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* Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
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* Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 3 of the License, or
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* (at your option) any later version.
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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* Additional permission under GNU GPL version 3 section 7:
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* If you modify this program, or any covered work, by linking or
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* combining it with the OpenSSL project's OpenSSL library (or a
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* modified version of that library), containing parts covered by the
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* terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
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* grants you additional permission to convey the resulting work.
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* Corresponding Source for a non-source form of such a combination
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* shall include the source code for the parts of OpenSSL used as well
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* as that of the covered work.
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#include "AudioRtpRecordHandler.h"
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#include "audio/audiolayer.h"
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static const SFLDataFormat initFadeinFactor = 32000;
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AudioRtpRecord::AudioRtpRecord () : _audioCodec (NULL)
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, _hasDynamicPayloadType (false)
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, _micDataConverted (NULL)
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, _micDataEncoded (NULL)
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, _spkrDataDecoded (NULL)
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, _spkrDataConverted (NULL)
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, _audioLayerSampleRate (0)
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, _codecSampleRate (0)
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, _audioLayerFrameSize (0)
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, _micFadeInComplete (false)
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, _spkrFadeInComplete (false)
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, _micAmplFactor (initFadeinFactor)
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, _spkrAmplFactor (initFadeinFactor)
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, _audioProcess (NULL)
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, _noiseSuppress (NULL)
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// _audioLayerFrameSize = manager->getAudioDriver()->getFrameSize(); // in ms
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// _audioLayerSampleRate = manager->getAudioDriver()->getSampleRate();
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_audioLayerFrameSize = Manager::instance().getAudioDriver()->getFrameSize(); // in ms
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_audioLayerSampleRate = Manager::instance().getAudioDriver()->getSampleRate();
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AudioRtpRecord::~AudioRtpRecord()
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if (_micDataConverted)
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delete [] _micDataConverted;
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_micDataConverted = NULL;
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delete [] _micDataEncoded;
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_micDataEncoded = NULL;
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delete [] _spkrDataDecoded;
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_spkrDataDecoded = NULL;
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if (_spkrDataConverted)
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delete [] _spkrDataConverted;
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_spkrDataConverted = NULL;
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delete _audioProcess;
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_audioProcess = NULL;
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if (_noiseSuppress) {
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delete _noiseSuppress;
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_noiseSuppress = NULL;
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AudioRtpRecordHandler::AudioRtpRecordHandler (ManagerImpl *manager, SIPCall *ca) : _audioRtpRecord (), _ca (ca) {}
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AudioRtpRecordHandler::~AudioRtpRecordHandler() {}
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void AudioRtpRecordHandler::setRtpMedia (AudioCodec* audioCodec)
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// Set varios codec info to reduce indirection
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_audioRtpRecord._audioCodec = audioCodec;
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_audioRtpRecord._codecPayloadType = audioCodec->getPayload();
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_audioRtpRecord._codecSampleRate = audioCodec->getClockRate();
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_audioRtpRecord._codecFrameSize = audioCodec->getFrameSize();
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_audioRtpRecord._hasDynamicPayloadType = audioCodec->hasDynamicPayload();
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void AudioRtpRecordHandler::init()
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// init noise reduction process
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// _noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
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// _audioProcess = new AudioProcessing (_noiseSuppress);
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// _audioRtpRecord._noiseSuppress
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void AudioRtpRecordHandler::initBuffers()
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int codecSampleRate = _audioRtpRecord._codecSampleRate;
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// Set sampling rate, main buffer choose the highest one
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Manager::instance().getAudioDriver()->getMainBuffer()->setInternalSamplingRate (codecSampleRate);
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// initialize SampleRate converter using AudioLayer's sampling rate
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// (internal buffers initialized with maximal sampling rate and frame size)
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_audioRtpRecord._converter = new SamplerateConverter (getAudioLayerSampleRate(), getAudioLayerFrameSize());
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// int nbSamplesMax = (int) ( (getCodecSampleRate() * getAudioLayerFrameSize() / 1000)) * 2;
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int nbSamplesMax = (int) ( (getCodecSampleRate() * 44100 / 1000)) * 2;
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_audioRtpRecord._micData = new SFLDataFormat[nbSamplesMax];
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_audioRtpRecord._micDataConverted = new SFLDataFormat[nbSamplesMax];
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_audioRtpRecord._micDataEncoded = new unsigned char[nbSamplesMax * 2];
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_audioRtpRecord._spkrDataConverted = new SFLDataFormat[nbSamplesMax];
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_audioRtpRecord._spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
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Manager::instance().addStream (_ca->getCallId());
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void AudioRtpRecordHandler::initNoiseSuppress()
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NoiseSuppress *noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
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AudioProcessing *processing = new AudioProcessing (noiseSuppress);
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_audioRtpRecord._noiseSuppress = noiseSuppress;
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_audioRtpRecord._audioProcess = processing;
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void AudioRtpRecordHandler::putDtmfEvent (int digit)
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sfl::DtmfEvent *dtmf = new sfl::DtmfEvent();
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dtmf->payload.event = digit;
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dtmf->payload.ebit = false; // end of event bit
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dtmf->payload.rbit = false; // reserved bit
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dtmf->payload.duration = 1; // duration for this event
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dtmf->newevent = true;
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getEventQueue()->push_back (dtmf);
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_debug ("AudioRtpSession: Put Dtmf Event %d", getEventQueue()->size());
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int AudioRtpRecordHandler::processDataEncode (void)
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AudioCodec *audioCodec = _audioRtpRecord._audioCodec;
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AudioLayer *audioLayer = Manager::instance().getAudioDriver();
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SFLDataFormat *micData = _audioRtpRecord._micData;
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unsigned char *micDataEncoded = _audioRtpRecord._micDataEncoded;
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SFLDataFormat *micDataConverted = _audioRtpRecord._micDataConverted;
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int codecFrameSize = audioCodec->getFrameSize();
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int codecSampleRate = audioCodec->getClockRate();
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int mainBufferSampleRate = audioLayer->getMainBuffer()->getInternalSamplingRate();
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// compute codec framesize in ms
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float fixedCodecFramesize = computeCodecFrameSize (codecFrameSize, codecSampleRate);
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// compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
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int bytesToGet = computeNbByteAudioLayer (fixedCodecFramesize);
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// available bytes inside ringbuffer
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int availBytesFromMic = audioLayer->getMainBuffer()->availForGet (_ca->getCallId());
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if (availBytesFromMic < bytesToGet)
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// Get bytes from micRingBuffer to data_from_mic
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int nbSample = audioLayer->getMainBuffer()->getData (micData, bytesToGet, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
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// process mic fade in
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if (!_audioRtpRecord._micFadeInComplete)
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_audioRtpRecord._micFadeInComplete = fadeIn (micData, nbSample, &_audioRtpRecord._micAmplFactor);
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// nb bytes to be sent over RTP
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// test if resampling is required
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if (codecSampleRate != mainBufferSampleRate) {
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int nbSampleUp = nbSample;
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nbSample = _audioRtpRecord._converter->downsampleData (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, nbSampleUp);
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if (Manager::instance().audioPreference.getNoiseReduce())
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_audioRtpRecord._audioProcess->processAudio (micDataConverted, nbSample * sizeof (SFLDataFormat));
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compSize = audioCodec->codecEncode (micDataEncoded, micDataConverted, nbSample * sizeof (SFLDataFormat));
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if (Manager::instance().audioPreference.getNoiseReduce())
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_audioRtpRecord._audioProcess->processAudio (micData, nbSample * sizeof (SFLDataFormat));
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// no resampling required
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compSize = audioCodec->codecEncode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
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void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned int size)
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AudioCodec *audioCodec = _audioRtpRecord._audioCodec;
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AudioLayer *audioLayer = Manager::instance().getAudioDriver();
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int codeFrameSize = audioCodec->getFrameSize();
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int codecSampleRate = audioCodec->getClockRate();
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SFLDataFormat *spkrDataDecoded = _audioRtpRecord._spkrDataConverted;
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SFLDataFormat *spkrDataConverted = _audioRtpRecord._spkrDataDecoded;
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int mainBufferSampleRate = audioLayer->getMainBuffer()->getInternalSamplingRate();
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// Return the size of data in bytes
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int expandedSize = audioCodec->codecDecode (spkrDataDecoded , spkrData , size);
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// buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
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int nbSample = expandedSize / sizeof (SFLDataFormat);
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if (!_audioRtpRecord._spkrFadeInComplete)
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_audioRtpRecord._spkrFadeInComplete = fadeIn (spkrDataDecoded, nbSample, &_audioRtpRecord._micAmplFactor);
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// test if resampling is required
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if (codecSampleRate != mainBufferSampleRate) {
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// Do sample rate conversion
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int nbSampleDown = nbSample;
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nbSample = _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSampleDown);
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// put data in audio layer, size in byte
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audioLayer->getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
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// put data in audio layer, size in byte
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audioLayer->getMainBuffer()->putData (spkrDataDecoded, expandedSize, 100, _ca->getCallId());
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bool AudioRtpRecordHandler::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
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// if factor reach 0, this function should no be called anymore
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// apply amplitude factor;
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audio[size] /= *factor;
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*factor /= FADEIN_STEP_SIZE;
added
removed
sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
