summaryrefslogtreecommitdiffstats
path: root/media/libopus/src/opus_encoder.c
diff options
context:
space:
mode:
Diffstat (limited to 'media/libopus/src/opus_encoder.c')
-rw-r--r--media/libopus/src/opus_encoder.c2536
1 files changed, 2536 insertions, 0 deletions
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);
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-11 22:09:00 +0000