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virtualx-engine/thirdparty/opus/silk/control_codec.c

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Add support for Opus audio format
2015-10-02 19:20:50 +02:00
/***********************************************************************
Copyright (c) 2006-2011, Skype Limited. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of Internet Society, IETF or IETF Trust, nor the
names of specific contributors, may be used to endorse or promote
products derived from this software without specific prior written
permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
***********************************************************************/
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef FIXED_POINT
#include "main_FIX.h"
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#define silk_encoder_state_Fxx silk_encoder_state_FIX
#else
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
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#include "main_FLP.h"
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#define silk_encoder_state_Fxx silk_encoder_state_FLP
#endif
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 20:37:38 +02:00
#include "stack_alloc.h"
#include "tuning_parameters.h"
#include "pitch_est_defines.h"
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static opus_int silk_setup_resamplers(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz /* I */
);
static opus_int silk_setup_fs(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz, /* I */
opus_int PacketSize_ms /* I */
);
static opus_int silk_setup_complexity(
silk_encoder_state *psEncC, /* I/O */
opus_int Complexity /* I */
);
static OPUS_INLINE opus_int silk_setup_LBRR(
silk_encoder_state *psEncC, /* I/O */
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const silk_EncControlStruct *encControl /* I */
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);
/* Control encoder */
opus_int silk_control_encoder(
silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */
silk_EncControlStruct *encControl, /* I Control structure */
const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */
const opus_int channelNb, /* I Channel number */
const opus_int force_fs_kHz
)
{
opus_int fs_kHz, ret = 0;
psEnc->sCmn.useDTX = encControl->useDTX;
psEnc->sCmn.useCBR = encControl->useCBR;
psEnc->sCmn.API_fs_Hz = encControl->API_sampleRate;
psEnc->sCmn.maxInternal_fs_Hz = encControl->maxInternalSampleRate;
psEnc->sCmn.minInternal_fs_Hz = encControl->minInternalSampleRate;
psEnc->sCmn.desiredInternal_fs_Hz = encControl->desiredInternalSampleRate;
psEnc->sCmn.useInBandFEC = encControl->useInBandFEC;
psEnc->sCmn.nChannelsAPI = encControl->nChannelsAPI;
psEnc->sCmn.nChannelsInternal = encControl->nChannelsInternal;
psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch;
psEnc->sCmn.channelNb = channelNb;
if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) {
if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) {
/* Change in API sampling rate in the middle of encoding a packet */
ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz );
}
return ret;
}
/* Beyond this point we know that there are no previously coded frames in the payload buffer */
/********************************************/
/* Determine internal sampling rate */
/********************************************/
fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn, encControl );
if( force_fs_kHz ) {
fs_kHz = force_fs_kHz;
}
/********************************************/
/* Prepare resampler and buffered data */
/********************************************/
ret += silk_setup_resamplers( psEnc, fs_kHz );
/********************************************/
/* Set internal sampling frequency */
/********************************************/
ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms );
/********************************************/
/* Set encoding complexity */
/********************************************/
ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity );
/********************************************/
/* Set packet loss rate measured by farend */
/********************************************/
psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage;
/********************************************/
/* Set LBRR usage */
/********************************************/
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ret += silk_setup_LBRR( &psEnc->sCmn, encControl );
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psEnc->sCmn.controlled_since_last_payload = 1;
return ret;
}
static opus_int silk_setup_resamplers(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz /* I */
)
{
opus_int ret = SILK_NO_ERROR;
SAVE_STACK;
if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz )
{
if( psEnc->sCmn.fs_kHz == 0 ) {
/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000, 1 );
} else {
VARDECL( opus_int16, x_buf_API_fs_Hz );
VARDECL( silk_resampler_state_struct, temp_resampler_state );
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 20:37:38 +02:00
#ifdef FIXED_POINT
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opus_int16 *x_bufFIX = psEnc->x_buf;
#else
VARDECL( opus_int16, x_bufFIX );
opus_int32 new_buf_samples;
#endif
opus_int32 api_buf_samples;
opus_int32 old_buf_samples;
opus_int32 buf_length_ms;
buf_length_ms = silk_LSHIFT( psEnc->sCmn.nb_subfr * 5, 1 ) + LA_SHAPE_MS;
old_buf_samples = buf_length_ms * psEnc->sCmn.fs_kHz;
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 20:37:38 +02:00
#ifndef FIXED_POINT
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new_buf_samples = buf_length_ms * fs_kHz;
ALLOC( x_bufFIX, silk_max( old_buf_samples, new_buf_samples ),
opus_int16 );
silk_float2short_array( x_bufFIX, psEnc->x_buf, old_buf_samples );
#endif
/* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
ALLOC( temp_resampler_state, 1, silk_resampler_state_struct );
ret += silk_resampler_init( temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
/* Calculate number of samples to temporarily upsample */
api_buf_samples = buf_length_ms * silk_DIV32_16( psEnc->sCmn.API_fs_Hz, 1000 );
/* Temporary resampling of x_buf data to API_fs_Hz */
ALLOC( x_buf_API_fs_Hz, api_buf_samples, opus_int16 );
ret += silk_resampler( temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, old_buf_samples );
/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 );
/* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */
ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, api_buf_samples );
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 20:37:38 +02:00
#ifndef FIXED_POINT
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silk_short2float_array( psEnc->x_buf, x_bufFIX, new_buf_samples);
#endif
}
}
psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
RESTORE_STACK;
return ret;
}
static opus_int silk_setup_fs(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz, /* I */
opus_int PacketSize_ms /* I */
)
{
opus_int ret = SILK_NO_ERROR;
/* Set packet size */
if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) {
if( ( PacketSize_ms != 10 ) &&
( PacketSize_ms != 20 ) &&
( PacketSize_ms != 40 ) &&
( PacketSize_ms != 60 ) ) {
ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED;
}
if( PacketSize_ms <= 10 ) {
psEnc->sCmn.nFramesPerPacket = 1;
psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1;
psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
if( psEnc->sCmn.fs_kHz == 8 ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
}
} else {
psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
if( psEnc->sCmn.fs_kHz == 8 ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
}
}
psEnc->sCmn.PacketSize_ms = PacketSize_ms;
psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
}
/* Set internal sampling frequency */
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celt_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
celt_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
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if( psEnc->sCmn.fs_kHz != fs_kHz ) {
/* reset part of the state */
silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) );
silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) );
silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) );
psEnc->sCmn.inputBufIx = 0;
psEnc->sCmn.nFramesEncoded = 0;
psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
/* Initialize non-zero parameters */
psEnc->sCmn.prevLag = 100;
psEnc->sCmn.first_frame_after_reset = 1;
psEnc->sShape.LastGainIndex = 10;
psEnc->sCmn.sNSQ.lagPrev = 100;
psEnc->sCmn.sNSQ.prev_gain_Q16 = 65536;
psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
psEnc->sCmn.fs_kHz = fs_kHz;
if( psEnc->sCmn.fs_kHz == 8 ) {
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
}
} else {
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
}
}
if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) {
psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER;
psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB;
} else {
psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER;
psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB;
}
psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz;
psEnc->sCmn.frame_length = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz );
psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz );
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
} else {
psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
}
if( psEnc->sCmn.fs_kHz == 16 ) {
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
} else if( psEnc->sCmn.fs_kHz == 12 ) {
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
} else {
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
}
}
/* Check that settings are valid */
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celt_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
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return ret;
}
static opus_int silk_setup_complexity(
silk_encoder_state *psEncC, /* I/O */
opus_int Complexity /* I */
)
{
opus_int ret = 0;
/* Set encoding complexity */
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celt_assert( Complexity >= 0 && Complexity <= 10 );
if( Complexity < 1 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 );
psEncC->pitchEstimationLPCOrder = 6;
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psEncC->shapingLPCOrder = 12;
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psEncC->la_shape = 3 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 1;
psEncC->useInterpolatedNLSFs = 0;
psEncC->NLSF_MSVQ_Survivors = 2;
psEncC->warping_Q16 = 0;
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} else if( Complexity < 2 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 );
psEncC->pitchEstimationLPCOrder = 8;
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psEncC->shapingLPCOrder = 14;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 1;
psEncC->useInterpolatedNLSFs = 0;
Update opus to 1.3.1 and opusfile to 0.11
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psEncC->NLSF_MSVQ_Survivors = 3;
psEncC->warping_Q16 = 0;
} else if( Complexity < 3 ) {
psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 );
psEncC->pitchEstimationLPCOrder = 6;
psEncC->shapingLPCOrder = 12;
psEncC->la_shape = 3 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 2;
psEncC->useInterpolatedNLSFs = 0;
psEncC->NLSF_MSVQ_Survivors = 2;
psEncC->warping_Q16 = 0;
} else if( Complexity < 4 ) {
psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 );
psEncC->pitchEstimationLPCOrder = 8;
psEncC->shapingLPCOrder = 14;
psEncC->la_shape = 5 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 2;
psEncC->useInterpolatedNLSFs = 0;
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psEncC->NLSF_MSVQ_Survivors = 4;
psEncC->warping_Q16 = 0;
} else if( Complexity < 6 ) {
psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.74, 16 );
psEncC->pitchEstimationLPCOrder = 10;
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psEncC->shapingLPCOrder = 16;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 2;
psEncC->useInterpolatedNLSFs = 1;
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psEncC->NLSF_MSVQ_Survivors = 6;
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psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
} else if( Complexity < 8 ) {
psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.72, 16 );
psEncC->pitchEstimationLPCOrder = 12;
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psEncC->shapingLPCOrder = 20;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = 3;
psEncC->useInterpolatedNLSFs = 1;
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psEncC->NLSF_MSVQ_Survivors = 8;
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psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
} else {
psEncC->pitchEstimationComplexity = SILK_PE_MAX_COMPLEX;
psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.7, 16 );
psEncC->pitchEstimationLPCOrder = 16;
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psEncC->shapingLPCOrder = 24;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
psEncC->nStatesDelayedDecision = MAX_DEL_DEC_STATES;
psEncC->useInterpolatedNLSFs = 1;
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psEncC->NLSF_MSVQ_Survivors = 16;
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psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
}
/* Do not allow higher pitch estimation LPC order than predict LPC order */
psEncC->pitchEstimationLPCOrder = silk_min_int( psEncC->pitchEstimationLPCOrder, psEncC->predictLPCOrder );
psEncC->shapeWinLength = SUB_FRAME_LENGTH_MS * psEncC->fs_kHz + 2 * psEncC->la_shape;
psEncC->Complexity = Complexity;
Update opus to 1.3.1 and opusfile to 0.11
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celt_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER );
celt_assert( psEncC->shapingLPCOrder <= MAX_SHAPE_LPC_ORDER );
celt_assert( psEncC->nStatesDelayedDecision <= MAX_DEL_DEC_STATES );
celt_assert( psEncC->warping_Q16 <= 32767 );
celt_assert( psEncC->la_shape <= LA_SHAPE_MAX );
celt_assert( psEncC->shapeWinLength <= SHAPE_LPC_WIN_MAX );
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return ret;
}
static OPUS_INLINE opus_int silk_setup_LBRR(
silk_encoder_state *psEncC, /* I/O */
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const silk_EncControlStruct *encControl /* I */
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)
{
Update Opus driver to 1.1.2 And opusfile to 0.7.
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opus_int LBRR_in_previous_packet, ret = SILK_NO_ERROR;
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Update Opus driver to 1.1.2 And opusfile to 0.7.
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LBRR_in_previous_packet = psEncC->LBRR_enabled;
Update opus to 1.3.1 and opusfile to 0.11
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psEncC->LBRR_enabled = encControl->LBRR_coded;
if( psEncC->LBRR_enabled ) {
/* Set gain increase for coding LBRR excitation */
if( LBRR_in_previous_packet == 0 ) {
/* Previous packet did not have LBRR, and was therefore coded at a higher bitrate */
psEncC->LBRR_GainIncreases = 7;
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} else {
Update opus to 1.3.1 and opusfile to 0.11
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psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
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}
}
return ret;
}
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