s<st->layout.nb_streams;s++)
+ {
+ OpusEncoder *enc;
+ enc = (OpusEncoder*)ptr;
+ if (s < st->layout.nb_coupled_streams)
+ ptr += align(coupled_size);
+ else
+ ptr += align(mono_size);
+ ret = opus_encoder_ctl(enc, OPUS_RESET_STATE);
+ if (ret != OPUS_OK)
+ break;
+ }
+ }
+ break;
+ default:
+ ret = OPUS_UNIMPLEMENTED;
+ break;
+ }
+
+ va_end(ap);
+ return ret;
+bad_arg:
+ va_end(ap);
+ return OPUS_BAD_ARG;
+}
+
+void opus_multistream_encoder_destroy(OpusMSEncoder *st)
+{
+ opus_free(st);
+}
diff --git a/media/libopus/src/opus_private.h b/media/libopus/src/opus_private.h
new file mode 100644
index 000000000..3b62eed09
--- /dev/null
+++ b/media/libopus/src/opus_private.h
@@ -0,0 +1,134 @@
+/* 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"
+#include "celt.h"
+
+#include <stddef.h> /* offsetof */
+
+struct OpusRepacketizer {
+ unsigned char toc;
+ int nb_frames;
+ const unsigned char *frames[48];
+ opus_int16 len[48];
+ int framesize;
+};
+
+typedef struct ChannelLayout {
+ int nb_channels;
+ int nb_streams;
+ int nb_coupled_streams;
+ unsigned char mapping[256];
+} ChannelLayout;
+
+int validate_layout(const ChannelLayout *layout);
+int get_left_channel(const ChannelLayout *layout, int stream_id, int prev);
+int get_right_channel(const ChannelLayout *layout, int stream_id, int prev);
+int get_mono_channel(const ChannelLayout *layout, int stream_id, int prev);
+
+
+
+#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)
+
+typedef void (*downmix_func)(const void *, opus_val32 *, int, int, int, int, int);
+void downmix_float(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C);
+void downmix_int(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C);
+
+int encode_size(int size, unsigned char *data);
+
+opus_int32 frame_size_select(opus_int32 frame_size, int variable_duration, opus_int32 Fs);
+
+opus_int32 compute_frame_size(const void *analysis_pcm, int frame_size,
+ int variable_duration, int C, opus_int32 Fs, int bitrate_bps,
+ int delay_compensation, downmix_func downmix
+#ifndef DISABLE_FLOAT_API
+ , float *subframe_mem
+#endif
+ );
+
+opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
+ unsigned char *data, opus_int32 out_data_bytes, int lsb_depth,
+ const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2,
+ int analysis_channels, downmix_func downmix, int float_api);
+
+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,
+ opus_int32 *packet_offset, int soft_clip);
+
+/* Make sure everything is properly aligned. */
+static OPUS_INLINE int align(int i)
+{
+ struct foo {char c; union { void* p; opus_int32 i; opus_val32 v; } u;};
+
+ unsigned int alignment = offsetof(struct foo, u);
+
+ /* Optimizing compilers should optimize div and multiply into and
+ for all sensible alignment values. */
+ return ((i + alignment - 1) / alignment) * alignment;
+}
+
+int opus_packet_parse_impl(const unsigned char *data, opus_int32 len,
+ int self_delimited, unsigned char *out_toc,
+ const unsigned char *frames[48], opus_int16 size[48],
+ int *payload_offset, opus_int32 *packet_offset);
+
+opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end,
+ unsigned char *data, opus_int32 maxlen, int self_delimited, int pad);
+
+int pad_frame(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+#endif /* OPUS_PRIVATE_H */
diff --git a/media/libopus/src/repacketizer.c b/media/libopus/src/repacketizer.c
new file mode 100644
index 000000000..c80ee7f00
--- /dev/null
+++ b/media/libopus/src/repacketizer.c
@@ -0,0 +1,348 @@
+/* Copyright (c) 2011 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 "config.h"
+#endif
+
+#include "opus.h"
+#include "opus_private.h"
+#include "os_support.h"
+
+
+int opus_repacketizer_get_size(void)
+{
+ return sizeof(OpusRepacketizer);
+}
+
+OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp)
+{
+ rp->nb_frames = 0;
+ return rp;
+}
+
+OpusRepacketizer *opus_repacketizer_create(void)
+{
+ OpusRepacketizer *rp;
+ rp=(OpusRepacketizer *)opus_alloc(opus_repacketizer_get_size());
+ if(rp==NULL)return NULL;
+ return opus_repacketizer_init(rp);
+}
+
+void opus_repacketizer_destroy(OpusRepacketizer *rp)
+{
+ opus_free(rp);
+}
+
+static int opus_repacketizer_cat_impl(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len, int self_delimited)
+{
+ unsigned char tmp_toc;
+ int curr_nb_frames,ret;
+ /* Set of check ToC */
+ if (len<1) return OPUS_INVALID_PACKET;
+ if (rp->nb_frames == 0)
+ {
+ rp->toc = data[0];
+ rp->framesize = opus_packet_get_samples_per_frame(data, 8000);
+ } else if ((rp->toc&0xFC) != (data[0]&0xFC))
+ {
+ /*fprintf(stderr, "toc mismatch: 0x%x vs 0x%x\n", rp->toc, data[0]);*/
+ return OPUS_INVALID_PACKET;
+ }
+ curr_nb_frames = opus_packet_get_nb_frames(data, len);
+ if(curr_nb_frames<1) return OPUS_INVALID_PACKET;
+
+ /* Check the 120 ms maximum packet size */
+ if ((curr_nb_frames+rp->nb_frames)*rp->framesize > 960)
+ {
+ return OPUS_INVALID_PACKET;
+ }
+
+ ret=opus_packet_parse_impl(data, len, self_delimited, &tmp_toc, &rp->frames[rp->nb_frames], &rp->len[rp->nb_frames], NULL, NULL);
+ if(ret<1)return ret;
+
+ rp->nb_frames += curr_nb_frames;
+ return OPUS_OK;
+}
+
+int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len)
+{
+ return opus_repacketizer_cat_impl(rp, data, len, 0);
+}
+
+int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp)
+{
+ return rp->nb_frames;
+}
+
+opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end,
+ unsigned char *data, opus_int32 maxlen, int self_delimited, int pad)
+{
+ int i, count;
+ opus_int32 tot_size;
+ opus_int16 *len;
+ const unsigned char **frames;
+ unsigned char * ptr;
+
+ if (begin<0 || begin>=end || end>rp->nb_frames)
+ {
+ /*fprintf(stderr, "%d %d %d\n", begin, end, rp->nb_frames);*/
+ return OPUS_BAD_ARG;
+ }
+ count = end-begin;
+
+ len = rp->len+begin;
+ frames = rp->frames+begin;
+ if (self_delimited)
+ tot_size = 1 + (len[count-1]>=252);
+ else
+ tot_size = 0;
+
+ ptr = data;
+ if (count==1)
+ {
+ /* Code 0 */
+ tot_size += len[0]+1;
+ if (tot_size > maxlen)
+ return OPUS_BUFFER_TOO_SMALL;
+ *ptr++ = rp->toc&0xFC;
+ } else if (count==2)
+ {
+ if (len[1] == len[0])
+ {
+ /* Code 1 */
+ tot_size += 2*len[0]+1;
+ if (tot_size > maxlen)
+ return OPUS_BUFFER_TOO_SMALL;
+ *ptr++ = (rp->toc&0xFC) | 0x1;
+ } else {
+ /* Code 2 */
+ tot_size += len[0]+len[1]+2+(len[0]>=252);
+ if (tot_size > maxlen)
+ return OPUS_BUFFER_TOO_SMALL;
+ *ptr++ = (rp->toc&0xFC) | 0x2;
+ ptr += encode_size(len[0], ptr);
+ }
+ }
+ if (count > 2 || (pad && tot_size < maxlen))
+ {
+ /* Code 3 */
+ int vbr;
+ int pad_amount=0;
+
+ /* Restart the process for the padding case */
+ ptr = data;
+ if (self_delimited)
+ tot_size = 1 + (len[count-1]>=252);
+ else
+ tot_size = 0;
+ vbr = 0;
+ for (i=1;i<count;i++)
+ {
+ if (len[i] != len[0])
+ {
+ vbr=1;
+ break;
+ }
+ }
+ if (vbr)
+ {
+ tot_size += 2;
+ for (i=0;i<count-1;i++)
+ tot_size += 1 + (len[i]>=252) + len[i];
+ tot_size += len[count-1];
+
+ if (tot_size > maxlen)
+ return OPUS_BUFFER_TOO_SMALL;
+ *ptr++ = (rp->toc&0xFC) | 0x3;
+ *ptr++ = count | 0x80;
+ } else {
+ tot_size += count*len[0]+2;
+ if (tot_size > maxlen)
+ return OPUS_BUFFER_TOO_SMALL;
+ *ptr++ = (rp->toc&0xFC) | 0x3;
+ *ptr++ = count;
+ }
+ pad_amount = pad ? (maxlen-tot_size) : 0;
+ if (pad_amount != 0)
+ {
+ int nb_255s;
+ data[1] |= 0x40;
+ nb_255s = (pad_amount-1)/255;
+ for (i=0;i<nb_255s;i++)
+ *ptr++ = 255;
+ *ptr++ = pad_amount-255*nb_255s-1;
+ tot_size += pad_amount;
+ }
+ if (vbr)
+ {
+ for (i=0;i<count-1;i++)
+ ptr += encode_size(len[i], ptr);
+ }
+ }
+ if (self_delimited) {
+ int sdlen = encode_size(len[count-1], ptr);
+ ptr += sdlen;
+ }
+ /* Copy the actual data */
+ for (i=0;i<count;i++)
+ {
+ /* Using OPUS_MOVE() instead of OPUS_COPY() in case we're doing in-place
+ padding from opus_packet_pad or opus_packet_unpad(). */
+ celt_assert(frames[i] + len[i] <= data || ptr <= frames[i]);
+ OPUS_MOVE(ptr, frames[i], len[i]);
+ ptr += len[i];
+ }
+ if (pad)
+ {
+ /* Fill padding with zeros. */
+ while (ptr<data+maxlen)
+ *ptr++=0;
+ }
+ return tot_size;
+}
+
+opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen)
+{
+ return opus_repacketizer_out_range_impl(rp, begin, end, data, maxlen, 0, 0);
+}
+
+opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen)
+{
+ return opus_repacketizer_out_range_impl(rp, 0, rp->nb_frames, data, maxlen, 0, 0);
+}
+
+int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len)
+{
+ OpusRepacketizer rp;
+ opus_int32 ret;
+ if (len < 1)
+ return OPUS_BAD_ARG;
+ if (len==new_len)
+ return OPUS_OK;
+ else if (len > new_len)
+ return OPUS_BAD_ARG;
+ opus_repacketizer_init(&rp);
+ /* Moving payload to the end of the packet so we can do in-place padding */
+ OPUS_MOVE(data+new_len-len, data, len);
+ ret = opus_repacketizer_cat(&rp, data+new_len-len, len);
+ if (ret != OPUS_OK)
+ return ret;
+ ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, data, new_len, 0, 1);
+ if (ret > 0)
+ return OPUS_OK;
+ else
+ return ret;
+}
+
+opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len)
+{
+ OpusRepacketizer rp;
+ opus_int32 ret;
+ if (len < 1)
+ return OPUS_BAD_ARG;
+ opus_repacketizer_init(&rp);
+ ret = opus_repacketizer_cat(&rp, data, len);
+ if (ret < 0)
+ return ret;
+ ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, data, len, 0, 0);
+ celt_assert(ret > 0 && ret <= len);
+ return ret;
+}
+
+int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams)
+{
+ int s;
+ int count;
+ unsigned char toc;
+ opus_int16 size[48];
+ opus_int32 packet_offset;
+ opus_int32 amount;
+
+ if (len < 1)
+ return OPUS_BAD_ARG;
+ if (len==new_len)
+ return OPUS_OK;
+ else if (len > new_len)
+ return OPUS_BAD_ARG;
+ amount = new_len - len;
+ /* Seek to last stream */
+ for (s=0;s<nb_streams-1;s++)
+ {
+ if (len<=0)
+ return OPUS_INVALID_PACKET;
+ count = opus_packet_parse_impl(data, len, 1, &toc, NULL,
+ size, NULL, &packet_offset);
+ if (count<0)
+ return count;
+ data += packet_offset;
+ len -= packet_offset;
+ }
+ return opus_packet_pad(data, len, len+amount);
+}
+
+opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams)
+{
+ int s;
+ unsigned char toc;
+ opus_int16 size[48];
+ opus_int32 packet_offset;
+ OpusRepacketizer rp;
+ unsigned char *dst;
+ opus_int32 dst_len;
+
+ if (len < 1)
+ return OPUS_BAD_ARG;
+ dst = data;
+ dst_len = 0;
+ /* Unpad all frames */
+ for (s=0;s<nb_streams;s++)
+ {
+ opus_int32 ret;
+ int self_delimited = s!=nb_streams-1;
+ if (len<=0)
+ return OPUS_INVALID_PACKET;
+ opus_repacketizer_init(&rp);
+ ret = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL,
+ size, NULL, &packet_offset);
+ if (ret<0)
+ return ret;
+ ret = opus_repacketizer_cat_impl(&rp, data, packet_offset, self_delimited);
+ if (ret < 0)
+ return ret;
+ ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, dst, len, self_delimited, 0);
+ if (ret < 0)
+ return ret;
+ else
+ dst_len += ret;
+ dst += ret;
+ data += packet_offset;
+ len -= packet_offset;
+ }
+ return dst_len;
+}
+
diff --git a/media/libopus/src/tansig_table.h b/media/libopus/src/tansig_table.h
new file mode 100644
index 000000000..c76f844a7
--- /dev/null
+++ b/media/libopus/src/tansig_table.h
@@ -0,0 +1,45 @@
+/* This file is auto-generated by gen_tables */
+
+static const float tansig_table[201] = {
+0.000000f, 0.039979f, 0.079830f, 0.119427f, 0.158649f,
+0.197375f, 0.235496f, 0.272905f, 0.309507f, 0.345214f,
+0.379949f, 0.413644f, 0.446244f, 0.477700f, 0.507977f,
+0.537050f, 0.564900f, 0.591519f, 0.616909f, 0.641077f,
+0.664037f, 0.685809f, 0.706419f, 0.725897f, 0.744277f,
+0.761594f, 0.777888f, 0.793199f, 0.807569f, 0.821040f,
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