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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /media/libopus/src/opus_encoder.c | |
parent | Initial commit. (diff) | |
download | firefox-upstream.tar.xz firefox-upstream.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'media/libopus/src/opus_encoder.c')
-rw-r--r-- | media/libopus/src/opus_encoder.c | 2780 |
1 files changed, 2780 insertions, 0 deletions
diff --git a/media/libopus/src/opus_encoder.c b/media/libopus/src/opus_encoder.c new file mode 100644 index 0000000000..8c8db5a546 --- /dev/null +++ b/media/libopus/src/opus_encoder.c @@ -0,0 +1,2780 @@ +/* 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 */ + int fec_config; +#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_ms_Q1; + 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_ms_Q1, /* number of consecutive milliseconds with no activity, in Q1 */ + int frame_size_ms_Q1 /* number of miliseconds in this update, in Q1 */ + ) + +{ + if (!activity) + { + /* The number of consecutive DTX frames should be within the allowed bounds. + Note that the allowed bound is defined in the SILK headers and assumes 20 ms + frames. As this function can be called with any frame length, a conversion to + milliseconds is done before the comparisons. */ + (*nb_no_activity_ms_Q1) += frame_size_ms_Q1; + if (*nb_no_activity_ms_Q1 > NB_SPEECH_FRAMES_BEFORE_DTX*20*2) + { + if (*nb_no_activity_ms_Q1 <= (NB_SPEECH_FRAMES_BEFORE_DTX + MAX_CONSECUTIVE_DTX)*20*2) + /* Valid frame for DTX! */ + return 1; + else + (*nb_no_activity_ms_Q1) = NB_SPEECH_FRAMES_BEFORE_DTX*20*2; + } + } else + (*nb_no_activity_ms_Q1) = 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; + } + } + + ret = opus_repacketizer_out_range_impl(rp, 0, nb_frames, data, repacketize_len, 0, !st->use_vbr); + + if (ret<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + + /* Discard configs that were forced locally for the purpose of repacketization */ + 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; +} + +static int compute_redundancy_bytes(opus_int32 max_data_bytes, opus_int32 bitrate_bps, int frame_rate, int channels) +{ + int redundancy_bytes_cap; + int redundancy_bytes; + opus_int32 redundancy_rate; + int base_bits; + opus_int32 available_bits; + base_bits = (40*channels+20); + + /* Equivalent rate for 5 ms frames. */ + redundancy_rate = bitrate_bps + base_bits*(200 - frame_rate); + /* For VBR, further increase the bitrate if we can afford it. It's pretty short + and we'll avoid artefacts. */ + redundancy_rate = 3*redundancy_rate/2; + redundancy_bytes = redundancy_rate/1600; + + /* Compute the max rate we can use given CBR or VBR with cap. */ + available_bits = max_data_bytes*8 - 2*base_bits; + redundancy_bytes_cap = (available_bits*240/(240+48000/frame_rate) + base_bits)/8; + redundancy_bytes = IMIN(redundancy_bytes, redundancy_bytes_cap); + /* It we can't get enough bits for redundancy to be worth it, rely on the decoder PLC. */ + if (redundancy_bytes > 4 + 8*channels) + redundancy_bytes = IMIN(257, redundancy_bytes); + else + redundancy_bytes = 0; + return redundancy_bytes; +} + +opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size, + unsigned char *data, opus_int32 out_data_bytes, int lsb_depth, + const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2, + int analysis_channels, downmix_func downmix, int float_api) +{ + void *silk_enc; + CELTEncoder *celt_enc; + int i; + int ret=0; + opus_int32 nBytes; + ec_enc enc; + int bytes_target; + int prefill=0; + int start_band = 0; + int redundancy = 0; + int redundancy_bytes = 0; /* Number of bytes to use for redundancy frame */ + int celt_to_silk = 0; + VARDECL(opus_val16, pcm_buf); + int nb_compr_bytes; + int to_celt = 0; + opus_uint32 redundant_rng = 0; + int cutoff_Hz, hp_freq_smth1; + int voice_est; /* Probability of voice in Q7 */ + opus_int32 equiv_rate; + int delay_compensation; + int frame_rate; + opus_int32 max_rate; /* Max bitrate we're allowed to use */ + int curr_bandwidth; + opus_val16 HB_gain; + opus_int32 max_data_bytes; /* Max number of bytes we're allowed to use */ + int total_buffer; + opus_val16 stereo_width; + const CELTMode *celt_mode; +#ifndef DISABLE_FLOAT_API + AnalysisInfo analysis_info; + int analysis_read_pos_bak=-1; + int analysis_read_subframe_bak=-1; + int is_silence = 0; +#endif + opus_int activity = VAD_NO_DECISION; + + VARDECL(opus_val16, tmp_prefill); + + ALLOC_STACK; + + max_data_bytes = IMIN(1276, out_data_bytes); + + st->rangeFinal = 0; + if (frame_size <= 0 || max_data_bytes <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + /* Cannot encode 100 ms in 1 byte */ + if (max_data_bytes==1 && st->Fs==(frame_size*10)) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + + silk_enc = (char*)st+st->silk_enc_offset; + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY) + delay_compensation = 0; + else + delay_compensation = st->delay_compensation; + + lsb_depth = IMIN(lsb_depth, st->lsb_depth); + + celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode)); +#ifndef DISABLE_FLOAT_API + analysis_info.valid = 0; +#ifdef FIXED_POINT + if (st->silk_mode.complexity >= 10 && st->Fs>=16000) +#else + if (st->silk_mode.complexity >= 7 && st->Fs>=16000) +#endif + { + is_silence = is_digital_silence(pcm, frame_size, st->channels, lsb_depth); + analysis_read_pos_bak = st->analysis.read_pos; + analysis_read_subframe_bak = st->analysis.read_subframe; + run_analysis(&st->analysis, celt_mode, analysis_pcm, analysis_size, frame_size, + c1, c2, analysis_channels, st->Fs, + lsb_depth, downmix, &analysis_info); + + /* Track the peak signal energy */ + if (!is_silence && analysis_info.activity_probability > DTX_ACTIVITY_THRESHOLD) + st->peak_signal_energy = MAX32(MULT16_32_Q15(QCONST16(0.999f, 15), st->peak_signal_energy), + compute_frame_energy(pcm, frame_size, st->channels, st->arch)); + } else if (st->analysis.initialized) { + tonality_analysis_reset(&st->analysis); + } +#else + (void)analysis_pcm; + (void)analysis_size; + (void)c1; + (void)c2; + (void)analysis_channels; + (void)downmix; +#endif + +#ifndef DISABLE_FLOAT_API + /* Reset voice_ratio if this frame is not silent or if analysis is disabled. + * Otherwise, preserve voice_ratio from the last non-silent frame */ + if (!is_silence) + st->voice_ratio = -1; + + if (is_silence) + { + activity = !is_silence; + } else if (analysis_info.valid) + { + activity = analysis_info.activity_probability >= DTX_ACTIVITY_THRESHOLD; + if (!activity) + { + /* Mark as active if this noise frame is sufficiently loud */ + opus_val32 noise_energy = compute_frame_energy(pcm, frame_size, st->channels, st->arch); + activity = st->peak_signal_energy < (PSEUDO_SNR_THRESHOLD * noise_energy); + } + } + + st->detected_bandwidth = 0; + if (analysis_info.valid) + { + int analysis_bandwidth; + if (st->signal_type == OPUS_AUTO) + { + float prob; + if (st->prev_mode == 0) + prob = analysis_info.music_prob; + else if (st->prev_mode == MODE_CELT_ONLY) + prob = analysis_info.music_prob_max; + else + prob = analysis_info.music_prob_min; + st->voice_ratio = (int)floor(.5+100*(1-prob)); + } + + analysis_bandwidth = analysis_info.bandwidth; + if (analysis_bandwidth<=12) + st->detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + else if (analysis_bandwidth<=14) + st->detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + else if (analysis_bandwidth<=16) + st->detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + else if (analysis_bandwidth<=18) + st->detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + else + st->detected_bandwidth = OPUS_BANDWIDTH_FULLBAND; + } +#else + st->voice_ratio = -1; +#endif + + if (st->channels==2 && st->force_channels!=1) + stereo_width = compute_stereo_width(pcm, frame_size, st->Fs, &st->width_mem); + else + stereo_width = 0; + total_buffer = delay_compensation; + st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes); + + frame_rate = st->Fs/frame_size; + if (!st->use_vbr) + { + int cbrBytes; + /* Multiply by 12 to make sure the division is exact. */ + int frame_rate12 = 12*st->Fs/frame_size; + /* We need to make sure that "int" values always fit in 16 bits. */ + cbrBytes = IMIN( (12*st->bitrate_bps/8 + frame_rate12/2)/frame_rate12, max_data_bytes); + st->bitrate_bps = cbrBytes*(opus_int32)frame_rate12*8/12; + /* Make sure we provide at least one byte to avoid failing. */ + max_data_bytes = IMAX(1, cbrBytes); + } + if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8 + || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400))) + { + /*If the space is too low to do something useful, emit 'PLC' frames.*/ + int tocmode = st->mode; + int bw = st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth; + int packet_code = 0; + int num_multiframes = 0; + + if (tocmode==0) + tocmode = MODE_SILK_ONLY; + if (frame_rate>100) + tocmode = MODE_CELT_ONLY; + /* 40 ms -> 2 x 20 ms if in CELT_ONLY or HYBRID mode */ + if (frame_rate==25 && tocmode!=MODE_SILK_ONLY) + { + frame_rate = 50; + packet_code = 1; + } + + /* >= 60 ms frames */ + if (frame_rate<=16) + { + /* 1 x 60 ms, 2 x 40 ms, 2 x 60 ms */ + if (out_data_bytes==1 || (tocmode==MODE_SILK_ONLY && frame_rate!=10)) + { + tocmode = MODE_SILK_ONLY; + + packet_code = frame_rate <= 12; + frame_rate = frame_rate == 12 ? 25 : 16; + } + else + { + num_multiframes = 50/frame_rate; + frame_rate = 50; + packet_code = 3; + } + } + + if(tocmode==MODE_SILK_ONLY&&bw>OPUS_BANDWIDTH_WIDEBAND) + bw=OPUS_BANDWIDTH_WIDEBAND; + else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND) + bw=OPUS_BANDWIDTH_NARROWBAND; + else if (tocmode==MODE_HYBRID&&bw<=OPUS_BANDWIDTH_SUPERWIDEBAND) + bw=OPUS_BANDWIDTH_SUPERWIDEBAND; + + data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels); + data[0] |= packet_code; + + ret = packet_code <= 1 ? 1 : 2; + + max_data_bytes = IMAX(max_data_bytes, ret); + + if (packet_code==3) + data[1] = num_multiframes; + + if (!st->use_vbr) + { + ret = opus_packet_pad(data, ret, max_data_bytes); + if (ret == OPUS_OK) + ret = max_data_bytes; + else + ret = OPUS_INTERNAL_ERROR; + } + RESTORE_STACK; + return ret; + } + max_rate = frame_rate*max_data_bytes*8; + + /* Equivalent 20-ms rate for mode/channel/bandwidth decisions */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->channels, st->Fs/frame_size, + st->use_vbr, 0, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + if (st->signal_type == OPUS_SIGNAL_VOICE) + voice_est = 127; + else if (st->signal_type == OPUS_SIGNAL_MUSIC) + voice_est = 0; + else if (st->voice_ratio >= 0) + { + voice_est = st->voice_ratio*327>>8; + /* For AUDIO, never be more than 90% confident of having speech */ + if (st->application == OPUS_APPLICATION_AUDIO) + voice_est = IMIN(voice_est, 115); + } else if (st->application == OPUS_APPLICATION_VOIP) + voice_est = 115; + else + voice_est = 48; + + if (st->force_channels!=OPUS_AUTO && st->channels == 2) + { + st->stream_channels = st->force_channels; + } else { +#ifdef FUZZING + (void)stereo_music_threshold; + (void)stereo_voice_threshold; + /* Random mono/stereo decision */ + if (st->channels == 2 && (rand()&0x1F)==0) + st->stream_channels = 3-st->stream_channels; +#else + /* Rate-dependent mono-stereo decision */ + if (st->channels == 2) + { + opus_int32 stereo_threshold; + stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14); + if (st->stream_channels == 2) + stereo_threshold -= 1000; + else + stereo_threshold += 1000; + st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1; + } else { + st->stream_channels = st->channels; + } +#endif + } + /* Update equivalent rate for channels decision. */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->stream_channels, st->Fs/frame_size, + st->use_vbr, 0, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + /* Allow SILK DTX if DTX is enabled but the generalized DTX cannot be used, + e.g. because of the complexity setting or sample rate. */ +#ifndef DISABLE_FLOAT_API + st->silk_mode.useDTX = st->use_dtx && !(analysis_info.valid || is_silence); +#else + st->silk_mode.useDTX = st->use_dtx; +#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 + (void)stereo_width; + (void)mode_thresholds; + /* Random mode switching */ + if ((rand()&0xF)==0) + { + if ((rand()&0x1)==0) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } else { + if (st->prev_mode==MODE_CELT_ONLY) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } +#else + opus_int32 mode_voice, mode_music; + opus_int32 threshold; + + /* Interpolate based on stereo width */ + mode_voice = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[0][0]) + + MULT16_32_Q15(stereo_width,mode_thresholds[1][0])); + mode_music = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[1][1]) + + MULT16_32_Q15(stereo_width,mode_thresholds[1][1])); + /* Interpolate based on speech/music probability */ + threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14); + /* Bias towards SILK for VoIP because of some useful features */ + if (st->application == OPUS_APPLICATION_VOIP) + threshold += 8000; + + /*printf("%f %d\n", stereo_width/(float)Q15ONE, threshold);*/ + /* Hysteresis */ + if (st->prev_mode == MODE_CELT_ONLY) + threshold -= 4000; + else if (st->prev_mode>0) + threshold += 4000; + + st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY; + + /* When FEC is enabled and there's enough packet loss, use SILK. + Unless the FEC is set to 2, in which case we don't switch to SILK if we're confident we have music. */ + if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4 && (st->fec_config != 2 || voice_est > 25)) + st->mode = MODE_SILK_ONLY; + /* When encoding voice and DTX is enabled but the generalized DTX cannot be used, + use SILK in order to make use of its DTX. */ + if (st->silk_mode.useDTX && voice_est > 100) + st->mode = MODE_SILK_ONLY; +#endif + + /* If max_data_bytes represents less than 6 kb/s, switch to CELT-only mode */ + if (max_data_bytes < (frame_rate > 50 ? 9000 : 6000)*frame_size / (st->Fs * 8)) + st->mode = MODE_CELT_ONLY; + } else { + st->mode = st->user_forced_mode; + } + + /* Override the chosen mode to make sure we meet the requested frame size */ + if (st->mode != MODE_CELT_ONLY && frame_size < st->Fs/100) + st->mode = MODE_CELT_ONLY; + if (st->lfe) + st->mode = MODE_CELT_ONLY; + + if (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; + } + } + } + + /* When encoding multiframes, we can ask for a switch to CELT only in the last frame. This switch + * is processed above as the requested mode shouldn't interrupt stereo->mono transition. */ + 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; + } + + /* Update equivalent rate with mode decision. */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->stream_channels, st->Fs/frame_size, + st->use_vbr, st->mode, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + if (st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) + { + silk_EncControlStruct dummy; + silk_InitEncoder( silk_enc, st->arch, &dummy); + prefill=1; + } + + /* Automatic (rate-dependent) bandwidth selection */ + if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch) + { + const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds; + opus_int32 bandwidth_thresholds[8]; + int bandwidth = OPUS_BANDWIDTH_FULLBAND; + + 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->auto_bandwidth >= bandwidth) + threshold -= hysteresis; + else + threshold += hysteresis; + } + if (equiv_rate >= threshold) + break; + } while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND); + /* We don't use mediumband anymore, except when explicitly requested or during + mode transitions. */ + if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + bandwidth = OPUS_BANDWIDTH_WIDEBAND; + st->bandwidth = st->auto_bandwidth = bandwidth; + /* Prevents any transition to SWB/FB until the SILK layer has fully + switched to WB mode and turned the variable LP filter off */ + if (!st->first && st->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + } + + if (st->bandwidth>st->max_bandwidth) + st->bandwidth = st->max_bandwidth; + + if (st->user_bandwidth != OPUS_AUTO) + st->bandwidth = st->user_bandwidth; + + /* This prevents us from using hybrid at unsafe CBR/max rates */ + if (st->mode != MODE_CELT_ONLY && max_rate < 15000) + { + st->bandwidth = IMIN(st->bandwidth, OPUS_BANDWIDTH_WIDEBAND); + } + + /* Prevents Opus from wasting bits on frequencies that are above + the Nyquist rate of the input signal */ + if (st->Fs <= 24000 && st->bandwidth > OPUS_BANDWIDTH_SUPERWIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + if (st->Fs <= 16000 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + if (st->Fs <= 12000 && st->bandwidth > OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + if (st->Fs <= 8000 && st->bandwidth > OPUS_BANDWIDTH_NARROWBAND) + st->bandwidth = OPUS_BANDWIDTH_NARROWBAND; +#ifndef DISABLE_FLOAT_API + /* Use detected bandwidth to reduce the encoded bandwidth. */ + if (st->detected_bandwidth && st->user_bandwidth == OPUS_AUTO) + { + int min_detected_bandwidth; + /* Makes bandwidth detection more conservative just in case the detector + gets it wrong when we could have coded a high bandwidth transparently. + When operating in SILK/hybrid mode, we don't go below wideband to avoid + more complicated switches that require redundancy. */ + if (equiv_rate <= 18000*st->stream_channels && st->mode == MODE_CELT_ONLY) + min_detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + else if (equiv_rate <= 24000*st->stream_channels && st->mode == MODE_CELT_ONLY) + min_detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + else if (equiv_rate <= 30000*st->stream_channels) + min_detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + else if (equiv_rate <= 44000*st->stream_channels) + min_detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + else + min_detected_bandwidth = OPUS_BANDWIDTH_FULLBAND; + + st->detected_bandwidth = IMAX(st->detected_bandwidth, min_detected_bandwidth); + st->bandwidth = IMIN(st->bandwidth, st->detected_bandwidth); + } +#endif + st->silk_mode.LBRR_coded = decide_fec(st->silk_mode.useInBandFEC, st->silk_mode.packetLossPercentage, + st->silk_mode.LBRR_coded, st->mode, &st->bandwidth, equiv_rate); + celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(lsb_depth)); + + /* CELT mode doesn't support mediumband, use wideband instead */ + if (st->mode == MODE_CELT_ONLY && st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + if (st->lfe) + st->bandwidth = OPUS_BANDWIDTH_NARROWBAND; + + 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; + + /* Can't support higher than >60 ms frames, and >20 ms when in Hybrid or CELT-only modes */ + if ((frame_size > st->Fs/50 && (st->mode != MODE_SILK_ONLY)) || frame_size > 3*st->Fs/50) + { + int enc_frame_size; + int nb_frames; + + if (st->mode == MODE_SILK_ONLY) + { + if (frame_size == 2*st->Fs/25) /* 80 ms -> 2x 40 ms */ + enc_frame_size = st->Fs/25; + else if (frame_size == 3*st->Fs/25) /* 120 ms -> 2x 60 ms */ + enc_frame_size = 3*st->Fs/50; + else /* 100 ms -> 5x 20 ms */ + enc_frame_size = st->Fs/50; + } + else + enc_frame_size = st->Fs/50; + + nb_frames = frame_size/enc_frame_size; + +#ifndef DISABLE_FLOAT_API + if (analysis_read_pos_bak!= -1) + { + st->analysis.read_pos = analysis_read_pos_bak; + st->analysis.read_subframe = analysis_read_subframe_bak; + } +#endif + + ret = encode_multiframe_packet(st, pcm, nb_frames, enc_frame_size, data, + out_data_bytes, to_celt, lsb_depth, float_api); + + RESTORE_STACK; + return ret; + } + + /* For the first frame at a new SILK bandwidth */ + if (st->silk_bw_switch) + { + redundancy = 1; + celt_to_silk = 1; + st->silk_bw_switch = 0; + /* Do a prefill without reseting the sampling rate control. */ + prefill=2; + } + + /* If we decided to go with CELT, make sure redundancy is off, no matter what + we decided earlier. */ + if (st->mode == MODE_CELT_ONLY) + redundancy = 0; + + if (redundancy) + { + redundancy_bytes = compute_redundancy_bytes(max_data_bytes, st->bitrate_bps, frame_rate, st->stream_channels); + if (redundancy_bytes == 0) + redundancy = 0; + } + + /* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, curr_bandwidth); */ + bytes_target = IMIN(max_data_bytes-redundancy_bytes, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1; + + data += 1; + + ec_enc_init(&enc, data, max_data_bytes-1); + + ALLOC(pcm_buf, (total_buffer+frame_size)*st->channels, opus_val16); + OPUS_COPY(pcm_buf, &st->delay_buffer[(st->encoder_buffer-total_buffer)*st->channels], total_buffer*st->channels); + + if (st->mode == MODE_CELT_ONLY) + hp_freq_smth1 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + else + hp_freq_smth1 = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.variable_HP_smth1_Q15; + + st->variable_HP_smth2_Q15 = silk_SMLAWB( st->variable_HP_smth2_Q15, + hp_freq_smth1 - st->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) ); + + /* convert from log scale to Hertz */ + cutoff_Hz = silk_log2lin( silk_RSHIFT( st->variable_HP_smth2_Q15, 8 ) ); + + if (st->application == OPUS_APPLICATION_VOIP) + { + hp_cutoff(pcm, cutoff_Hz, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs, st->arch); + } else { + dc_reject(pcm, 3, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs); + } +#ifndef FIXED_POINT + if (float_api) + { + opus_val32 sum; + sum = celt_inner_prod(&pcm_buf[total_buffer*st->channels], &pcm_buf[total_buffer*st->channels], frame_size*st->channels, st->arch); + /* This should filter out both NaNs and ridiculous signals that could + cause NaNs further down. */ + if (!(sum < 1e9f) || celt_isnan(sum)) + { + OPUS_CLEAR(&pcm_buf[total_buffer*st->channels], frame_size*st->channels); + st->hp_mem[0] = st->hp_mem[1] = st->hp_mem[2] = st->hp_mem[3] = 0; + } + } +#endif + + + /* SILK processing */ + HB_gain = Q15ONE; + if (st->mode != MODE_CELT_ONLY) + { + opus_int32 total_bitRate, celt_rate; +#ifdef FIXED_POINT + const opus_int16 *pcm_silk; +#else + VARDECL(opus_int16, pcm_silk); + ALLOC(pcm_silk, st->channels*frame_size, opus_int16); +#endif + + /* Distribute bits between SILK and CELT */ + total_bitRate = 8 * bytes_target * frame_rate; + if( st->mode == MODE_HYBRID ) { + /* Base rate for SILK */ + st->silk_mode.bitRate = compute_silk_rate_for_hybrid(total_bitRate, + curr_bandwidth, st->Fs == 50 * frame_size, st->use_vbr, st->silk_mode.LBRR_coded, + st->stream_channels); + if (!st->energy_masking) + { + /* Increasingly attenuate high band when it gets allocated fewer bits */ + celt_rate = total_bitRate - st->silk_mode.bitRate; + HB_gain = Q15ONE - SHR32(celt_exp2(-celt_rate * QCONST16(1.f/1024, 10)), 1); + } + } else { + /* SILK gets all bits */ + st->silk_mode.bitRate = total_bitRate; + } + + /* Surround masking for SILK */ + if (st->energy_masking && st->use_vbr && !st->lfe) + { + opus_val32 mask_sum=0; + opus_val16 masking_depth; + opus_int32 rate_offset; + int c; + int end = 17; + opus_int16 srate = 16000; + if (st->bandwidth == OPUS_BANDWIDTH_NARROWBAND) + { + end = 13; + srate = 8000; + } else if (st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + { + end = 15; + srate = 12000; + } + for (c=0;c<st->channels;c++) + { + for(i=0;i<end;i++) + { + opus_val16 mask; + mask = MAX16(MIN16(st->energy_masking[21*c+i], + QCONST16(.5f, DB_SHIFT)), -QCONST16(2.0f, DB_SHIFT)); + if (mask > 0) + mask = HALF16(mask); + mask_sum += mask; + } + } + /* Conservative rate reduction, we cut the masking in half */ + masking_depth = mask_sum / end*st->channels; + masking_depth += QCONST16(.2f, DB_SHIFT); + rate_offset = (opus_int32)PSHR32(MULT16_16(srate, masking_depth), DB_SHIFT); + rate_offset = MAX32(rate_offset, -2*st->silk_mode.bitRate/3); + /* Split the rate change between the SILK and CELT part for hybrid. */ + if (st->bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND || st->bandwidth==OPUS_BANDWIDTH_FULLBAND) + st->silk_mode.bitRate += 3*rate_offset/5; + else + st->silk_mode.bitRate += rate_offset; + } + + 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 { + celt_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; + } + + st->silk_mode.maxInternalSampleRate = 16000; + if (st->mode == MODE_SILK_ONLY) + { + opus_int32 effective_max_rate = max_rate; + if (frame_rate > 50) + effective_max_rate = effective_max_rate*2/3; + if (effective_max_rate < 8000) + { + st->silk_mode.maxInternalSampleRate = 12000; + st->silk_mode.desiredInternalSampleRate = IMIN(12000, st->silk_mode.desiredInternalSampleRate); + } + if (effective_max_rate < 7000) + { + st->silk_mode.maxInternalSampleRate = 8000; + st->silk_mode.desiredInternalSampleRate = IMIN(8000, st->silk_mode.desiredInternalSampleRate); + } + } + + st->silk_mode.useCBR = !st->use_vbr; + + /* Call SILK encoder for the low band */ + + /* Max bits for SILK, counting ToC, redundancy bytes, and optionally redundancy. */ + st->silk_mode.maxBits = (max_data_bytes-1)*8; + if (redundancy && redundancy_bytes >= 2) + { + /* Counting 1 bit for redundancy position and 20 bits for flag+size (only for hybrid). */ + st->silk_mode.maxBits -= redundancy_bytes*8 + 1; + if (st->mode == MODE_HYBRID) + st->silk_mode.maxBits -= 20; + } + if (st->silk_mode.useCBR) + { + if (st->mode == MODE_HYBRID) + { + st->silk_mode.maxBits = IMIN(st->silk_mode.maxBits, st->silk_mode.bitRate * frame_size / st->Fs); + } + } else { + /* Constrained VBR. */ + if (st->mode == MODE_HYBRID) + { + /* Compute SILK bitrate corresponding to the max total bits available */ + opus_int32 maxBitRate = compute_silk_rate_for_hybrid(st->silk_mode.maxBits*st->Fs / frame_size, + curr_bandwidth, st->Fs == 50 * frame_size, st->use_vbr, st->silk_mode.LBRR_coded, + st->stream_channels); + st->silk_mode.maxBits = maxBitRate * frame_size / st->Fs; + } + } + + if (prefill) + { + opus_int32 zero=0; + int prefill_offset; + /* Use a smooth onset for the SILK prefill to avoid the encoder trying to encode + a discontinuity. The exact location is what we need to avoid leaving any "gap" + in the audio when mixing with the redundant CELT frame. Here we can afford to + overwrite st->delay_buffer because the only thing that uses it before it gets + rewritten is tmp_prefill[] and even then only the part after the ramp really + gets used (rather than sent to the encoder and discarded) */ + prefill_offset = st->channels*(st->encoder_buffer-st->delay_compensation-st->Fs/400); + gain_fade(st->delay_buffer+prefill_offset, st->delay_buffer+prefill_offset, + 0, Q15ONE, celt_mode->overlap, st->Fs/400, st->channels, celt_mode->window, st->Fs); + OPUS_CLEAR(st->delay_buffer, prefill_offset); +#ifdef FIXED_POINT + pcm_silk = st->delay_buffer; +#else + for (i=0;i<st->encoder_buffer*st->channels;i++) + pcm_silk[i] = FLOAT2INT16(st->delay_buffer[i]); +#endif + silk_Encode( silk_enc, &st->silk_mode, pcm_silk, st->encoder_buffer, NULL, &zero, prefill, activity ); + /* Prevent a second switch in the real encode call. */ + st->silk_mode.opusCanSwitch = 0; + } + +#ifdef FIXED_POINT + pcm_silk = pcm_buf+total_buffer*st->channels; +#else + for (i=0;i<frame_size*st->channels;i++) + pcm_silk[i] = FLOAT2INT16(pcm_buf[total_buffer*st->channels + i]); +#endif + ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0, activity ); + if( ret ) { + /*fprintf (stderr, "SILK encode error: %d\n", ret);*/ + /* Handle error */ + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + + /* 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 { + celt_assert( st->silk_mode.internalSampleRate == 16000 ); + } + + st->silk_mode.opusCanSwitch = st->silk_mode.switchReady && !st->nonfinal_frame; + + 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; + } + + /* FIXME: How do we allocate the redundancy for CBR? */ + if (st->silk_mode.opusCanSwitch) + { + redundancy_bytes = compute_redundancy_bytes(max_data_bytes, st->bitrate_bps, frame_rate, st->stream_channels); + redundancy = (redundancy_bytes != 0); + celt_to_silk = 0; + st->silk_bw_switch = 1; + } + } + + /* CELT processing */ + { + int endband=21; + + switch(curr_bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + endband = 13; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + case OPUS_BANDWIDTH_WIDEBAND: + endband = 17; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + endband = 19; + break; + case OPUS_BANDWIDTH_FULLBAND: + endband = 21; + break; + } + celt_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband)); + celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels)); + } + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + if (st->mode != MODE_SILK_ONLY) + { + opus_val32 celt_pred=2; + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + /* We may still decide to disable prediction later */ + if (st->silk_mode.reducedDependency) + celt_pred = 0; + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(celt_pred)); + + if (st->mode == MODE_HYBRID) + { + if( st->use_vbr ) { + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps-st->silk_mode.bitRate)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(0)); + } + } 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)); + } + } + } + + ALLOC(tmp_prefill, st->channels*st->Fs/400, opus_val16); + if (st->mode != MODE_SILK_ONLY && st->mode != st->prev_mode && st->prev_mode > 0) + { + OPUS_COPY(tmp_prefill, &st->delay_buffer[(st->encoder_buffer-total_buffer-st->Fs/400)*st->channels], st->channels*st->Fs/400); + } + + if (st->channels*(st->encoder_buffer-(frame_size+total_buffer)) > 0) + { + OPUS_MOVE(st->delay_buffer, &st->delay_buffer[st->channels*frame_size], st->channels*(st->encoder_buffer-frame_size-total_buffer)); + OPUS_COPY(&st->delay_buffer[st->channels*(st->encoder_buffer-frame_size-total_buffer)], + &pcm_buf[0], + (frame_size+total_buffer)*st->channels); + } else { + OPUS_COPY(st->delay_buffer, &pcm_buf[(frame_size+total_buffer-st->encoder_buffer)*st->channels], st->encoder_buffer*st->channels); + } + /* gain_fade() and stereo_fade() need to be after the buffer copying + because we don't want any of this to affect the SILK part */ + if( st->prev_HB_gain < Q15ONE || HB_gain < Q15ONE ) { + gain_fade(pcm_buf, pcm_buf, + st->prev_HB_gain, HB_gain, celt_mode->overlap, frame_size, st->channels, celt_mode->window, st->Fs); + } + st->prev_HB_gain = HB_gain; + if (st->mode != MODE_HYBRID || st->stream_channels==1) + { + if (equiv_rate > 32000) + st->silk_mode.stereoWidth_Q14 = 16384; + else if (equiv_rate < 16000) + st->silk_mode.stereoWidth_Q14 = 0; + else + st->silk_mode.stereoWidth_Q14 = 16384 - 2048*(opus_int32)(32000-equiv_rate)/(equiv_rate-14000); + } + if( !st->energy_masking && st->channels == 2 ) { + /* Apply stereo width reduction (at low bitrates) */ + if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) { + opus_val16 g1, g2; + g1 = st->hybrid_stereo_width_Q14; + g2 = (opus_val16)(st->silk_mode.stereoWidth_Q14); +#ifdef FIXED_POINT + g1 = g1==16384 ? Q15ONE : SHL16(g1,1); + g2 = g2==16384 ? Q15ONE : SHL16(g2,1); +#else + g1 *= (1.f/16384); + g2 *= (1.f/16384); +#endif + stereo_fade(pcm_buf, pcm_buf, g1, g2, celt_mode->overlap, + frame_size, st->channels, celt_mode->window, st->Fs); + st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14; + } + } + + if ( st->mode != MODE_CELT_ONLY && ec_tell(&enc)+17+20*(st->mode == MODE_HYBRID) <= 8*(max_data_bytes-1)) + { + /* For SILK mode, the redundancy is inferred from the length */ + if (st->mode == MODE_HYBRID) + ec_enc_bit_logp(&enc, redundancy, 12); + if (redundancy) + { + int max_redundancy; + ec_enc_bit_logp(&enc, celt_to_silk, 1); + if (st->mode == MODE_HYBRID) + { + /* Reserve the 8 bits needed for the redundancy length, + and at least a few bits for CELT if possible */ + max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+8+3+7)>>3); + } + else + max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+7)>>3); + /* Target the same bit-rate for redundancy as for the rest, + up to a max of 257 bytes */ + redundancy_bytes = IMIN(max_redundancy, redundancy_bytes); + redundancy_bytes = IMIN(257, IMAX(2, redundancy_bytes)); + if (st->mode == MODE_HYBRID) + ec_enc_uint(&enc, redundancy_bytes-2, 256); + } + } else { + redundancy = 0; + } + + if (!redundancy) + { + st->silk_bw_switch = 0; + redundancy_bytes = 0; + } + if (st->mode != MODE_CELT_ONLY)start_band=17; + + if (st->mode == MODE_SILK_ONLY) + { + ret = (ec_tell(&enc)+7)>>3; + ec_enc_done(&enc); + nb_compr_bytes = ret; + } else { + nb_compr_bytes = (max_data_bytes-1)-redundancy_bytes; + ec_enc_shrink(&enc, nb_compr_bytes); + } + +#ifndef DISABLE_FLOAT_API + if (redundancy || st->mode != MODE_SILK_ONLY) + celt_encoder_ctl(celt_enc, CELT_SET_ANALYSIS(&analysis_info)); +#endif + if (st->mode == MODE_HYBRID) { + SILKInfo info; + info.signalType = st->silk_mode.signalType; + info.offset = st->silk_mode.offset; + celt_encoder_ctl(celt_enc, CELT_SET_SILK_INFO(&info)); + } + + /* 5 ms redundant frame for CELT->SILK */ + if (redundancy && celt_to_silk) + { + int err; + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + err = celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + } + + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(start_band)); + + if (st->mode != MODE_SILK_ONLY) + { + if (st->mode != st->prev_mode && st->prev_mode > 0) + { + unsigned char dummy[2]; + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + + /* Prefilling */ + celt_encode_with_ec(celt_enc, tmp_prefill, st->Fs/400, dummy, 2, NULL); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + } + /* If false, we already busted the budget and we'll end up with a "PLC frame" */ + if (ec_tell(&enc) <= 8*nb_compr_bytes) + { + /* Set the bitrate again if it was overridden in the redundancy code above*/ + if (redundancy && celt_to_silk && st->mode==MODE_HYBRID && st->use_vbr) + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps-st->silk_mode.bitRate)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(st->use_vbr)); + ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc); + if (ret < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + /* Put CELT->SILK redundancy data in the right place. */ + if (redundancy && celt_to_silk && st->mode==MODE_HYBRID && st->use_vbr) + { + OPUS_MOVE(data+ret, data+nb_compr_bytes, redundancy_bytes); + nb_compr_bytes = nb_compr_bytes+redundancy_bytes; + } + } + } + + /* 5 ms redundant frame for SILK->CELT */ + if (redundancy && !celt_to_silk) + { + int err; + unsigned char dummy[2]; + int N2, N4; + N2 = st->Fs/200; + N4 = st->Fs/400; + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + + if (st->mode == MODE_HYBRID) + { + /* Shrink packet to what the encoder actually used. */ + nb_compr_bytes = ret; + ec_enc_shrink(&enc, nb_compr_bytes); + } + /* NOTE: We could speed this up slightly (at the expense of code size) by just adding a function that prefills the buffer */ + celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2-N4), N4, dummy, 2, NULL); + + err = celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + } + + + + /* Signalling the mode in the first byte */ + data--; + data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + + st->rangeFinal = enc.rng ^ redundant_rng; + + if (to_celt) + st->prev_mode = MODE_CELT_ONLY; + else + st->prev_mode = st->mode; + st->prev_channels = st->stream_channels; + st->prev_framesize = frame_size; + + st->first = 0; + + /* DTX decision */ +#ifndef DISABLE_FLOAT_API + if (st->use_dtx && (analysis_info.valid || is_silence)) + { + if (decide_dtx_mode(activity, &st->nb_no_activity_ms_Q1, 2*1000*frame_size/st->Fs)) + { + st->rangeFinal = 0; + data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + RESTORE_STACK; + return 1; + } + } else { + st->nb_no_activity_ms_Q1 = 0; + } +#endif + + /* In the unlikely case that the SILK encoder busted its target, tell + the decoder to call the PLC */ + if (ec_tell(&enc) > (max_data_bytes-1)*8) + { + if (max_data_bytes < 2) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + data[1] = 0; + ret = 1; + st->rangeFinal = 0; + } else if (st->mode==MODE_SILK_ONLY&&!redundancy) + { + /*When in LPC only mode it's perfectly + reasonable to strip off trailing zero bytes as + the required range decoder behavior is to + fill these in. This can't be done when the MDCT + modes are used because the decoder needs to know + the actual length for allocation purposes.*/ + while(ret>2&&data[ret]==0)ret--; + } + /* Count ToC and redundancy */ + ret += 1+redundancy_bytes; + if (!st->use_vbr) + { + if (opus_packet_pad(data, ret, max_data_bytes) != OPUS_OK) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + ret = max_data_bytes; + } + RESTORE_STACK; + return ret; +} + +#ifdef FIXED_POINT + +#ifndef DISABLE_FLOAT_API +opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + int frame_size; + VARDECL(opus_int16, in); + ALLOC_STACK; + + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + if (frame_size <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + ALLOC(in, frame_size*st->channels, opus_int16); + + for (i=0;i<frame_size*st->channels;i++) + in[i] = FLOAT2INT16(pcm[i]); + ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1); + RESTORE_STACK; + return ret; +} +#endif + +opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 out_data_bytes) +{ + int frame_size; + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0); +} + +#else +opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + int frame_size; + VARDECL(float, in); + ALLOC_STACK; + + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + if (frame_size <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + ALLOC(in, frame_size*st->channels, float); + + for (i=0;i<frame_size*st->channels;i++) + in[i] = (1.0f/32768)*pcm[i]; + ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0); + RESTORE_STACK; + return ret; +} +opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 out_data_bytes) +{ + int frame_size; + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 24, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1); +} +#endif + + +int opus_encoder_ctl(OpusEncoder *st, int request, ...) +{ + int ret; + CELTEncoder *celt_enc; + va_list ap; + + ret = OPUS_OK; + va_start(ap, request); + + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + switch (request) + { + case OPUS_SET_APPLICATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ( (value != OPUS_APPLICATION_VOIP && value != OPUS_APPLICATION_AUDIO + && value != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + || (!st->first && st->application != value)) + { + ret = OPUS_BAD_ARG; + break; + } + st->application = value; +#ifndef DISABLE_FLOAT_API + st->analysis.application = value; +#endif + } + break; + case OPUS_GET_APPLICATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->application; + } + break; + case OPUS_SET_BITRATE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value != OPUS_AUTO && value != OPUS_BITRATE_MAX) + { + if (value <= 0) + goto bad_arg; + else if (value <= 500) + value = 500; + else if (value > (opus_int32)300000*st->channels) + value = (opus_int32)300000*st->channels; + } + st->user_bitrate_bps = value; + } + break; + case OPUS_GET_BITRATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = user_bitrate_to_bitrate(st, st->prev_framesize, 1276); + } + break; + case OPUS_SET_FORCE_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if((value<1 || value>st->channels) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->force_channels = value; + } + break; + case OPUS_GET_FORCE_CHANNELS_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->force_channels; + } + break; + case OPUS_SET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) + { + goto bad_arg; + } + st->max_bandwidth = value; + if (st->max_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->max_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->max_bandwidth; + } + break; + case OPUS_SET_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->user_bandwidth = value; + if (st->user_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->user_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->bandwidth; + } + break; + case OPUS_SET_DTX_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->use_dtx = value; + } + break; + case OPUS_GET_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->use_dtx; + } + break; + case OPUS_SET_COMPLEXITY_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>10) + { + goto bad_arg; + } + st->silk_mode.complexity = value; + celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(value)); + } + break; + case OPUS_GET_COMPLEXITY_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->silk_mode.complexity; + } + break; + case OPUS_SET_INBAND_FEC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>2) + { + goto bad_arg; + } + st->fec_config = value; + st->silk_mode.useInBandFEC = (value != 0); + } + break; + case OPUS_GET_INBAND_FEC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->fec_config; + } + break; + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < 0 || value > 100) + { + goto bad_arg; + } + st->silk_mode.packetLossPercentage = value; + celt_encoder_ctl(celt_enc, OPUS_SET_PACKET_LOSS_PERC(value)); + } + break; + case OPUS_GET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->silk_mode.packetLossPercentage; + } + break; + case OPUS_SET_VBR_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->use_vbr = value; + st->silk_mode.useCBR = 1-value; + } + break; + case OPUS_GET_VBR_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->use_vbr; + } + break; + case OPUS_SET_VOICE_RATIO_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<-1 || value>100) + { + goto bad_arg; + } + st->voice_ratio = value; + } + break; + case OPUS_GET_VOICE_RATIO_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->voice_ratio; + } + break; + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->vbr_constraint = value; + } + break; + case OPUS_GET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->vbr_constraint; + } + break; + case OPUS_SET_SIGNAL_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value!=OPUS_AUTO && value!=OPUS_SIGNAL_VOICE && value!=OPUS_SIGNAL_MUSIC) + { + goto bad_arg; + } + st->signal_type = value; + } + break; + case OPUS_GET_SIGNAL_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->signal_type; + } + break; + case OPUS_GET_LOOKAHEAD_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->Fs/400; + if (st->application != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + *value += st->delay_compensation; + } + break; + case OPUS_GET_SAMPLE_RATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->Fs; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + if (!value) + { + goto bad_arg; + } + *value = st->rangeFinal; + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<8 || value>24) + { + goto bad_arg; + } + st->lsb_depth=value; + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->lsb_depth; + } + break; + case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value != OPUS_FRAMESIZE_ARG && value != OPUS_FRAMESIZE_2_5_MS && + value != OPUS_FRAMESIZE_5_MS && value != OPUS_FRAMESIZE_10_MS && + value != OPUS_FRAMESIZE_20_MS && value != OPUS_FRAMESIZE_40_MS && + value != OPUS_FRAMESIZE_60_MS && value != OPUS_FRAMESIZE_80_MS && + value != OPUS_FRAMESIZE_100_MS && value != OPUS_FRAMESIZE_120_MS) + { + goto bad_arg; + } + st->variable_duration = value; + } + break; + case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->variable_duration; + } + break; + case OPUS_SET_PREDICTION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value > 1 || value < 0) + goto bad_arg; + st->silk_mode.reducedDependency = value; + } + break; + case OPUS_GET_PREDICTION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + goto bad_arg; + *value = st->silk_mode.reducedDependency; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + celt_encoder_ctl(celt_enc, OPUS_SET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + celt_encoder_ctl(celt_enc, OPUS_GET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_RESET_STATE: + { + void *silk_enc; + silk_EncControlStruct dummy; + char *start; + silk_enc = (char*)st+st->silk_enc_offset; +#ifndef DISABLE_FLOAT_API + tonality_analysis_reset(&st->analysis); +#endif + + start = (char*)&st->OPUS_ENCODER_RESET_START; + OPUS_CLEAR(start, sizeof(OpusEncoder) - (start - (char*)st)); + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + silk_InitEncoder( silk_enc, st->arch, &dummy ); + st->stream_channels = st->channels; + st->hybrid_stereo_width_Q14 = 1 << 14; + st->prev_HB_gain = Q15ONE; + st->first = 1; + st->mode = MODE_HYBRID; + st->bandwidth = OPUS_BANDWIDTH_FULLBAND; + st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + } + break; + case OPUS_SET_FORCE_MODE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < MODE_SILK_ONLY || value > MODE_CELT_ONLY) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->user_forced_mode = value; + } + break; + case OPUS_SET_LFE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->lfe = value; + ret = celt_encoder_ctl(celt_enc, OPUS_SET_LFE(value)); + } + break; + case OPUS_SET_ENERGY_MASK_REQUEST: + { + opus_val16 *value = va_arg(ap, opus_val16*); + st->energy_masking = value; + ret = celt_encoder_ctl(celt_enc, OPUS_SET_ENERGY_MASK(value)); + } + break; + case OPUS_GET_IN_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + if (st->silk_mode.useDTX && (st->prev_mode == MODE_SILK_ONLY || st->prev_mode == MODE_HYBRID)) { + /* DTX determined by Silk. */ + silk_encoder *silk_enc = (silk_encoder*)(void *)((char*)st+st->silk_enc_offset); + *value = silk_enc->state_Fxx[0].sCmn.noSpeechCounter >= NB_SPEECH_FRAMES_BEFORE_DTX; + /* Stereo: check second channel unless only the middle channel was encoded. */ + if(*value == 1 && st->silk_mode.nChannelsInternal == 2 && silk_enc->prev_decode_only_middle == 0) { + *value = silk_enc->state_Fxx[1].sCmn.noSpeechCounter >= NB_SPEECH_FRAMES_BEFORE_DTX; + } + } +#ifndef DISABLE_FLOAT_API + else if (st->use_dtx) { + /* DTX determined by Opus. */ + *value = st->nb_no_activity_ms_Q1 >= NB_SPEECH_FRAMES_BEFORE_DTX*20*2; + } +#endif + else { + *value = 0; + } + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (!value) + { + goto bad_arg; + } + ret = celt_encoder_ctl(celt_enc, CELT_GET_MODE(value)); + } + break; + default: + /* fprintf(stderr, "unknown opus_encoder_ctl() request: %d", request);*/ + ret = OPUS_UNIMPLEMENTED; + break; + } + va_end(ap); + return ret; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +} + +void opus_encoder_destroy(OpusEncoder *st) +{ + opus_free(st); +} |