Evgeny
4fde0b0325
- Add lib/libopus/ (opus v1.6.1 embedded source: CELT 18 .c, SILK 77 .c, SILK float 28 .c, OPUS core 14 .c) - Add lib/opus_codec.h/.c (wrapper: encoder, decoder, raw PCM int16) - Add tests/test_opus_codec.c (7 tests: silence, sine, compression, PLC, null, defaults, multi-frame) - chatgui: AudioRecorder (miniaudio capture, 48kHz mono) - chatgui: VoiceEncoder (PCM→Opus file with custom container) - chatgui: VoicePlayer (Opus→PCM playback via SoundManager) - chatgui: PTT button in InputBar, voice message send/receive in MessageList - chatgui: AudioDeviceSettingsPage (device selection, persisted in ui_state) - chatgui: MessageDelegate voice message rendering (play button + duration) - Update autotools: configure.ac, Makefile.am, src/Makefile.am, tests/Makefile.am - Update CMake: chatgui CMakeLists.txt, libutun CMakeLists.txt (link libopus_internal.a)
104 lines
5.0 KiB
C
104 lines
5.0 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_FLP.h"
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#include "tuning_parameters.h"
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/* Processing of gains */
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void silk_process_gains_FLP(
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silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */
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silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */
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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_FLP *psShapeSt = &psEnc->sShape;
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opus_int k;
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opus_int32 pGains_Q16[ MAX_NB_SUBFR ];
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silk_float s, InvMaxSqrVal, gain, quant_offset;
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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 = 1.0f - 0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) );
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for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
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psEncCtrl->Gains[ k ] *= s;
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}
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}
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/* Limit the quantized signal */
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InvMaxSqrVal = ( silk_float )( pow( 2.0f, 0.33f * ( 21.0f - psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) ) ) / 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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gain = psEncCtrl->Gains[ k ];
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gain = ( silk_float )sqrt( gain * gain + psEncCtrl->ResNrg[ k ] * InvMaxSqrVal );
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psEncCtrl->Gains[ k ] = silk_min_float( gain, 32767.0f );
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}
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/* Prepare gains for noise shaping quantization */
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for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
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pGains_Q16[ k ] = (opus_int32)( psEncCtrl->Gains[ k ] * 65536.0f );
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}
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/* Save unquantized gains and gain Index */
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silk_memcpy( psEncCtrl->GainsUnq_Q16, pGains_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, pGains_Q16,
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&psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr );
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/* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */
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for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
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psEncCtrl->Gains[ k ] = pGains_Q16[ k ] / 65536.0f;
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}
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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 + psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f ) > 1.0f ) {
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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 = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ] / 1024.0f;
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psEncCtrl->Lambda = LAMBDA_OFFSET
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+ LAMBDA_DELAYED_DECISIONS * psEnc->sCmn.nStatesDelayedDecision
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+ LAMBDA_SPEECH_ACT * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f )
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+ LAMBDA_INPUT_QUALITY * psEncCtrl->input_quality
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+ LAMBDA_CODING_QUALITY * psEncCtrl->coding_quality
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+ LAMBDA_QUANT_OFFSET * quant_offset;
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silk_assert( psEncCtrl->Lambda > 0.0f );
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silk_assert( psEncCtrl->Lambda < 2.0f );
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}
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