ref: 172f66a3ac321d1c39c4341944983d3de723d199
dir: /src/opus_encoder.c/
/* 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 "opus_private.h"
#include "os_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;
#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;
StereoWidthState width_mem;
opus_val16 delay_buffer[MAX_ENCODER_BUFFER*2];
#ifndef FIXED_POINT
TonalityAnalysisState analysis;
int detected_bandwidth;
int analysis_offset;
#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 */
20000, 1000, /* SWB<->FB */
};
static const opus_int32 mono_music_bandwidth_thresholds[8] = {
14000, 1000, /* MB not allowed */
18000, 2000, /* MB<->WB */
24000, 2000, /* WB<->SWB */
33000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_voice_bandwidth_thresholds[8] = {
11000, 1000, /* NB<->MB */
14000, 1000, /* MB<->WB */
21000, 2000, /* WB<->SWB */
32000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_music_bandwidth_thresholds[8] = {
14000, 1000, /* MB not allowed */
18000, 2000, /* MB<->WB */
24000, 2000, /* WB<->SWB */
48000, 2000, /* SWB<->FB */
};
/* Threshold bit-rates for switching between mono and stereo */
static const opus_int32 stereo_voice_threshold = 31000;
static const opus_int32 stereo_music_threshold = 31000;
/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
static const opus_int32 mode_thresholds[2][2] = {
/* voice */ /* music */
{ 64000, 20000}, /* mono */
{ 36000, 20000}, /* 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;
ret = silk_InitEncoder( silk_enc, &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 = 10;
st->silk_mode.useInBandFEC = 0;
st->silk_mode.useDTX = 0;
st->silk_mode.useCBR = 0;
/* Create CELT encoder */
/* Initialize CELT encoder */
err = celt_encoder_init(celt_enc, Fs, channels);
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(10));
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;
return OPUS_OK;
}
static int pad_frame(unsigned char *data, opus_int32 len, opus_int32 new_len)
{
if (len == new_len)
return 0;
if (len > new_len)
return 1;
if ((data[0]&0x3)==0)
{
int i;
int padding, nb_255s;
padding = new_len - len;
if (padding >= 2)
{
nb_255s = (padding-2)/255;
for (i=len-1;i>=1;i--)
data[i+nb_255s+2] = data[i];
data[0] |= 0x3;
data[1] = 0x41;
for (i=0;i<nb_255s;i++)
data[i+2] = 255;
data[nb_255s+2] = padding-255*nb_255s-2;
for (i=len+3+nb_255s;i<new_len;i++)
data[i] = 0;
} else {
for (i=len-1;i>=1;i--)
data[i+1] = data[i];
data[0] |= 0x3;
data[1] = 1;
}
return 0;
} else {
return 1;
}
}
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;
/* 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]);
/* 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]);
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 FIXED_POINT
/* 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,sqrt(MAX16(0,.05*(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;
}
void downmix_float(const void *_x, float *sub, int subframe, int offset, int C)
{
const float *x;
int c, j;
x = (const float *)_x;
for (j=0;j<subframe;j++)
sub[j] = x[(j+offset)*C];
for (c=1;c<C;c++)
for (j=0;j<subframe;j++)
sub[j] += x[(j+offset)*C+c];
}
void downmix_int(const void *_x, float *sub, int subframe, int offset, int C)
{
const opus_int16 *x;
int c, j;
x = (const opus_int16 *)_x;
for (j=0;j<subframe;j++)
sub[j] = x[(j+offset)*C];
for (c=1;c<C;c++)
for (j=0;j<subframe;j++)
sub[j] += x[(j+offset)*C+c];
for (j=0;j<subframe;j++)
sub[j] *= (1.f/32768);
}
int optimize_framesize(const opus_val16 *x, int len, int C, opus_int32 Fs,
int bitrate, opus_val16 tonality, opus_val32 *mem, int buffering,
downmix_func downmix)
{
int N;
int i;
float e[MAX_DYNAMIC_FRAMESIZE+4];
float e_1[MAX_DYNAMIC_FRAMESIZE+3];
float memx;
int bestLM=0;
int subframe;
int pos;
VARDECL(opus_val16, sub);
subframe = Fs/400;
ALLOC(sub, subframe, opus_val16);
e[0]=mem[0];
e_1[0]=1./(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 */
int offset = 2*subframe - buffering;
celt_assert(offset>=0 && offset <= subframe);
x += C*offset;
len -= offset;
e[1]=mem[1];
e_1[1]=1./(EPSILON+mem[1]);
e[2]=mem[2];
e_1[2]=1./(EPSILON+mem[2]);
pos = 3;
} else {
pos=1;
}
N=IMIN(len/subframe, MAX_DYNAMIC_FRAMESIZE);
memx = x[0];
for (i=0;i<N;i++)
{
float tmp;
float tmpx;
int j;
tmp=EPSILON;
downmix(x, sub, subframe, i*subframe, C);
if (i==0)
memx = sub[0];
for (j=0;j<subframe;j++)
{
tmpx = sub[j];
tmp += (tmpx-memx)*(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, (1.f+.5*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
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_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int32 Fs, StereoWidthState *mem)
{
opus_val16 corr;
opus_val16 ldiff;
opus_val16 width;
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 - 25*Q15ONE/IMAX(50,frame_rate);
xx=xy=yy=0;
for (i=0;i<frame_size;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))
{
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 = 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);
} else {
width = 0;
corr=Q15ONE;
ldiff=0;
}
/*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,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
#ifndef FIXED_POINT
, AnalysisInfo *analysis_info
#endif
)
{
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;
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);
st->voice_ratio = -1;
#ifndef FIXED_POINT
st->detected_bandwidth = 0;
if (analysis_info->valid)
{
if (st->signal_type == OPUS_AUTO)
st->voice_ratio = (int)floor(.5+100*(1-analysis_info->music_prob));
st->detected_bandwidth = analysis_info->opus_bandwidth;
}
#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 (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 (bw<=OPUS_BANDWIDTH_SUPERWIDEBAND)
bw=OPUS_BANDWIDTH_SUPERWIDEBAND;
data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels);
RESTORE_STACK;
return 1;
}
if (!st->use_vbr)
{
int cbrBytes;
cbrBytes = IMIN( (st->bitrate_bps + 4*frame_rate)/(8*frame_rate) , max_data_bytes);
st->bitrate_bps = cbrBytes * (8*frame_rate);
max_data_bytes = cbrBytes;
}
max_rate = frame_rate*max_data_bytes*8;
/* Equivalent 20-ms rate for mode/channel/bandwidth decisions */
equiv_rate = st->bitrate_bps - 60*(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
}
/* 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 = MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[0][0]) + MULT16_32_Q15(stereo_width,mode_thresholds[1][0]);
mode_music = 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->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, &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 FIXED_POINT
/* Use detected bandwidth to reduce the encoded bandwidth. */
if (st->detected_bandwidth && st->user_bandwidth == OPUS_AUTO)
{
/* When operating in SILK/hybrid mode, we don't go below wideband to avoid
more complicated switches that require redundancy */
if (st->mode == MODE_CELT_ONLY)
st->bandwidth = IMIN(st->bandwidth, st->detected_bandwidth);
else
st->bandwidth = IMIN(st->bandwidth, IMAX(OPUS_BANDWIDTH_WIDEBAND, st->detected_bandwidth));
}
#endif
celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(lsb_depth));
/* 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;
/* 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;
st->mode = MODE_CELT_ONLY;
}
/* 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;
OpusRepacketizer rp;
opus_int32 bytes_per_frame;
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);
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
#ifndef FIXED_POINT
, analysis_info
#endif
);
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;
}
}
ret = opus_repacketizer_out(&rp, data, out_data_bytes);
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);
for (i=0;i<total_buffer*st->channels;i++)
pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-total_buffer)*st->channels+i];
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);
}
/* 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;
}
/* 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 < Q15ONE*6/7 ? HB_gain + Q15ONE/7 : Q15ONE;
} else {
/* SILK gets all bits */
st->silk_mode.bitRate = total_bitRate;
}
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;
const CELTMode *celt_mode;
int prefill_offset;
celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode));
/* 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);
for(i=0;i<prefill_offset;i++)
st->delay_buffer[i]=0;
#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)
{
celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
/* Allow prediction unless we decide to disable it later */
celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(2));
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 FIXED_POINT
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 = bonus*(1.f+.5*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)
{
for (i=0;i<st->channels*st->Fs/400;i++)
tmp_prefill[i] = st->delay_buffer[(st->encoder_buffer-total_buffer-st->Fs/400)*st->channels + i];
}
for (i=0;i<st->channels*(st->encoder_buffer-(frame_size+total_buffer));i++)
st->delay_buffer[i] = st->delay_buffer[i+st->channels*frame_size];
for (;i<st->encoder_buffer*st->channels;i++)
st->delay_buffer[i] = pcm_buf[(frame_size+total_buffer-st->encoder_buffer)*st->channels+i];
/* 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 ) {
const CELTMode *celt_mode;
celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode));
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),IMAX(0,st->bitrate_bps-32000));
if( 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;
const CELTMode *celt_mode;
celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode));
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-1;
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);
}
/* 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)
{
#ifndef FIXED_POINT
celt_encoder_ctl(celt_enc, CELT_SET_ANALYSIS(analysis_info));
#endif
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 && ret >= 3)
{
if (pad_frame(data, ret, max_data_bytes))
{
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 frame_size,
unsigned char *data, opus_int32 max_data_bytes)
{
int i, ret;
VARDECL(opus_int16, in);
ALLOC_STACK;
frame_size = frame_size_select(frame_size, st->variable_duration, st->Fs);
if(frame_size<0)
{
RESTORE_STACK;
return OPUS_BAD_ARG;
}
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);
RESTORE_STACK;
return ret;
}
#endif
opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int frame_size,
unsigned char *data, opus_int32 out_data_bytes)
{
frame_size = frame_size_select(frame_size, st->variable_duration, st->Fs);
if(frame_size<0)
{
return OPUS_BAD_ARG;
}
return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 16);
}
#else
opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int frame_size,
unsigned char *data, opus_int32 max_data_bytes)
{
int i, ret;
const CELTMode *celt_mode;
int delay_compensation;
int lsb_depth;
VARDECL(float, in);
AnalysisInfo analysis_info;
ALLOC_STACK;
opus_encoder_ctl(st, CELT_GET_MODE(&celt_mode));
if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
delay_compensation = 0;
else
delay_compensation = st->delay_compensation;
lsb_depth = IMIN(16, st->lsb_depth);
analysis_info.valid = 0;
if (st->silk_mode.complexity >= 7 && st->Fs==48000)
{
frame_size = run_analysis(&st->analysis, celt_mode, pcm, pcm+st->channels*st->analysis.analysis_offset,
frame_size, st->variable_duration, st->channels, st->Fs, st->bitrate_bps, delay_compensation, lsb_depth, downmix_int, &analysis_info);
} else {
frame_size = frame_size_select(frame_size, st->variable_duration, st->Fs);
}
if(frame_size<0)
{
RESTORE_STACK;
return OPUS_BAD_ARG;
}
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, &analysis_info);
RESTORE_STACK;
return ret;
}
opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int frame_size,
unsigned char *data, opus_int32 out_data_bytes)
{
const CELTMode *celt_mode;
int delay_compensation;
int lsb_depth;
AnalysisInfo analysis_info;
opus_encoder_ctl(st, CELT_GET_MODE(&celt_mode));
if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
delay_compensation = 0;
else
delay_compensation = st->delay_compensation;
lsb_depth = IMIN(24, st->lsb_depth);
analysis_info.valid = 0;
if (st->silk_mode.complexity >= 7 && st->Fs==48000)
{
frame_size = run_analysis(&st->analysis, celt_mode, pcm, pcm+st->channels*st->analysis.analysis_offset,
frame_size, st->variable_duration, st->channels, st->Fs, st->bitrate_bps, delay_compensation, lsb_depth, downmix_float, &analysis_info);
} else {
frame_size = frame_size_select(frame_size, st->variable_duration, st->Fs);
}
if(frame_size<0)
{
return OPUS_BAD_ARG;
}
return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 24, &analysis_info);
}
#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*);
*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*);
*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)
return OPUS_BAD_ARG;
st->force_channels = value;
}
break;
case OPUS_GET_FORCE_CHANNELS_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*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)
return OPUS_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*);
*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)
return OPUS_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*);
*value = st->bandwidth;
}
break;
case OPUS_SET_DTX_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if(value<0 || value>1)
return OPUS_BAD_ARG;
st->silk_mode.useDTX = value;
}
break;
case OPUS_GET_DTX_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*value = st->silk_mode.useDTX;
}
break;
case OPUS_SET_COMPLEXITY_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if(value<0 || value>10)
return OPUS_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*);
*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)
return OPUS_BAD_ARG;
st->silk_mode.useInBandFEC = value;
}
break;
case OPUS_GET_INBAND_FEC_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*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)
return OPUS_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*);
*value = st->silk_mode.packetLossPercentage;
}
break;
case OPUS_SET_VBR_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if(value<0 || value>1)
return OPUS_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*);
*value = st->use_vbr;
}
break;
case OPUS_SET_VOICE_RATIO_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if (value>100 || value<-1)
goto bad_arg;
st->voice_ratio = value;
}
break;
case OPUS_GET_VOICE_RATIO_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*value = st->voice_ratio;
}
break;
case OPUS_SET_VBR_CONSTRAINT_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if(value<0 || value>1)
return OPUS_BAD_ARG;
st->vbr_constraint = value;
}
break;
case OPUS_GET_VBR_CONSTRAINT_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*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)
return OPUS_BAD_ARG;
st->signal_type = value;
}
break;
case OPUS_GET_SIGNAL_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*value = st->signal_type;
}
break;
case OPUS_GET_LOOKAHEAD_REQUEST:
{
opus_int32 *value = va_arg(ap, opus_int32*);
*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==NULL)
{
ret = OPUS_BAD_ARG;
break;
}
*value = st->Fs;
}
break;
case OPUS_GET_FINAL_RANGE_REQUEST:
{
opus_uint32 *value = va_arg(ap, opus_uint32*);
*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*);
*value = st->lsb_depth;
}
break;
case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
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*);
*value = st->variable_duration;
}
break;
case OPUS_RESET_STATE:
{
void *silk_enc;
silk_EncControlStruct dummy;
silk_enc = (char*)st+st->silk_enc_offset;
OPUS_CLEAR((char*)&st->OPUS_ENCODER_RESET_START,
sizeof(OpusEncoder)-
((char*)&st->OPUS_ENCODER_RESET_START - (char*)st));
celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
silk_InitEncoder( silk_enc, &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;
celt_encoder_ctl(celt_enc, OPUS_SET_LFE(value));
}
break;
case CELT_GET_MODE_REQUEST:
{
const CELTMode ** value = va_arg(ap, const CELTMode**);
if (value==0)
goto bad_arg;
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);
}