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opus_encoder.c
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/* 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
#ifndef DISABLE_FLOAT_API
#define PSEUDO_SNR_THRESHOLD 316.23f /* 10^(25/10) */
#endif
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;
int use_dtx; /* general DTX for both SILK and CELT */
#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;
/* Bandwidth determined automatically from the rate (before any other adjustment) */
int auto_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;
int nb_no_activity_frames;
opus_val32 peak_signal_energy;
#endif
int nonfinal_frame; /* current frame is not the final in a packet */
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] = {
9000, 700, /* NB<->MB */
9000, 700, /* MB<->WB */
13500, 1000, /* WB<->SWB */
14000, 2000, /* SWB<->FB */
};
static const opus_int32 mono_music_bandwidth_thresholds[8] = {
9000, 700, /* NB<->MB */
9000, 700, /* MB<->WB */
11000, 1000, /* WB<->SWB */
12000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_voice_bandwidth_thresholds[8] = {
9000, 700, /* NB<->MB */
9000, 700, /* MB<->WB */
13500, 1000, /* WB<->SWB */
14000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_music_bandwidth_thresholds[8] = {
9000, 700, /* NB<->MB */
9000, 700, /* MB<->WB */
11000, 1000, /* WB<->SWB */
12000, 2000, /* SWB<->FB */
};
/* Threshold bit-rates for switching between mono and stereo */
static const opus_int32 stereo_voice_threshold = 19000;
static const opus_int32 stereo_music_threshold = 17000;
/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
static const opus_int32 mode_thresholds[2][2] = {
/* voice */ /* music */
{ 64000, 10000}, /* mono */
{ 44000, 10000}, /* stereo */
};
static const opus_int32 fec_thresholds[] = {
12000, 1000, /* NB */
14000, 1000, /* MB */
16000, 1000, /* WB */
20000, 1000, /* SWB */
22000, 1000, /* FB */
};
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, st->Fs);
st->analysis.application = st->application;
#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, int arch)
{
opus_int32 B_Q28[ 3 ], A_Q28[ 2 ];
opus_int32 Fc_Q19, r_Q28, r_Q22;
(void)arch;
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
if( channels == 1 ) {
silk_biquad_alt_stride1( in, B_Q28, A_Q28, hp_mem, out, len );
} else {
silk_biquad_alt_stride2( in, B_Q28, A_Q28, hp_mem, out, len, arch );
}
#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(6.3*cutoff_Hz/Fs)) */
shift=celt_ilog2(Fs/(cutoff_Hz*4));
for (c=0;c<channels;c++)
{
for (i=0;i<len;i++)
{
opus_val32 x, y;
x = SHL32(EXTEND32(in[channels*i+c]), 14);
y = x-hp_mem[2*c];
hp_mem[2*c] = hp_mem[2*c] + PSHR32(x - hp_mem[2*c], shift);
out[channels*i+c] = EXTRACT16(SATURATE(PSHR32(y, 14), 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 i;
float coef, coef2;
coef = 6.3f*cutoff_Hz/Fs;
coef2 = 1-coef;
if (channels==2)
{
float m0, m2;
m0 = hp_mem[0];
m2 = hp_mem[2];
for (i=0;i<len;i++)
{
opus_val32 x0, x1, out0, out1;
x0 = in[2*i+0];
x1 = in[2*i+1];
out0 = x0-m0;
out1 = x1-m2;
m0 = coef*x0 + VERY_SMALL + coef2*m0;
m2 = coef*x1 + VERY_SMALL + coef2*m2;
out[2*i+0] = out0;
out[2*i+1] = out1;
}
hp_mem[0] = m0;
hp_mem[2] = m2;
} else {
float m0;
m0 = hp_mem[0];
for (i=0;i<len;i++)
{
opus_val32 x, y;
x = in[i];
y = x-m0;
m0 = coef*x + VERY_SMALL + coef2*m0;
out[i] = y;
}
hp_mem[0] = m0;
}
}
#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
#ifdef FIXED_POINT
#define PCM2VAL(x) FLOAT2INT16(x)
#else
#define PCM2VAL(x) SCALEIN(x)
#endif
void downmix_float(const void *_x, opus_val32 *y, int subframe, int offset, int c1, int c2, int C)
{
const float *x;
int j;
x = (const float *)_x;
for (j=0;j<subframe;j++)
y[j] = PCM2VAL(x[(j+offset)*C+c1]);
if (c2>-1)
{
for (j=0;j<subframe;j++)
y[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++)
y[j] += PCM2VAL(x[(j+offset)*C+c]);
}
}
}
#endif
void downmix_int(const void *_x, opus_val32 *y, int subframe, int offset, int c1, int c2, int C)
{
const opus_int16 *x;
int j;
x = (const opus_int16 *)_x;
for (j=0;j<subframe;j++)
y[j] = x[(j+offset)*C+c1];
if (c2>-1)
{
for (j=0;j<subframe;j++)
y[j] += x[(j+offset)*C+c2];
} else if (c2==-2)
{
int c;
for (c=1;c<C;c++)
{
for (j=0;j<subframe;j++)
y[j] += x[(j+offset)*C+c];
}
}
}
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_2_5_MS && variable_duration <= OPUS_FRAMESIZE_120_MS)
{
if (variable_duration <= OPUS_FRAMESIZE_40_MS)
new_size = (Fs/400)<<(variable_duration-OPUS_FRAMESIZE_2_5_MS);
else
new_size = (variable_duration-OPUS_FRAMESIZE_2_5_MS-2)*Fs/50;
}
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 &&
50*new_size!=4*Fs && 50*new_size!=5*Fs && 50*new_size!=6*Fs)
return -1;
return new_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);
}
#ifndef FIXED_POINT
if (!(xx < 1e9f) || celt_isnan(xx) || !(yy < 1e9f) || celt_isnan(yy))
{
xy = xx = yy = 0;
}
#endif
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)));
}
static int decide_fec(int useInBandFEC, int PacketLoss_perc, int last_fec, int mode, int *bandwidth, opus_int32 rate)
{
int orig_bandwidth;
if (!useInBandFEC || PacketLoss_perc == 0 || mode == MODE_CELT_ONLY)
return 0;
orig_bandwidth = *bandwidth;
for (;;)
{
opus_int32 hysteresis;
opus_int32 LBRR_rate_thres_bps;
/* Compute threshold for using FEC at the current bandwidth setting */
LBRR_rate_thres_bps = fec_thresholds[2*(*bandwidth - OPUS_BANDWIDTH_NARROWBAND)];
hysteresis = fec_thresholds[2*(*bandwidth - OPUS_BANDWIDTH_NARROWBAND) + 1];
if (last_fec == 1) LBRR_rate_thres_bps -= hysteresis;
if (last_fec == 0) LBRR_rate_thres_bps += hysteresis;
LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps,
125 - silk_min( PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) );
/* If loss <= 5%, we look at whether we have enough rate to enable FEC.
If loss > 5%, we decrease the bandwidth until we can enable FEC. */
if (rate > LBRR_rate_thres_bps)
return 1;
else if (PacketLoss_perc <= 5)
return 0;
else if (*bandwidth > OPUS_BANDWIDTH_NARROWBAND)
(*bandwidth)--;
else
break;
}
/* Couldn't find any bandwidth to enable FEC, keep original bandwidth. */
*bandwidth = orig_bandwidth;
return 0;
}
static int compute_silk_rate_for_hybrid(int rate, int bandwidth, int frame20ms, int vbr, int fec, int channels) {
int entry;
int i;
int N;
int silk_rate;
static int rate_table[][5] = {
/* |total| |-------- SILK------------|
|-- No FEC -| |--- FEC ---|
10ms 20ms 10ms 20ms */
{ 0, 0, 0, 0, 0},
{12000, 10000, 10000, 11000, 11000},
{16000, 13500, 13500, 15000, 15000},
{20000, 16000, 16000, 18000, 18000},
{24000, 18000, 18000, 21000, 21000},
{32000, 22000, 22000, 28000, 28000},
{64000, 38000, 38000, 50000, 50000}
};
/* Do the allocation per-channel. */
rate /= channels;
entry = 1 + frame20ms + 2*fec;
N = sizeof(rate_table)/sizeof(rate_table[0]);
for (i=1;i<N;i++)
{
if (rate_table[i][0] > rate) break;
}
if (i == N)
{
silk_rate = rate_table[i-1][entry];
/* For now, just give 50% of the extra bits to SILK. */
silk_rate += (rate-rate_table[i-1][0])/2;
} else {
opus_int32 lo, hi, x0, x1;
lo = rate_table[i-1][entry];
hi = rate_table[i][entry];
x0 = rate_table[i-1][0];
x1 = rate_table[i][0];
silk_rate = (lo*(x1-rate) + hi*(rate-x0))/(x1-x0);
}
if (!vbr)
{
/* Tiny boost to SILK for CBR. We should probably tune this better. */
silk_rate += 100;
}
if (bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND)
silk_rate += 300;
silk_rate *= channels;
/* Small adjustment for stereo (calibrated for 32 kb/s, haven't tried other bitrates). */
if (channels == 2 && rate >= 12000)
silk_rate -= 1000;
return silk_rate;
}
/* Returns the equivalent bitrate corresponding to 20 ms frames,
complexity 10 VBR operation. */
static opus_int32 compute_equiv_rate(opus_int32 bitrate, int channels,
int frame_rate, int vbr, int mode, int complexity, int loss)
{
opus_int32 equiv;
equiv = bitrate;
/* Take into account overhead from smaller frames. */
if (frame_rate > 50)
equiv -= (40*channels+20)*(frame_rate - 50);
/* CBR is about a 8% penalty for both SILK and CELT. */
if (!vbr)
equiv -= equiv/12;
/* Complexity makes about 10% difference (from 0 to 10) in general. */
equiv = equiv * (90+complexity)/100;
if (mode == MODE_SILK_ONLY || mode == MODE_HYBRID)
{
/* SILK complexity 0-1 uses the non-delayed-decision NSQ, which
costs about 20%. */
if (complexity<2)
equiv = equiv*4/5;
equiv -= equiv*loss/(6*loss + 10);
} else if (mode == MODE_CELT_ONLY) {
/* CELT complexity 0-4 doesn't have the pitch filter, which costs
about 10%. */
if (complexity<5)
equiv = equiv*9/10;
} else {
/* Mode not known yet */
/* Half the SILK loss*/
equiv -= equiv*loss/(12*loss + 20);
}
return equiv;
}
#ifndef DISABLE_FLOAT_API
int is_digital_silence(const opus_val16* pcm, int frame_size, int channels, int lsb_depth)
{
int silence = 0;
opus_val32 sample_max = 0;
#ifdef MLP_TRAINING
return 0;
#endif
sample_max = celt_maxabs16(pcm, frame_size*channels);
#ifdef FIXED_POINT
silence = (sample_max == 0);
(void)lsb_depth;
#else
silence = (sample_max <= (opus_val16) 1 / (1 << lsb_depth));
#endif
return silence;
}
#ifdef FIXED_POINT
static opus_val32 compute_frame_energy(const opus_val16 *pcm, int frame_size, int channels, int arch)
{
int i;
opus_val32 sample_max;
int max_shift;
int shift;
opus_val32 energy = 0;
int len = frame_size*channels;
(void)arch;
/* Max amplitude in the signal */
sample_max = celt_maxabs16(pcm, len);
/* Compute the right shift required in the MAC to avoid an overflow */
max_shift = celt_ilog2(len);
shift = IMAX(0, (celt_ilog2(sample_max) << 1) + max_shift - 28);
/* Compute the energy */
for (i=0; i<len; i++)
energy += SHR32(MULT16_16(pcm[i], pcm[i]), shift);
/* Normalize energy by the frame size and left-shift back to the original position */
energy /= len;
energy = SHL32(energy, shift);
return energy;
}
#else
static opus_val32 compute_frame_energy(const opus_val16 *pcm, int frame_size, int channels, int arch)
{
int len = frame_size*channels;
return celt_inner_prod(pcm, pcm, len, arch)/len;
}
#endif
/* Decides if DTX should be turned on (=1) or off (=0) */
static int decide_dtx_mode(opus_int activity, /* indicates if this frame contains speech/music */
int *nb_no_activity_frames /* number of consecutive frames with no activity */
)
{
if (!activity)
{
/* The number of consecutive DTX frames should be within the allowed bounds */
(*nb_no_activity_frames)++;
if (*nb_no_activity_frames > NB_SPEECH_FRAMES_BEFORE_DTX)
{
if (*nb_no_activity_frames <= (NB_SPEECH_FRAMES_BEFORE_DTX + MAX_CONSECUTIVE_DTX))
/* Valid frame for DTX! */
return 1;
else
(*nb_no_activity_frames) = NB_SPEECH_FRAMES_BEFORE_DTX;
}
} else
(*nb_no_activity_frames) = 0;
return 0;
}
#endif
static opus_int32 encode_multiframe_packet(OpusEncoder *st,
const opus_val16 *pcm,
int nb_frames,
int frame_size,
unsigned char *data,
opus_int32 out_data_bytes,
int to_celt,
int lsb_depth,
int float_api)
{
int i;
int ret = 0;
VARDECL(unsigned char, tmp_data);
int bak_mode, bak_bandwidth, bak_channels, bak_to_mono;
VARDECL(OpusRepacketizer, rp);
int max_header_bytes;
opus_int32 bytes_per_frame;
opus_int32 cbr_bytes;
opus_int32 repacketize_len;
int tmp_len;
ALLOC_STACK;
/* Worst cases:
* 2 frames: Code 2 with different compressed sizes
* >2 frames: Code 3 VBR */
max_header_bytes = nb_frames == 2 ? 3 : (2+(nb_frames-1)*2);
if (st->use_vbr || st->user_bitrate_bps==OPUS_BITRATE_MAX)
repacketize_len = out_data_bytes;
else {
cbr_bytes = 3*st->bitrate_bps/(3*8*st->Fs/(frame_size*nb_frames));
repacketize_len = IMIN(cbr_bytes, out_data_bytes);
}
bytes_per_frame = IMIN(1276, 1+(repacketize_len-max_header_bytes)/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++)
{
st->silk_mode.toMono = 0;
st->nonfinal_frame = i<(nb_frames-1);
/* 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*frame_size), frame_size,
tmp_data+i*bytes_per_frame, bytes_per_frame, lsb_depth, NULL, 0, 0, 0, 0,
NULL, 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;
}
}