4ede311055
Audio notes (opus codec) A lot of different improvements and bug fixes Thanks to: https://github.com/DrKLO/Telegram/issues/293 https://github.com/DrKLO/Telegram/issues/256 FOSS configuration not ready yet I will move main dev branch to github in next couple commits
118 lines
6.4 KiB
C
118 lines
6.4 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, are permitted provided that the following conditions
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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 Internet Society, IETF or IETF Trust, nor the
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names of specific contributors, may be used to endorse or promote
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products derived from this software without specific prior written
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permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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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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#include "main_FIX.h"
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#include "tuning_parameters.h"
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/* Processing of gains */
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void silk_process_gains_FIX(
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silk_encoder_state_FIX *psEnc, /* I/O Encoder state */
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silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */
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opus_int condCoding /* I The type of conditional coding to use */
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)
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{
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silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
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opus_int k;
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opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10;
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/* Gain reduction when LTP coding gain is high */
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if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
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/*s = -0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */
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s_Q16 = -silk_sigm_Q15( silk_RSHIFT_ROUND( psEncCtrl->LTPredCodGain_Q7 - SILK_FIX_CONST( 12.0, 7 ), 4 ) );
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for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
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psEncCtrl->Gains_Q16[ k ] = silk_SMLAWB( psEncCtrl->Gains_Q16[ k ], psEncCtrl->Gains_Q16[ k ], s_Q16 );
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}
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}
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/* Limit the quantized signal */
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/* InvMaxSqrVal = pow( 2.0f, 0.33f * ( 21.0f - SNR_dB ) ) / subfr_length; */
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InvMaxSqrVal_Q16 = silk_DIV32_16( silk_log2lin(
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silk_SMULWB( SILK_FIX_CONST( 21 + 16 / 0.33, 7 ) - psEnc->sCmn.SNR_dB_Q7, SILK_FIX_CONST( 0.33, 16 ) ) ), psEnc->sCmn.subfr_length );
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for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
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/* Soft limit on ratio residual energy and squared gains */
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ResNrg = psEncCtrl->ResNrg[ k ];
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ResNrgPart = silk_SMULWW( ResNrg, InvMaxSqrVal_Q16 );
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if( psEncCtrl->ResNrgQ[ k ] > 0 ) {
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ResNrgPart = silk_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] );
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} else {
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if( ResNrgPart >= silk_RSHIFT( silk_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) {
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ResNrgPart = silk_int32_MAX;
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} else {
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ResNrgPart = silk_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] );
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}
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}
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gain = psEncCtrl->Gains_Q16[ k ];
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gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) );
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if( gain_squared < silk_int16_MAX ) {
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/* recalculate with higher precision */
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gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain );
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silk_assert( gain_squared > 0 );
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gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */
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gain = silk_min( gain, silk_int32_MAX >> 8 );
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psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 8 ); /* Q16 */
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} else {
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gain = silk_SQRT_APPROX( gain_squared ); /* Q0 */
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gain = silk_min( gain, silk_int32_MAX >> 16 );
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psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 16 ); /* Q16 */
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}
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}
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/* Save unquantized gains and gain Index */
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silk_memcpy( psEncCtrl->GainsUnq_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) );
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psEncCtrl->lastGainIndexPrev = psShapeSt->LastGainIndex;
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/* Quantize gains */
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silk_gains_quant( psEnc->sCmn.indices.GainsIndices, psEncCtrl->Gains_Q16,
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&psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr );
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/* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */
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if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
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if( psEncCtrl->LTPredCodGain_Q7 + silk_RSHIFT( psEnc->sCmn.input_tilt_Q15, 8 ) > SILK_FIX_CONST( 1.0, 7 ) ) {
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psEnc->sCmn.indices.quantOffsetType = 0;
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} else {
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psEnc->sCmn.indices.quantOffsetType = 1;
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}
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}
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/* Quantizer boundary adjustment */
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quant_offset_Q10 = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ];
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psEncCtrl->Lambda_Q10 = SILK_FIX_CONST( LAMBDA_OFFSET, 10 )
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+ silk_SMULBB( SILK_FIX_CONST( LAMBDA_DELAYED_DECISIONS, 10 ), psEnc->sCmn.nStatesDelayedDecision )
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+ silk_SMULWB( SILK_FIX_CONST( LAMBDA_SPEECH_ACT, 18 ), psEnc->sCmn.speech_activity_Q8 )
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+ silk_SMULWB( SILK_FIX_CONST( LAMBDA_INPUT_QUALITY, 12 ), psEncCtrl->input_quality_Q14 )
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+ silk_SMULWB( SILK_FIX_CONST( LAMBDA_CODING_QUALITY, 12 ), psEncCtrl->coding_quality_Q14 )
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+ silk_SMULWB( SILK_FIX_CONST( LAMBDA_QUANT_OFFSET, 16 ), quant_offset_Q10 );
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silk_assert( psEncCtrl->Lambda_Q10 > 0 );
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silk_assert( psEncCtrl->Lambda_Q10 < SILK_FIX_CONST( 2, 10 ) );
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}
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