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opus_encoder.c
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/* Copyright (c) 2010-2012 IETF Trust, Xiph.Org Foundation, Skype Limited. All rights reserved.
Written by Jean-Marc Valin and Koen Vos */
/*
This file is extracted from RFC6716. Please see that RFC for additional
information.
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.
- Neither the name of Internet Society, IETF or IETF Trust, nor the
names of specific contributors, may be used to endorse or promote
products derived from this software without specific prior written
permission.
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 "tuning_parameters.h"
#ifdef FIXED_POINT
#include "fixed/structs_FIX.h"
#else
#include "float/structs_FLP.h"
#endif
#define MAX_ENCODER_BUFFER 480
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;
opus_int32 bitrate_bps;
opus_int32 user_bitrate_bps;
int encoder_buffer;
#define OPUS_ENCODER_RESET_START stream_channels
int stream_channels;
opus_int16 hybrid_stereo_width_Q14;
opus_int32 variable_HP_smth2_Q15;
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 delay_buffer[MAX_ENCODER_BUFFER*2];
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 */
21000, 2000, /* WB<->SWB */
29000, 2000, /* 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 = 26000;
static const opus_int32 stereo_music_threshold = 36000;
/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
static const opus_int32 mode_thresholds[2][2] = {
/* voice */ /* music */
{ 48000, 24000}, /* mono */
{ 48000, 24000}, /* 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;
/* 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->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, int len, int 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
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./((opus_int32)1<<28)));
A[1] = (opus_val32)(A_Q28[1] * (1./((opus_int32)1<<28)));
B[0] = (opus_val32)(B_Q28[0] * (1./((opus_int32)1<<28)));
B[1] = (opus_val32)(B_Q28[1] * (1./((opus_int32)1<<28)));
B[2] = (opus_val32)(B_Q28[2] * (1./((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
}
static void stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2,
int overlap, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs)
{
int i;
int inc = 48000/Fs;
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;
}
}
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;
}
#ifdef FIXED_POINT
#define opus_encode_native opus_encode
int opus_encode(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
unsigned char *data, int max_data_bytes)
#else
#define opus_encode_native opus_encode_float
int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
unsigned char *data, int max_data_bytes)
#endif
{
void *silk_enc;
CELTEncoder *celt_enc;
int i;
int ret=0;
int nBytes;
ec_enc enc;
int bytes_target;
int prefill=0;
int start_band = 0;
int redundancy = 0;
int redundancy_bytes = 0;
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;
opus_int32 equiv_rate;
int delay_compensation;
int frame_rate;
opus_int32 max_rate;
int curr_bandwidth;
VARDECL(opus_val16, tmp_prefill);
ALLOC_STACK;
max_data_bytes = IMIN(1276, max_data_bytes);
st->rangeFinal = 0;
if (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)
{
RESTORE_STACK;
return OPUS_BAD_ARG;
}
if (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;
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)))
{
int tocmode = st->mode;
if (tocmode==0)
tocmode = MODE_SILK_ONLY;
if (frame_rate>100)
tocmode = MODE_CELT_ONLY;
if (frame_rate < 50)
tocmode = MODE_SILK_ONLY;
data[0] = gen_toc(tocmode, frame_rate,
st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth,
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;
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 -= 4000;
else
stereo_threshold += 4000;
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
int chan;
opus_int32 mode_voice, mode_music;
opus_int32 threshold;
chan = (st->channels==2) && st->force_channels!=1;
mode_voice = mode_thresholds[chan][0];
mode_music = mode_thresholds[chan][1];
threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14);
/* 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;
}
}
}
if (st->silk_bw_switch)
{
redundancy = 1;
celt_to_silk = 1;
st->silk_bw_switch = 0;
}
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;
/* 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;
/* 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;
int bytes_per_frame;
nb_frames = frame_size > st->Fs/25 ? 3 : 2;
bytes_per_frame = max_data_bytes/nb_frames-3;
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);
if (tmp_len<0)
return OPUS_INTERNAL_ERROR;
ret = opus_repacketizer_cat(&rp, tmp_data+i*bytes_per_frame, tmp_len);
if (ret<0)
return OPUS_INTERNAL_ERROR;
}
ret = opus_repacketizer_out(&rp, data, max_data_bytes);
if (ret<0)
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, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1;
data += 1;
ec_enc_init(&enc, data, max_data_bytes-1);
ALLOC(pcm_buf, (delay_compensation+frame_size)*st->channels, opus_val16);
for (i=0;i<delay_compensation*st->channels;i++)
pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-delay_compensation)*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[delay_compensation*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
} else {
for (i=0;i<frame_size*st->channels;i++)
pcm_buf[delay_compensation*st->channels + i] = pcm[i];
}
/* SILK processing */
if (st->mode != MODE_CELT_ONLY)
{
#ifdef FIXED_POINT
const opus_int16 *pcm_silk;
#else
VARDECL(opus_int16, pcm_silk);
ALLOC(pcm_silk, st->channels*frame_size, opus_int16);
#endif
st->silk_mode.bitRate = 8*bytes_target*frame_rate;
if( st->mode == MODE_HYBRID ) {
st->silk_mode.bitRate /= st->stream_channels;
if( curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND ) {
if( st->Fs == 100 * frame_size ) {
/* 24 kHz, 10 ms */
st->silk_mode.bitRate = ( ( st->silk_mode.bitRate + 2000 + st->use_vbr * 1000 ) * 2 ) / 3;
} else {
/* 24 kHz, 20 ms */
st->silk_mode.bitRate = ( ( st->silk_mode.bitRate + 1000 + st->use_vbr * 1000 ) * 2 ) / 3;
}
} else {
if( st->Fs == 100 * frame_size ) {
/* 48 kHz, 10 ms */
st->silk_mode.bitRate = ( st->silk_mode.bitRate + 8000 + st->use_vbr * 3000 ) / 2;
} else {
/* 48 kHz, 20 ms */
st->silk_mode.bitRate = ( st->silk_mode.bitRate + 9000 + st->use_vbr * 1000 ) / 2;
}
}
st->silk_mode.bitRate *= st->stream_channels;
/* don't let SILK use more than 80% */
if( st->silk_mode.bitRate > ( st->bitrate_bps - 8*st->Fs/frame_size ) * 4/5 ) {
st->silk_mode.bitRate = ( st->bitrate_bps - 8*st->Fs/frame_size ) * 4/5;
}
}
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);
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 (redundancy)
st->silk_mode.maxBits -= st->silk_mode.maxBits/(1 + frame_size/(st->Fs/200));
if (prefill)
{
int zero=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+delay_compensation*st->channels;
#else
for (i=0;i<frame_size*st->channels;i++)
pcm_silk[i] = FLOAT2INT16(pcm_buf[delay_compensation*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 */
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;
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: