ref: 888756691836ca8ce419a870a768f910330fb9d1
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 FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <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 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;
/* 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;
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->force_channels = OPUS_AUTO;
st->user_forced_mode = OPUS_AUTO;
st->voice_ratio = -1;
st->encoder_buffer = st->Fs/100;
st->delay_compensation = st->Fs/400;
/* This part is meant to compensate for the resampler delay as a function
of the API sampling rate */
if (st->Fs == 48000)
st->delay_compensation += 23;
else if (st->Fs == 24000)
st->delay_compensation += 15;
else
st->delay_compensation += 2;
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 unsigned char gen_toc(int mode, int framerate, int bandwidth, int silk_bandwidth, int channels)
{
int period;
unsigned char toc;
period = 0;
while (framerate < 400)
{
framerate <<= 1;
period++;
}
if (mode == MODE_SILK_ONLY)
{
toc = (silk_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] = A_Q28[0] * (1./((opus_int32)1<<28));
A[1] = A_Q28[1] * (1./((opus_int32)1<<28));
B[0] = B_Q28[0] * (1./((opus_int32)1<<28));
B[1] = B_Q28[1] * (1./((opus_int32)1<<28));
B[2] = 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 silk_internal_bandwidth=-1;
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;
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;
}
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);
/* 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 = 64;
#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->force_channels!=OPUS_AUTO && st->channels == 2)
{
st->stream_channels = st->force_channels;
} else 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
if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0)
{
/* 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;
}
#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
/* 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)
{
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;
} else {
st->mode = st->user_forced_mode;
}
#endif
/* 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->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->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;
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_rate >= 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->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
}
if (st->user_bandwidth != OPUS_AUTO)
st->bandwidth = st->user_bandwidth;
/* 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;
/* 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);
VARDECL(unsigned char, rp_);
int nb_frames;
int bak_mode, bak_bandwidth, bak_channels;
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);
ALLOC(rp_, opus_repacketizer_get_size(), unsigned char);
rp = opus_repacketizer_init((OpusRepacketizer*)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;
for (i=0;i<nb_frames;i++)
{
int tmp_len;
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);
ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len);
}
ret = opus_repacketizer_out(rp, data, max_data_bytes);
st->user_forced_mode = bak_mode;
st->user_bandwidth = bak_bandwidth;
st->force_channels = bak_channels;
RESTORE_STACK;
return ret;
}
/* 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;
/* 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 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
st->mode = MODE_HYBRID;
if (st->mode == MODE_HYBRID && st->bandwidth <= OPUS_BANDWIDTH_WIDEBAND)
st->mode = MODE_SILK_ONLY;
/* If max_data_bytes represents less than 8 kb/s, switch to CELT-only mode */
if (max_data_bytes < 8000*frame_size / (st->Fs * 8))
st->mode = MODE_CELT_ONLY;
/* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, st->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 = st->bitrate_bps - 8*st->Fs/frame_size;
if( st->mode == MODE_HYBRID ) {
st->silk_mode.bitRate /= st->stream_channels;
if( st->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;
}
}
/* SILK is not allow to use more than 50% of max_data_bytes */
if (max_data_bytes < 2*st->silk_mode.bitRate*frame_size / (st->Fs * 8))
st->silk_mode.bitRate = max_data_bytes*st->Fs*4/frame_size;
st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs;
st->silk_mode.nChannelsAPI = st->channels;
st->silk_mode.nChannelsInternal = st->stream_channels;
if (st->bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
st->silk_mode.desiredInternalSampleRate = 8000;
} else if (st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
st->silk_mode.desiredInternalSampleRate = 12000;
} else {
silk_assert( st->mode == MODE_HYBRID || st->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;
}
st->silk_mode.maxInternalSampleRate = 16000;
/* Call SILK encoder for the low band */
nBytes = IMIN(1275, max_data_bytes-1);
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)
{
data[-1] = gen_toc(st->mode, st->Fs/frame_size, st->bandwidth, silk_internal_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 ) {
silk_internal_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
} else if( st->silk_mode.internalSampleRate == 12000 ) {
silk_internal_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
} else if( st->silk_mode.internalSampleRate == 16000 ) {
silk_internal_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
}
} else {
silk_assert( st->silk_mode.internalSampleRate == 16000 );
}
}
/* CELT processing */
{
int endband=21;
switch(st->bandwidth)
{
case OPUS_BANDWIDTH_NARROWBAND:
endband = 13;
break;
case OPUS_BANDWIDTH_MEDIUMBAND:
case OPUS_BANDWIDTH_WIDEBAND:
endband = 17;
break;
case OPUS_BANDWIDTH_SUPERWIDEBAND:
endband = 19;
break;
case OPUS_BANDWIDTH_FULLBAND:
endband = 21;
break;
}
celt_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband));
celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels));
}
if (st->mode != MODE_SILK_ONLY)
{
celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX));
if (st->prev_mode == MODE_SILK_ONLY)
{
unsigned char dummy[10];
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, &st->delay_buffer[(st->encoder_buffer-delay_compensation-st->Fs/400)*st->channels], st->Fs/400, dummy, 10, NULL);
} else {
celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(2));
}
if (st->mode == MODE_HYBRID)
{
int len;
len = (ec_tell(&enc)+7)>>3;
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)
{
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));
nb_compr_bytes = max_data_bytes-1;
} else {
nb_compr_bytes = bytes_target;
}
}
} else {
nb_compr_bytes = 0;
}
for (i=0;i<st->channels*(st->encoder_buffer-(frame_size+delay_compensation));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+delay_compensation-st->encoder_buffer)*st->channels+i];
if (st->mode != MODE_HYBRID || st->stream_channels==1)
st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14 = 1<<14;
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 = st->silk_mode.stereoWidth_Q14;
#ifdef FIXED_POINT
g1 *= (1./16384);
g2 *= (1./16384);
#else
g1 = g1==16384 ? Q15ONE : SHL16(g1,1);
g2 = g2==16384 ? Q15ONE : SHL16(g2,1);
#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)+29+8*(st->mode == MODE_HYBRID) < 8*nb_compr_bytes)
{
/* Check if we have a redundant 0-8 kHz band */
ec_enc_bit_logp(&enc, redundancy, 12);
if (redundancy)
{
int max_redundancy;
ec_enc_bit_logp(&enc, celt_to_silk, 1);
max_redundancy = nb_compr_bytes-((ec_tell(&enc)+7)>>3)-(st->mode == MODE_HYBRID);
/* 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 (st->mode != MODE_CELT_ONLY)start_band=17;
if (st->mode == MODE_SILK_ONLY)
{
ret = (ec_tell(&enc)+7)>>3;
ec_enc_done(&enc);
/*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.*/
if(!redundancy)
while(ret>2&&data[ret-1]==0)ret--;
nb_compr_bytes = ret;
} else {
nb_compr_bytes = IMIN(1275-redundancy_bytes, nb_compr_bytes);
ec_enc_shrink(&enc, nb_compr_bytes);
}
/* 5 ms redundant frame for CELT->SILK */
if (redundancy && celt_to_silk)
{
celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0));
celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL);
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)
{
ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc);
if (ret < 0)
return OPUS_INTERNAL_ERROR;
}
/* 5 ms redundant frame for SILK->CELT */
if (redundancy && !celt_to_silk)
{
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, data+nb_compr_bytes, redundancy_bytes, NULL);
celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL);
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, st->bandwidth, silk_internal_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;
RESTORE_STACK;
return ret+1+redundancy_bytes;
}
#ifdef FIXED_POINT
#ifndef DISABLE_FLOAT_API
int opus_encode_float(OpusEncoder *st, const float *pcm, int frame_size,
unsigned char *data, int max_data_bytes)
{
int i, ret;
VARDECL(opus_int16, in);
ALLOC_STACK;
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(st, in, frame_size, data, max_data_bytes);
RESTORE_STACK;
return ret;
}
#endif
#else
int opus_encode(OpusEncoder *st, const opus_int16 *pcm, int frame_size,
unsigned char *data, int max_data_bytes)
{
int i, ret;
VARDECL(float, in);
ALLOC_STACK;
ALLOC(in, frame_size*st->channels, float);
for (i=0;i<frame_size*st->channels;i++)
in[i] = (1./32768)*pcm[i];
ret = opus_encode_float(st, in, frame_size, data, max_data_bytes);
RESTORE_STACK;
return ret;
}
#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_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_FINAL_RANGE_REQUEST:
{
opus_uint32 *value = va_arg(ap, opus_uint32*);
*value = st->rangeFinal;
}
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->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;
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);
}