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ac76521150
This is the IETF Opus audio codec reference implementation. The source was copied into the tree using the included update.sh script, from a checkout of the v0.9.9 git tag, which corresponds to the source code published in https://tools.ietf.org/id/draft-ietf-codec-opus-11.txt
412 lines
20 KiB
C
412 lines
20 KiB
C
/***********************************************************************
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Copyright (c) 2006-2011, Skype Limited. All rights reserved.
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Redistribution and use in source and binary forms, with or without
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modification, (subject to the limitations in the disclaimer below)
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are permitted provided that the following conditions are met:
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- Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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- Neither the name of Skype Limited, nor the names of specific
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contributors, may be used to endorse or promote products derived from
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this software without specific prior written permission.
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NO EXPRESS OR IMPLIED LICENSES TO ANY PARTY'S PATENT RIGHTS ARE GRANTED
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BY THIS LICENSE. THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND
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CONTRIBUTORS ''AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING,
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BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
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FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
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COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
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INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
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NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
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USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
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ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
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OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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***********************************************************************/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#ifdef FIXED_POINT
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#include "main_FIX.h"
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#define silk_encoder_state_Fxx silk_encoder_state_FIX
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#else
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#include "main_FLP.h"
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#define silk_encoder_state_Fxx silk_encoder_state_FLP
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#endif
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#include "tuning_parameters.h"
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#include "pitch_est_defines.h"
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opus_int silk_setup_resamplers(
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silk_encoder_state_Fxx *psEnc, /* I/O */
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opus_int fs_kHz /* I */
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);
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opus_int silk_setup_fs(
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silk_encoder_state_Fxx *psEnc, /* I/O */
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opus_int fs_kHz, /* I */
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opus_int PacketSize_ms /* I */
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);
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opus_int silk_setup_complexity(
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silk_encoder_state *psEncC, /* I/O */
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opus_int Complexity /* I */
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);
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static inline opus_int silk_setup_LBRR(
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silk_encoder_state *psEncC, /* I/O */
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const opus_int32 TargetRate_bps /* I */
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);
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/* Control encoder */
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opus_int silk_control_encoder(
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silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */
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silk_EncControlStruct *encControl, /* I Control structure */
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const opus_int32 TargetRate_bps, /* I Target max bitrate (bps) */
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const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */
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const opus_int channelNb, /* I Channel number */
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const opus_int force_fs_kHz
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)
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{
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opus_int fs_kHz, ret = 0;
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psEnc->sCmn.useDTX = encControl->useDTX;
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psEnc->sCmn.useCBR = encControl->useCBR;
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psEnc->sCmn.API_fs_Hz = encControl->API_sampleRate;
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psEnc->sCmn.maxInternal_fs_Hz = encControl->maxInternalSampleRate;
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psEnc->sCmn.minInternal_fs_Hz = encControl->minInternalSampleRate;
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psEnc->sCmn.desiredInternal_fs_Hz = encControl->desiredInternalSampleRate;
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psEnc->sCmn.useInBandFEC = encControl->useInBandFEC;
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psEnc->sCmn.nChannelsAPI = encControl->nChannelsAPI;
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psEnc->sCmn.nChannelsInternal = encControl->nChannelsInternal;
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psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch;
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psEnc->sCmn.channelNb = channelNb;
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if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) {
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if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) {
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/* Change in API sampling rate in the middle of encoding a packet */
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ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz );
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}
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return ret;
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}
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/* Beyond this point we know that there are no previously coded frames in the payload buffer */
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/********************************************/
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/* Determine internal sampling rate */
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/********************************************/
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fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn, encControl );
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if( force_fs_kHz ) {
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fs_kHz = force_fs_kHz;
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}
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/********************************************/
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/* Prepare resampler and buffered data */
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/********************************************/
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ret += silk_setup_resamplers( psEnc, fs_kHz );
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/********************************************/
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/* Set internal sampling frequency */
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/********************************************/
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ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms );
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/********************************************/
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/* Set encoding complexity */
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/********************************************/
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ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity );
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/********************************************/
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/* Set packet loss rate measured by farend */
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/********************************************/
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psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage;
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/********************************************/
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/* Set LBRR usage */
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/********************************************/
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ret += silk_setup_LBRR( &psEnc->sCmn, TargetRate_bps );
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psEnc->sCmn.controlled_since_last_payload = 1;
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return ret;
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}
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opus_int silk_setup_resamplers(
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silk_encoder_state_Fxx *psEnc, /* I/O */
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opus_int fs_kHz /* I */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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opus_int32 nSamples_temp;
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if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz )
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{
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if( psEnc->sCmn.fs_kHz == 0 ) {
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/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
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ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000, 1 );
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} else {
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/* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */
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opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ];
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silk_resampler_state_struct temp_resampler_state;
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#ifdef FIXED_POINT
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opus_int16 *x_bufFIX = psEnc->x_buf;
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#else
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opus_int16 x_bufFIX[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];
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#endif
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nSamples_temp = silk_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz;
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#ifndef FIXED_POINT
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silk_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp );
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#endif
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/* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
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ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
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/* Temporary resampling of x_buf data to API_fs_Hz */
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ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp );
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/* Calculate number of samples that has been temporarily upsampled */
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nSamples_temp = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) );
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/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
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ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 );
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/* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */
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ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp );
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#ifndef FIXED_POINT
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silk_short2float_array( psEnc->x_buf, x_bufFIX, ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * fs_kHz );
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#endif
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}
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}
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psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
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return ret;
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}
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opus_int silk_setup_fs(
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silk_encoder_state_Fxx *psEnc, /* I/O */
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opus_int fs_kHz, /* I */
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opus_int PacketSize_ms /* I */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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/* Set packet size */
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if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) {
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if( ( PacketSize_ms != 10 ) &&
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( PacketSize_ms != 20 ) &&
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( PacketSize_ms != 40 ) &&
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( PacketSize_ms != 60 ) ) {
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ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED;
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}
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if( PacketSize_ms <= 10 ) {
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psEnc->sCmn.nFramesPerPacket = 1;
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psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1;
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psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz );
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psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
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if( psEnc->sCmn.fs_kHz == 8 ) {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
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} else {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
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}
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} else {
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psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
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psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
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psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz );
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psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
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if( psEnc->sCmn.fs_kHz == 8 ) {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
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} else {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
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}
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}
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psEnc->sCmn.PacketSize_ms = PacketSize_ms;
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psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
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}
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/* Set internal sampling frequency */
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silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
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silk_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
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if( psEnc->sCmn.fs_kHz != fs_kHz ) {
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/* reset part of the state */
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silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) );
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silk_memset( &psEnc->sPrefilt, 0, sizeof( psEnc->sPrefilt ) );
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silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) );
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silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
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silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) );
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psEnc->sCmn.inputBufIx = 0;
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psEnc->sCmn.nFramesEncoded = 0;
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psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
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/* Initialize non-zero parameters */
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psEnc->sCmn.prevLag = 100;
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psEnc->sCmn.first_frame_after_reset = 1;
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psEnc->sPrefilt.lagPrev = 100;
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psEnc->sShape.LastGainIndex = 10;
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psEnc->sCmn.sNSQ.lagPrev = 100;
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psEnc->sCmn.sNSQ.prev_gain_Q16 = 65536;
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psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
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psEnc->sCmn.fs_kHz = fs_kHz;
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if( psEnc->sCmn.fs_kHz == 8 ) {
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if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
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} else {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
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}
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} else {
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if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
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} else {
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psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
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}
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}
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if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) {
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psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER;
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psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB;
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} else {
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psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER;
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psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB;
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}
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psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz;
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psEnc->sCmn.frame_length = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
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psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
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psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz );
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psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz );
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if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
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psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
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} else {
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psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
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}
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if( psEnc->sCmn.fs_kHz == 16 ) {
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psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_WB, 9 );
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psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
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} else if( psEnc->sCmn.fs_kHz == 12 ) {
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psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_MB, 9 );
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psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
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} else {
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psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_NB, 9 );
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psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
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}
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}
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/* Check that settings are valid */
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silk_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
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return ret;
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}
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opus_int silk_setup_complexity(
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silk_encoder_state *psEncC, /* I/O */
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opus_int Complexity /* I */
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)
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{
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opus_int ret = 0;
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/* Set encoding complexity */
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silk_assert( Complexity >= 0 && Complexity <= 10 );
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if( Complexity < 2 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX;
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psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 );
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psEncC->pitchEstimationLPCOrder = 6;
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psEncC->shapingLPCOrder = 8;
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psEncC->la_shape = 3 * psEncC->fs_kHz;
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psEncC->nStatesDelayedDecision = 1;
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psEncC->useInterpolatedNLSFs = 0;
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psEncC->LTPQuantLowComplexity = 1;
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psEncC->NLSF_MSVQ_Survivors = 2;
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psEncC->warping_Q16 = 0;
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} else if( Complexity < 4 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
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psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 );
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psEncC->pitchEstimationLPCOrder = 8;
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psEncC->shapingLPCOrder = 10;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
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psEncC->nStatesDelayedDecision = 1;
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psEncC->useInterpolatedNLSFs = 0;
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psEncC->LTPQuantLowComplexity = 0;
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psEncC->NLSF_MSVQ_Survivors = 4;
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psEncC->warping_Q16 = 0;
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} else if( Complexity < 6 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
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psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.74, 16 );
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psEncC->pitchEstimationLPCOrder = 10;
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psEncC->shapingLPCOrder = 12;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
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psEncC->nStatesDelayedDecision = 2;
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psEncC->useInterpolatedNLSFs = 1;
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psEncC->LTPQuantLowComplexity = 0;
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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 );
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} else if( Complexity < 8 ) {
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psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
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psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.72, 16 );
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psEncC->pitchEstimationLPCOrder = 12;
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psEncC->shapingLPCOrder = 14;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
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psEncC->nStatesDelayedDecision = 3;
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psEncC->useInterpolatedNLSFs = 1;
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psEncC->LTPQuantLowComplexity = 0;
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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 );
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} else {
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psEncC->pitchEstimationComplexity = SILK_PE_MAX_COMPLEX;
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psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.7, 16 );
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psEncC->pitchEstimationLPCOrder = 16;
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psEncC->shapingLPCOrder = 16;
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psEncC->la_shape = 5 * psEncC->fs_kHz;
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psEncC->nStatesDelayedDecision = MAX_DEL_DEC_STATES;
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psEncC->useInterpolatedNLSFs = 1;
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psEncC->LTPQuantLowComplexity = 0;
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psEncC->NLSF_MSVQ_Survivors = 32;
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psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
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}
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/* Do not allow higher pitch estimation LPC order than predict LPC order */
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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;
|
|
|
|
silk_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER );
|
|
silk_assert( psEncC->shapingLPCOrder <= MAX_SHAPE_LPC_ORDER );
|
|
silk_assert( psEncC->nStatesDelayedDecision <= MAX_DEL_DEC_STATES );
|
|
silk_assert( psEncC->warping_Q16 <= 32767 );
|
|
silk_assert( psEncC->la_shape <= LA_SHAPE_MAX );
|
|
silk_assert( psEncC->shapeWinLength <= SHAPE_LPC_WIN_MAX );
|
|
silk_assert( psEncC->NLSF_MSVQ_Survivors <= NLSF_VQ_MAX_SURVIVORS );
|
|
|
|
return ret;
|
|
}
|
|
|
|
static inline opus_int silk_setup_LBRR(
|
|
silk_encoder_state *psEncC, /* I/O */
|
|
const opus_int32 TargetRate_bps /* I */
|
|
)
|
|
{
|
|
opus_int ret = SILK_NO_ERROR;
|
|
opus_int32 LBRR_rate_thres_bps;
|
|
|
|
psEncC->LBRR_enabled = 0;
|
|
if( psEncC->useInBandFEC && psEncC->PacketLoss_perc > 0 ) {
|
|
if( psEncC->fs_kHz == 8 ) {
|
|
LBRR_rate_thres_bps = LBRR_NB_MIN_RATE_BPS;
|
|
} else if( psEncC->fs_kHz == 12 ) {
|
|
LBRR_rate_thres_bps = LBRR_MB_MIN_RATE_BPS;
|
|
} else {
|
|
LBRR_rate_thres_bps = LBRR_WB_MIN_RATE_BPS;
|
|
}
|
|
LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, 125 - silk_min( psEncC->PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) );
|
|
|
|
if( TargetRate_bps > LBRR_rate_thres_bps ) {
|
|
/* Set gain increase for coding LBRR excitation */
|
|
psEncC->LBRR_enabled = 1;
|
|
psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|