diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index 41f572972ad7f06aa09f9aeacb2d7ca50bdc0948..d4e649ca77b6c137ef5a5f2674bf0ee0fbb9535b 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
* Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index b5183d67b2a509957423d08a41549e6af32ae672..a88cfe2a3a128637b56a36d22535f854f4ed68c1 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
* Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..d783b894e0f7f9850635c3454ad688ec896a7516
--- /dev/null
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -0,0 +1,437 @@
+/*
+ * Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Additional permission under GNU GPL version 3 section 7:
+ *
+ * If you modify this program, or any covered work, by linking or
+ * combining it with the OpenSSL project's OpenSSL library (or a
+ * modified version of that library), containing parts covered by the
+ * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ * grants you additional permission to convey the resulting work.
+ * Corresponding Source for a non-source form of such a combination
+ * shall include the source code for the parts of OpenSSL used as well
+ * as that of the covered work.
+ */
+
+#include "AudioRtpRecordHandler.h"
+
+namespace sfl
+{
+
+static const SFLDataFormat initFadeinFactor = 32000;
+
+AudioRtpRecord::AudioRtpRecord() : _audioCodec(NULL)
+ , _hasDynamicPayloadType(false)
+ , _micData(NULL)
+ , _micDataConverted(NULL)
+ , _micDataEncoded(NULL)
+ , _spkrDataDecoded(NULL)
+ , _spkrDataConverted(NULL)
+ , _converter(NULL)
+ , _audioLayerSampleRate(0)
+ , _codecSampleRate(0)
+ , _audioLayerFrameSize(0)
+ , _codecFrameSize(0)
+ , _converterSamplingRate(0)
+ , _micFadeInComplete(false)
+ , _spkrFadeInComplete(false)
+ , _micAmplFactor(initFadeinFactor)
+ , _spkrAmplFactor(initFadeinFactor)
+ , _audioProcess(NULL)
+ , _noiseSuppress(NULL)
+{
+ _audioLayerFrameSize = _manager->getAudioDriver()->getFrameSize(); // in ms
+ _audioLayerSampleRate = _manager->getAudioDriver()->getSampleRate();
+}
+
+
+AudioRtpRecord::~AudioRtpRecord()
+{
+ if (_micData)
+ delete [] _micData;
+ _micData = NULL;
+
+ if (_micDataConverted)
+ delete [] _micDataConverted;
+ _micDataConverted = NULL;
+
+ if (_micDataEncoded)
+ delete [] _micDataEncoded;
+ _micDataEncoded = NULL;
+
+ if (_spkrDataDecoded)
+ delete [] _spkrDataDecoded;
+ _spkrDataDecoded = NULL;
+
+ if (_spkrDataConverted)
+ delete [] _spkrDataConverted;
+ _spkrDataConverted = NULL;
+
+
+ if(_converter)
+ delete _converter;
+ _converter = NULL;
+
+ if (_audioCodec) {
+ delete _audioCodec;
+ _audioCodec = NULL;
+ }
+
+ if (_audioProcess) {
+ delete _audioProcess;
+ _audioProcess = NULL;
+ }
+
+ if (_noiseSuppress) {
+ delete _noiseSuppress;
+ _noiseSuppress = NULL;
+ }
+}
+
+AudioCodec *AudioRtpRecord::getAudiocodec() const
+{
+ return _audioCodec;
+}
+
+int AudioRtpRecord::getCodecPayloadType() const
+{
+ return _codecPayloadType;
+}
+
+bool AudioRtpRecord::getHasDynamicPayload() const
+{
+ return _hasDynamicPayloadType;
+}
+
+int AudioRtpRecord::getAudioLayerFrameSize() const
+{
+ return _audioLayerFrameSize;
+}
+
+int AudioRtpRecord::getAudioLayerSampleRate() const
+{
+ return _audioLayerSampleRate;
+}
+
+int AudioRtpRecord::getCodecFrameSize() const
+{
+ return _codecFrameSize;
+}
+
+int AudioRtpRecord::getCodecSampleRate() const
+{
+ return _codecSampleRate;
+}
+
+SamplerateConverter *AudioRtpRecord::getConverter() const
+{
+ return _converter;
+}
+
+int AudioRtpRecord::getConverterSamplingRate() const
+{
+ return _converterSamplingRate;
+}
+
+EventQueue *AudioRtpRecord::getEventQueue() const
+{
+ return &_eventQueue;
+}
+
+int AudioRtpRecord::getEventQueueSize() const
+{
+ return _eventQueue.size();
+}
+
+unsigned char *getEncodedData() const
+{
+ return _micDataEncoded;
+}
+
+void AudioRtpRecord::setAudiocodec(AudioCodec *audiocodec)
+{
+ this->_audiocodec = audiocodec;
+}
+
+void AudioRtpRecord::setCodecPayloadType(int codecPayloadType)
+{
+ this->codecPayloadType = codecPayloadType;
+}
+
+bool AudioRtpRecord::setHasDynamicPayload(bool hasDynamicPayload)
+{
+ _hasDynamicPayloadType = hasDynamicPayload;
+}
+
+void AudioRtpRecord::setAudioLayerFrameSize(int _audioLayerFrameSize)
+{
+ this->_audioLayerFrameSize = _audioLayerFrameSize;
+}
+
+void AudioRtpRecord::setAudioLayerSampleRate(int _audioLayerSampleRate)
+{
+ this->_audioLayerSampleRate = _audioLayerSampleRate;
+}
+
+void AudioRtpRecord::setCodecFrameSize(int _codecFrameSize)
+{
+ this->_codecFrameSize = _codecFrameSize;
+}
+
+void AudioRtpRecord::setCodecSampleRate(int _codecSampleRate)
+{
+ this->_codecSampleRate = _codecSampleRate;
+}
+
+void AudioRtpRecord::setConverter(SamplerateConverter *_converter)
+{
+ this->_converter = _converter;
+}
+
+void AudioRtpRecord::setConverterSamplingRate(int _converterSamplingRate)
+{
+ this->_converterSamplingRate = _converterSamplingRate;
+}
+
+void AudioRtpRecord::setEventQueue(EventQueue _eventQueue)
+{
+ this->_eventQueue = _eventQueue;
+}
+
+
+AudioRtpRecordHandler::AudioRtpRecordHandler() {
+ // TODO Auto-generated constructor stub
+
+}
+
+
+AudioRtpRecordHandler::~AudioRtpRecordHandler() {
+ // TODO Auto-generated destructor stub
+}
+
+void AudioRtpRecord::SetRtpMedia(AudioCodec* audioCodec) {
+
+ // Set varios codec info to reduce indirection
+ _audioRtpRecord.setAudioCodec(audioCodec);
+ _audioRtpRecord.setCodecPayloadType(audioCodec->getPayloadType());
+ _audioRtpRecord.setCodecSampleRate(audioCodec->getClockRate());
+ _audioRtpRecord.setCodecFrameSize(audioCodec->getFrameSize());
+ _audioRtpRecord.setHasDynamicPayload(audioCodec->hasDynamicPayload());
+}
+
+
+void AudioRtpRecord::init()
+{
+ // init noise reduction process
+ // _noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
+ // _audioProcess = new AudioProcessing (_noiseSuppress);
+
+ // _audioRtpRecord._noiseSuppress
+ // _audioRtpRecord.
+}
+
+void AudioRtpRecord::initBuffers()
+{
+ // Set sampling rate, main buffer choose the highest one
+ _manager->getAudioDriver()->getMainBuffer()->setInternalSamplingRate (_codecSampleRate);
+
+ // may be different than one already set
+ _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
+ // initialize SampleRate converter using AudioLayer's sampling rate
+ // (internal buffers initialized with maximal sampling rate and frame size)
+ _converter = new SamplerateConverter (_layerSampleRate, _layerFrameSize);
+
+ int nbSamplesMax = (int) (_codecSampleRate * _layerFrameSize /1000) *2;
+ _micData = new SFLDataFormat[nbSamplesMax];
+ _micDataConverted = new SFLDataFormat[nbSamplesMax];
+ _micDataEncoded = new unsigned char[nbSamplesMax*2];
+ _spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ _spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+
+
+ memset (_micData, 0, nbSamplesMax*sizeof (SFLDataFormat));
+ memset (_micDataConverted, 0, nbSamplesMax*sizeof (SFLDataFormat));
+ memset (_micDataEncoded, 0, nbSamplesMax*2);
+ memset (_spkrDataConverted, 0, nbSamplesMax*sizeof (SFLDataFormat));
+ memset (_spkrDataDecoded, 0, nbSamplesMax*sizeof (SFLDataFormat));
+
+ _manager->addStream (_ca->getCallId());
+}
+
+
+void AudioRtpRecordHandler::putDtmfEvent (int digit)
+{
+
+ sfl::DtmfEvent *dtmf = new sfl::DtmfEvent();
+
+ dtmf->payload.event = digit;
+ dtmf->payload.ebit = false; // end of event bit
+ dtmf->payload.rbit = false; // reserved bit
+ dtmf->payload.duration = 1; // duration for this event
+ dtmf->newevent = true;
+ dtmf->length = 1000;
+
+ _eventQueue.push_back (dtmf);
+
+ _debug ("AudioRtpSession: Put Dtmf Event %d", _eventQueue.size());
+}
+
+int AudioRtpRecordHandler::processDataEncode (void)
+{
+ assert (_audiocodec);
+ assert (_audiolayer);
+
+ int _mainBufferSampleRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
+ // compute codec framesize in ms
+ float fixed_codec_framesize = computeCodecFrameSize (_audiocodec->getFrameSize(), _audiocodec->getClockRate());
+
+ // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
+ int maxBytesToGet = computeNbByteAudioLayer (fixed_codec_framesize);
+
+ // available bytes inside ringbuffer
+ int availBytesFromMic = _manager->getAudioDriver()->getMainBuffer()->availForGet (_ca->getCallId());
+
+ // set available byte to maxByteToGet
+ int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+
+ if (bytesAvail == 0) {
+ memset (_micDataEncoded, 0, sizeof (SFLDataFormat));
+ return _audiocodec->getFrameSize();
+ }
+
+ // Get bytes from micRingBuffer to data_from_mic
+ int nbSample = _manager->getAudioDriver()->getMainBuffer()->getData (_micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+
+ if (!_micFadeInComplete)
+ _micFadeInComplete = fadeIn (_micData, nbSample, &_micAmplFactor);
+
+ if (nbSample == 0)
+ return nbSample;
+
+ // nb bytes to be sent over RTP
+ int compSize = 0;
+
+ // test if resampling is required
+ if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
+ int nb_sample_up = nbSample;
+
+ _nSamplesMic = nbSample;
+
+ nbSample = _converter->downsampleData (_micData , _micDataConverted , _audiocodec->getClockRate(), _mainBufferSampleRate, nb_sample_up);
+
+ if (_manager->audioPreference.getNoiseReduce())
+ _audioProcess->processAudio (_micDataConverted, nbSample*sizeof (SFLDataFormat));
+
+ compSize = _audiocodec->codecEncode (_micDataEncoded, _micDataConverted, nbSample*sizeof (SFLDataFormat));
+
+ } else {
+
+ _nSamplesMic = nbSample;
+
+ if (_manager->audioPreference.getNoiseReduce())
+ _audioProcess->processAudio (_micData, nbSample*sizeof (SFLDataFormat));
+
+ // no resampling required
+ compSize = _audiocodec->codecEncode (_micDataEncoded, _micData, nbSample*sizeof (SFLDataFormat));
+
+ }
+
+ return compSize;
+}
+
+void AudioRtpRecordHandler::processDataDecode (unsigned char * spkrData, unsigned int size)
+{
+ if (_audiocodec != NULL) {
+
+
+ int _mainBufferSampleRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
+ // Return the size of data in bytes
+ int expandedSize = _audiocodec->codecDecode (_spkrDataDecoded , spkrData , size);
+
+ // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
+ int nbSample = expandedSize / sizeof (SFLDataFormat);
+
+ if (!_spkrFadeInComplete)
+ _spkrFadeInComplete = fadeIn (_spkrDataDecoded, nbSample, &_spkrAmplFactor);
+
+ // test if resampling is required
+ if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
+
+ // Do sample rate conversion
+ int nb_sample_down = nbSample;
+
+ nbSample = _converter->upsampleData (_spkrDataDecoded, _spkrDataConverted, _codecSampleRate, _mainBufferSampleRate, nb_sample_down);
+
+ // Store the number of samples for recording
+ _nSamplesSpkr = nbSample;
+
+ // put data in audio layer, size in byte
+ _manager->getAudioDriver()->getMainBuffer()->putData (_spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
+
+
+ } else {
+ // Store the number of samples for recording
+ _nSamplesSpkr = nbSample;
+
+ // put data in audio layer, size in byte
+ _manager->getAudioDriver()->getMainBuffer()->putData (_spkrDataDecoded, expandedSize, 100, _ca->getCallId());
+ }
+
+ // Notify (with a beep) an incoming call when there is already a call
+ if (_manager->incomingCallWaiting() > 0) {
+ _countNotificationTime += _time->getSecond();
+ int countTimeModulo = _countNotificationTime % 5000;
+
+ // _debug("countNotificationTime: %d\n", countNotificationTime);
+ // _debug("countTimeModulo: %d\n", countTimeModulo);
+ if ( (countTimeModulo - _countNotificationTime) < 0) {
+ _manager->notificationIncomingCall();
+ }
+
+ _countNotificationTime = countTimeModulo;
+ }
+
+ }
+}
+
+
+bool AudioRtpRecordHandler::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
+{
+
+ // apply amplitude factor;
+ while (size) {
+ size--;
+ audio[size] /= *factor;
+ }
+
+ // decrease factor
+ *factor /= FADEIN_STEP_SIZE;
+
+ // if factor reach 0, thsi function should no be called anymore
+ if (*factor == 0)
+ return true;
+
+ return false;
+}
+
+
+
+}
+
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
new file mode 100644
index 0000000000000000000000000000000000000000..76d63ccce3cd99a76131eb60ce5099005c64e85d
--- /dev/null
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -0,0 +1,324 @@
+/*
+ * Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Additional permission under GNU GPL version 3 section 7:
+ *
+ * If you modify this program, or any covered work, by linking or
+ * combining it with the OpenSSL project's OpenSSL library (or a
+ * modified version of that library), containing parts covered by the
+ * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ * grants you additional permission to convey the resulting work.
+ * Corresponding Source for a non-source form of such a combination
+ * shall include the source code for the parts of OpenSSL used as well
+ * as that of the covered work.
+ */
+
+#ifndef AUDIORTPRECORDHANDLER_H_
+#define AUDIORTPRECORDHANDLER_H_
+
+#include "audio/codecs/audiocodec.h"
+#include "audio/samplerateconverter.h"
+#include "audio/audioprocessing.h"
+#include "audio/noisesuppress.h"
+
+namespace sfl
+{
+
+typedef struct DtmfEvent {
+ ost::RTPPacket::RFC2833Payload payload;
+ int length;
+ bool newevent;
+} DtmfEvent;
+
+typedef list<DtmfEvent *> EventQueue;
+
+/**
+ * Class meant to store internal data in order to encode/decode,
+ * resample, process, and packetize audio streams. This class should not be
+ * handled directly. Use AudioRtpRecorrdHandeler
+ */
+class AudioRtpRecord {
+public:
+ AudioRtpRecord();
+ virtual ~AudioRtpRecord();
+
+ /**
+ * Return a pointer to the current codec instance
+ */
+ inline AudioCodec *getAudioCodec() const;
+
+ /**
+ * Get current payload type (static or dynamic)
+ */
+ int getCodecPayloadType() const;
+
+ /**
+ * Return wether or not this codec has dynamic payload to be
+ * negotiated during SDP session
+ */
+ bool getHasDynamicPayload() const;
+
+ /**
+ * Get current audio layer frame size
+ */
+ int getAudioLayerFrameSize() const;
+
+ /**
+ * Get current audio layer sampling rate
+ */
+ int getAudioLayerSampleRate() const;
+
+ /**
+ * Get current codec frame size
+ */
+ int getCodecFrameSize() const;
+
+ /**
+ * Return current sample rate
+ */
+ int getCodecSampleRate() const;
+
+ /**
+ * Get the sampling rate converter for this session
+ */
+ SamplerateConverter *getConverter() const;
+
+ /**
+ * Get sampling rate converter's sampling rate
+ */
+ inline int getConverterSamplingRate() const;
+
+ /**
+ * Return a poiner to the DTMF event queue
+ */
+ EventQueue *getEventQueue() const;
+
+ /**
+ * Return the number of DTMF event waiting in the queue
+ */
+ int getEventQueueSize() const;
+
+ unsigned char *getEncodedData() const;
+
+ /**
+ * Return a pointer to the current codec instance
+ */
+ void setAudioCodec(AudioCodec *audioCodec);
+
+ /**
+ * Set current codec payload (static or dynamic)
+ */
+ void setCodecPayloadType(int codecPayloadType);
+
+ /**
+ * Set audio layer frame size
+ */
+ void setAudioLayerFrameSize(int _audioLayerFrameSize);
+
+ /**
+ * Set current audio layer sampling rate
+ * used to process sampling rate conversion
+ */
+ void setAudioLayerSampleRate(int _audioLayerSampleRate);
+
+ /**
+ * Set codec frame size used to compute sampling rate conversion
+ */
+ void setCodecFrameSize(int _codecFrameSize);
+
+ /**
+ * Set codec sampling rate used to compute sampling rate conversion
+ */
+ void setCodecSampleRate(int _codecSampleRate);
+
+ /**
+ * Set sampling rate converter for this session
+ */
+ void setConverter(SamplerateConverter *_converter);
+
+ /**
+ * Set converter sampling rate used to compute conversion buffer size
+ */
+ void setConverterSamplingRate(int _converterSamplingRate);
+
+private:
+ /**
+ * Pointer to the session's codec
+ */
+ AudioCodec * _audioCodec;
+
+ /**
+ * Codec payload type
+ */
+ int _codecPayloadType;
+
+ /**
+ * Dynamic payload are negotiated during sdp session while static payload
+ * are predefined numbers identifying the codec
+ */
+ bool _hasDynamicPayloadType;
+
+ /**
+ * Mic-data related buffers
+ */
+ SFLDataFormat* _micData;
+ SFLDataFormat* _micDataConverted;
+ unsigned char* _micDataEncoded;
+
+ /**
+ * Speaker-data related buffers
+ */
+ SFLDataFormat* _spkrDataDecoded;
+ SFLDataFormat* _spkrDataConverted;
+
+ /**
+ * Sample rate converter object
+ */
+ SamplerateConverter * _converter;
+
+ /**
+ * Variables to process audio stream: sample rate for playing sound (typically 44100HZ)
+ */
+ int _audioLayerSampleRate;
+
+ /**
+ * Sample rate of the codec we use to encode and decode (most of time 8000HZ)
+ */
+ int _codecSampleRate;
+
+ /**
+ * Length of the sound frame we capture in ms (typically 20ms)
+ */
+ int _audioLayerFrameSize;
+
+ /**
+ * Codecs frame size in samples (20 ms => 882 at 44.1kHz)
+ * The exact value is stored in the codec
+ */
+ int _codecFrameSize;
+
+ /**
+ * Sampling rate of audio converter
+ */
+ int _converterSamplingRate;
+
+ /**
+ * EventQueue used to store list of DTMF-
+ */
+ EventQueue _eventQueue;
+
+ /**
+ * State of mic fade in
+ */
+ bool _micFadeInComplete;
+
+ /**
+ * State of spkr fade in
+ */
+ bool _spkrFadeInComplete;
+
+ /**
+ * Ampliturde factor to fade in mic data
+ */
+ SFLDataFormat _micAmplFactor;
+
+ /**
+ * Amplitude factor to fade in spkr data
+ */
+ SFLDataFormat _spkrAmplFactor;
+
+ /**
+ * Audio process containing noise reduction engine
+ */
+ AudioProcessing *_audioProcess;
+
+ /**
+ * Noise reduction engine
+ */
+ NoiseSuppress *_noiseSuppress;
+
+};
+
+
+class AudioRtpRecordHandler {
+public:
+ AudioRtpRecordHandler();
+ virtual ~AudioRtpRecordHandler();
+
+ /**
+ * Set rtp media for this session
+ */
+ void setRtpMedia(AudioCodec* audioCodec);
+
+ int getCodecPayloadType(void) { return _audioRtpRecord.getCodecPayloadType(); }
+
+ int getCodecSampleRate(void) { return _audioRtpRecord.getCodecSampleRate(); }
+
+ int getCodecFrameSize(void) { return _audioRtpRecord.getCodecFrameSize(); }
+
+ int getHasDynamicPayload(void) { return _audioRtpRecord.getHasDynamicPayload(); }
+
+ int getAudioLayerFrameSize(void) { return _audioRtpRecord.getAudioLayerFrameSize(); }
+
+ int getAudioLayerSampleRate(void) { return _audioRtpRecord.getAudioLayerSampleRate(); }
+
+ EventQueue *getEventQueue(void) { return _audioRtpRecord.getEventQueue(); }
+
+ int getEventQueueSize(void) { return _audioRtpRecord.getEventQueueSize(); }
+
+ unsigned char *getEncodedData(void) {return _audioRtpRecord.getEncodedData(); }
+
+ inline float computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate) {
+ return ( (float) codecSamplePerFrame * 1000.0) / (float) codecClockRate;
+ }
+
+ int computeNbByteAudioLayer (float codecFrameSize) {
+ return (int) ( ( (float) _audioRtpRecord.getCodecSampleRate() * codecFrameSize * sizeof (SFLDataFormat)) / 1000.0);
+ }
+
+ void init(void);
+
+ /**
+ * Allocate memory for RTP buffers and fill them with zeros
+ * @prereq Session codec needs to be initialized prior calling this method
+ */
+ void initBuffers (void);
+
+ /**
+ * Encode audio data from mainbuffer
+ */
+ int processDataEncode (void);
+
+ /**
+ * Decode audio data received from peer
+ */
+ void processDataDecode (unsigned char * spkrData, unsigned int size);
+
+ /**
+ * Ramp In audio data to avoid audio click from peer
+ */
+ bool fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor);
+
+private:
+
+ AudioRtpRecord _audioRtpRecord;
+
+};
+
+}
+
+#endif /* AUDIORTPRECORD_H_ */
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..64d92c5e674befb47678ed68bb0c5c93e1a85af8
--- /dev/null
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -0,0 +1,329 @@
+/*
+ * Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
+ * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
+ * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
+ * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ * Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Additional permission under GNU GPL version 3 section 7:
+ *
+ * If you modify this program, or any covered work, by linking or
+ * combining it with the OpenSSL project's OpenSSL library (or a
+ * modified version of that library), containing parts covered by the
+ * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ * grants you additional permission to convey the resulting work.
+ * Corresponding Source for a non-source form of such a combination
+ * shall include the source code for the parts of OpenSSL used as well
+ * as that of the covered work.
+ */
+
+#include "AudioRtpSession.h"
+
+namespace sfl
+{
+
+AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
+ ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
+ , _time (new ost::Time())
+ , _mainloopSemaphore (0)
+ , _manager (manager)
+ , _timestamp (0)
+ , _timestampIncrement (0)
+ , _timestampCount (0)
+ , _countNotificationTime (0)
+ , _ca (sipcall)
+{
+ setCancel (cancelDefault);
+
+ assert (_ca);
+
+ _info ("AudioRtpSession: Local audio port %i will be used", _ca->getLocalAudioPort());
+
+}
+
+AudioRtpSession::~AudioRtpSession()
+{
+ _debug ("AudioRtpSession: Delete AudioRtpSession instance");
+
+ try {
+ terminate();
+ } catch (...) {
+ _debugException ("AudioRtpSession: Thread destructor didn't terminate correctly");
+ throw;
+ }
+
+ _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
+
+ if(_time)
+ delete _time;
+ _time = NULL;
+
+}
+
+void AudioRtpSession::setSessionTimeouts (void)
+{
+ setSchedulingTimeout (sfl::schedulingTimeout);
+ setExpireTimeout (sfl::expireTimeout);
+}
+
+void AudioRtpSession::setSessionMedia (AudioCodec* audioCodec)
+{
+ // set internal codec info for this session
+ setRtpMedia(audioCodec);
+
+ // store codec info locally
+ int payloadType = getCodecPayloadType();
+ int frameSize = getCodecFrameSize();
+ int smplRate = getCodecSampleRate();
+ bool dynamic = getHasDynamicPayload();
+
+ // G722 requires timestamp to be incremented at 8 kHz
+ if (payloadType == 9)
+ _timestampIncrement = 160;
+ else
+ _timestampIncrement = frameSize;
+
+ _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
+ _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
+ _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+
+ // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
+ if (payloadType == 9) {
+ _debug ("AudioRtpSession: Setting G722 payload format");
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
+ } else if (dynamic) {
+ _debug ("AudioRtpSession: Setting dynamic payload format");
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
+ } else if (dynamic && payloadType != 9) {
+ _debug ("AudioRtpSession: Setting static payload format");
+ setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
+ }
+
+}
+
+void AudioRtpSession::setDestinationIpAddress (void)
+{
+ if (_ca == NULL) {
+ _error ("AudioRtpSession: Sipcall is gone.");
+ throw AudioRtpSessionException();
+ }
+
+ _info ("AudioRtpSession: Setting IP address for the RTP session");
+
+ // Store remote ip in case we would need to forget current destination
+ _remote_ip = ost::InetHostAddress (_ca->getLocalSDP()->get_remote_ip().c_str());
+
+ if (!_remote_ip) {
+ _warn ("AudioRtpSession: Target IP address (%s) is not correct!",
+ _ca->getLocalSDP()->get_remote_ip().data());
+ return;
+ }
+
+ // Store remote port in case we would need to forget current destination
+ _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
+
+ _info ("AudioRtpSession: New remote address for session: %s:%d",
+ _ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
+
+ if (!addDestination (_remote_ip, _remote_port)) {
+ _warn ("AudioRtpSession: Can't add new destination to session!");
+ return;
+ }
+}
+
+void AudioRtpSession::updateDestinationIpAddress (void)
+{
+ // Destination address are stored in a list in ccrtp
+ // This method remove the current destination entry
+
+ if (!forgetDestination (_remote_ip, _remote_port, _remote_port+1))
+ _warn ("AudioRtpSession: Could not remove previous destination");
+
+ // new destination is stored in call
+ // we just need to recall this method
+ setDestinationIpAddress();
+}
+
+void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
+{
+ _debug ("AudioRtpSession: Send Dtmf %d", getEventQueueSize());
+
+ _timestamp += 160;
+
+ // discard equivalent size of audio
+ processDataEncode();
+
+ // change Payload type for DTMF payload
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) 101, 8000));
+
+ // Set marker in case this is a new Event
+ if (dtmf->newevent)
+ setMark (true);
+
+ putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+
+ // This is no more a new event
+ if (dtmf->newevent) {
+ dtmf->newevent = false;
+ setMark (false);
+ }
+
+ // get back the payload to audio
+ setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) getCodecPayloadType()));
+
+ // decrease length remaining to process for this event
+ dtmf->length -= 160;
+
+ dtmf->payload.duration += 1;
+
+ // next packet is going to be the last one
+ if ( (dtmf->length - 160) < 160)
+ dtmf->payload.ebit = true;
+
+ if (dtmf->length < 160) {
+ delete dtmf;
+ getEventQueue()->pop_front();
+ }
+}
+
+bool onRTPPacketRecv (ost::IncomingRTPPkt&)
+{
+ _debug ("AudioRtpSession: onRTPPacketRecv");
+
+ return true;
+}
+
+
+void AudioRtpSession::sendMicData()
+{
+ // Increment timestamp for outgoing packet
+ _timestamp += _timestampIncrement;
+
+ int compSize = processDataEncode();
+
+ // putData put the data on RTP queue, sendImmediate bypass this queue
+ putData (_timestamp, getEncodedData(), compSize);
+}
+
+
+void AudioRtpSession::receiveSpeakerData ()
+{
+ const ost::AppDataUnit* adu = NULL;
+
+ int packetTimestamp = getFirstTimestamp();
+
+ adu = getData (packetTimestamp);
+
+ if (!adu) {
+ return;
+ }
+
+ unsigned char* spkrDataIn = NULL;
+ unsigned int size = 0;
+
+ if (adu) {
+
+ spkrDataIn = (unsigned char*) adu->getData(); // data in char
+ size = adu->getSize(); // size in char
+
+ } else {
+ _debug ("AudioRtpSession: No RTP packet available");
+ }
+
+ // DTMF over RTP, size must be over 4 in order to process it as voice data
+ if (size > 4) {
+ processDataDecode (spkrDataIn, size);
+ }
+
+ delete adu;
+}
+
+
+int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
+{
+ _debug ("RTP: Starting main thread");
+ setSessionTimeouts();
+ setSessionMedia (audiocodec);
+ initBuffers();
+ return start (_mainloopSemaphore);
+}
+
+void AudioRtpSession::run ()
+{
+
+ // Timestamp must be initialized randomly
+ _timestamp = getCurrentTimestamp();
+
+ int threadSleep = 0;
+
+ if (getCodecSampleRate() != 0) {
+ threadSleep = (getCodecFrameSize() * 1000) / getCodecSampleRate();
+ } else {
+ // TODO should not be dependent of audio layer frame size
+ threadSleep = getAudioLayerFrameSize();
+ }
+
+ TimerPort::setTimer (threadSleep);
+
+ // Set recording sampling rate
+ _ca->setRecordingSmplRate (getCodecSampleRate());
+
+ // Start audio stream (if not started) AND flush all buffers (main and urgent)
+ _manager->getAudioDriver()->startStream();
+ startRunning();
+
+ _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
+
+ while (!testCancel()) {
+
+ // Reset timestamp to make sure the timing information are up to date
+ if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
+ _timestamp = getCurrentTimestamp();
+ _timestampCount = 0;
+ }
+
+ _timestampCount++;
+
+ _manager->getAudioLayerMutex()->enter();
+
+ // TODO should not be linked to audio layer here
+ // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
+ // _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
+ // Send session
+ if (getEventQueueSize() > 0) {
+ sendDtmfEvent (getEventQueue()->front());
+ } else {
+ sendMicData ();
+ }
+
+ // Recv session
+ // TODO should not be called here anymore
+ // receiveSpeakerData ();
+
+ _manager->getAudioLayerMutex()->leave();
+
+ // Let's wait for the next transmit cycle
+ Thread::sleep (TimerPort::getTimer());
+ TimerPort::incTimer (threadSleep);
+ }
+
+ _debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
+}
+
+}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 2483b56bdc3041b79afa7fbf3bc97fe0a58681bc..07d215df1da59cca650a0c786eab27c82bc25e5d 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -40,13 +40,13 @@
#include "global.h"
+#include "AudioRtpRecordHandler.h"
#include "sip/sipcall.h"
#include "sip/sdp.h"
#include "audio/audiolayer.h"
#include "audio/codecs/audiocodec.h"
-#include "audio/samplerateconverter.h"
-#include "audio/audioprocessing.h"
-#include "audio/noisesuppress.h"
+
+
#include "managerimpl.h"
@@ -77,17 +77,9 @@ class AudioRtpSessionException: public std::exception
}
};
-typedef struct DtmfEvent {
- ost::RTPPacket::RFC2833Payload payload;
- int length;
- bool newevent;
-} DtmfEvent;
-
-typedef list<DtmfEvent *> EventQueue;
-
-template <typename D>
-class AudioRtpSession : public ost::Thread, public ost::TimerPort
+// class AudioRtpSession : public ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession
+class AudioRtpSession : public ost::TimerPort, public ost::SymmetricRTPSession, public AudioRtpRecordHandler
{
public:
/**
@@ -118,21 +110,8 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
*/
void sendDtmfEvent (sfl::DtmfEvent *dtmf);
- inline float computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate) {
- return ( (float) codecSamplePerFrame * 1000.0) / (float) codecClockRate;
- }
-
- int computeNbByteAudioLayer (float codecFrameSize) {
- return (int) ( ( (float) _converterSamplingRate * codecFrameSize * sizeof (SFLDataFormat)) / 1000.0);
- }
-
private:
- /**
- * Allocate memory for RTP buffers and fill them with zeros
- */
- void initBuffers (void);
-
/**
* Set RTP Sockets send/receive timeouts
*/
@@ -148,15 +127,6 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
*/
void setDestinationIpAddress (void);
- /**
- * Encode audio data from mainbuffer
- */
- int processDataEncode (void);
-
- /**
- * Decode audio data received from peer
- */
- void processDataDecode (unsigned char * spkrData, unsigned int size);
/**
* Send encoded data to peer
@@ -168,11 +138,6 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
*/
void receiveSpeakerData ();
- /**
- * Ramp In audio data to avoid audio click from peer
- */
- bool fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor);
-
ost::Time * _time;
// This semaphore is not used
@@ -195,62 +160,12 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
// this destination and update a new one
unsigned short _remote_port;
- // Pointer to the session's codec
- AudioCodec * _audiocodec;
-
- // Pointer to audio layer
- AudioLayer * _audiolayer;
-
- /** Mic-data related buffers */
- SFLDataFormat* _micData;
- SFLDataFormat* _micDataConverted;
- unsigned char* _micDataEncoded;
-
- /** Speaker-data related buffers */
- SFLDataFormat* _spkrDataDecoded;
- SFLDataFormat* _spkrDataConverted;
-
- /** Sample rate converter object */
- SamplerateConverter * _converter;
-
- /** Variables to process audio stream: sample rate for playing sound (typically 44100HZ) */
- int _layerSampleRate;
-
- /** Sample rate of the codec we use to encode and decode (most of time 8000HZ) */
- int _codecSampleRate;
-
- /** Length of the sound frame we capture in ms (typically 20ms) */
- int _layerFrameSize;
-
- /** Codecs frame size in samples (20 ms => 882 at 44.1kHz)
- The exact value is stored in the codec */
- int _codecFrameSize;
-
- /** Speaker buffer length in samples once the data are resampled
- * (used for mixing and recording)
- */
- int _nSamplesSpkr;
-
- /** Mic buffer length in samples once the data are resampled
- * (used for mixing and recording)
- */
- int _nSamplesMic;
-
- /**
- * Maximum number of sample for audio buffers (mic and spkr)
- */
- int _nbSamplesMax;
/**
* Manager instance.
*/
ManagerImpl * _manager;
- /**
- * Sampling rate of audio converter
- */
- int _converterSamplingRate;
-
/**
* Timestamp for this session
*/
@@ -272,660 +187,12 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
*/
int _countNotificationTime;
- /**
- * EventQueue used to store list of DTMF-
- */
- EventQueue _eventQueue;
-
- /**
- * Packet size in ms
- */
- int _packetLength;
-
- int _ts;
-
- /**
- * Current time in ms
- */
- int _currentTime;
-
- /**
- * State of mic fade in
- */
- bool _micFadeInComplete;
-
- /**
- * State of spkr fade in
- */
- bool _spkrFadeInComplete;
-
- /**
- * Ampliturde factor to fade in mic data
- */
- SFLDataFormat _micAmplFactor;
-
- /**
- * Amplitude factor to fade in spkr data
- */
- SFLDataFormat _spkrAmplFactor;
-
- /**
- * Audio process containing noise reduction engine
- */
- AudioProcessing *_audioProcess;
-
- /**
- * Noise reduction engine
- */
- NoiseSuppress *_noiseSuppress;
protected:
SIPCall * _ca;
-
- bool onRTPPacketRecv (ost::IncomingRTPPkt&);
};
-template <typename D>
-AudioRtpSession<D>::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
- _time (new ost::Time()),
- _mainloopSemaphore (0),
- _audiocodec (NULL),
- _audiolayer (NULL),
- _micData (NULL),
- _micDataConverted (NULL),
- _micDataEncoded (NULL),
- _spkrDataDecoded (NULL),
- _spkrDataConverted (NULL),
- _converter (NULL),
- _layerSampleRate (0),
- _codecSampleRate (0),
- _layerFrameSize (0),
- _manager (manager),
- _converterSamplingRate (0),
- _timestamp (0),
- _timestampIncrement (0),
- _timestampCount (0),
- _countNotificationTime (0),
- _micFadeInComplete (false),
- _spkrFadeInComplete (false),
- _micAmplFactor (32000),
- _spkrAmplFactor (32000),
- _audioProcess (NULL),
- _noiseSuppress (NULL),
- _ca (sipcall)
-{
- setCancel (cancelDefault);
-
- assert (_ca);
-
- _info ("Rtp: Local audio port %i will be used", _ca->getLocalAudioPort());
-
- //mic, we receive from soundcard in stereo, and we send encoded
- _audiolayer = _manager->getAudioDriver();
-
- if (_audiolayer == NULL) {
- throw AudioRtpSessionException();
- }
-
- _layerFrameSize = _audiolayer->getFrameSize(); // in ms
- _layerSampleRate = _audiolayer->getSampleRate();
-
- _ts= 0;
- _packetLength = 20;
- _currentTime = 0;
-
-}
-
-template <typename D>
-AudioRtpSession<D>::~AudioRtpSession()
-{
- _debug ("RTP: Delete AudioRtpSession instance");
-
- try {
- terminate();
- } catch (...) {
- _debugException ("Thread destructor didn't terminate correctly");
- throw;
- }
-
- _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
-
- if (_micData) {
- delete [] _micData;
- _micData = NULL;
- }
-
- if (_micDataConverted) {
- delete [] _micDataConverted;
- _micDataConverted = NULL;
- }
-
- if (_micDataEncoded) {
- delete [] _micDataEncoded;
- _micDataEncoded = NULL;
- }
-
- if (_spkrDataDecoded) {
- delete [] _spkrDataDecoded;
- _spkrDataDecoded = NULL;
- }
-
- if (_spkrDataConverted) {
- delete [] _spkrDataConverted;
- _spkrDataConverted = NULL;
- }
-
- delete _time;
- delete _converter;
-
- if (_audiocodec) {
- delete _audiocodec;
- _audiocodec = NULL;
- }
-
- if (_audioProcess) {
- delete _audioProcess;
- _audioProcess = NULL;
- }
-
- if (_noiseSuppress) {
- delete _noiseSuppress;
- _noiseSuppress = NULL;
- }
-}
-
-template <typename D>
-void AudioRtpSession<D>::initBuffers()
-{
- // Set sampling rate, main buffer choose the highest one
- _manager->getAudioDriver()->getMainBuffer()->setInternalSamplingRate (_codecSampleRate);
-
- // may be different than one already set
- _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
- // initialize SampleRate converter using AudioLayer's sampling rate
- // (internal buffers initialized with maximal sampling rate and frame size)
- _converter = new SamplerateConverter (_layerSampleRate, _layerFrameSize);
-
- int nbSamplesMax = (int) (_codecSampleRate * _layerFrameSize /1000) *2;
- _micData = new SFLDataFormat[nbSamplesMax];
- _micDataConverted = new SFLDataFormat[nbSamplesMax];
- _micDataEncoded = new unsigned char[nbSamplesMax*2];
- _spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- _spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
-
-
- memset (_micData, 0, nbSamplesMax*sizeof (SFLDataFormat));
- memset (_micDataConverted, 0, nbSamplesMax*sizeof (SFLDataFormat));
- memset (_micDataEncoded, 0, nbSamplesMax*2);
- memset (_spkrDataConverted, 0, nbSamplesMax*sizeof (SFLDataFormat));
- memset (_spkrDataDecoded, 0, nbSamplesMax*sizeof (SFLDataFormat));
-
- _manager->addStream (_ca->getCallId());
-}
-
-template <typename D>
-void AudioRtpSession<D>::setSessionTimeouts (void)
-{
- try {
- static_cast<D*> (this)->setSchedulingTimeout (schedulingTimeout);
- static_cast<D*> (this)->setExpireTimeout (expireTimeout);
- } catch (...) {
- _debugException ("AudioRtpSession: Initialization failed while setting timeouts");
- throw AudioRtpSessionException();
- }
-}
-
-template <typename D>
-void AudioRtpSession<D>::setSessionMedia (AudioCodec* audiocodec)
-{
- _audiocodec = audiocodec;
-
- _debug ("RTP: Init codec payload %i", _audiocodec->getPayload());
-
- _codecSampleRate = _audiocodec->getClockRate();
- _codecFrameSize = _audiocodec->getFrameSize();
-
- // G722 requires timestamp to be incremented at 8 kHz
- if (_audiocodec->getPayload() == 9)
- _timestampIncrement = 160;
- else
- _timestampIncrement = _codecFrameSize;
-
-
- _debug ("RTP: Codec sampling rate: %d", _codecSampleRate);
- _debug ("RTP: Codec frame size: %d", _codecFrameSize);
- _debug ("RTP: RTP timestamp increment: %d", _timestampIncrement);
-
- // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
- if (_audiocodec->getPayload() == 9) {
- _debug ("RTP: Setting G722 payload format");
- static_cast<D*> (this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else if (_audiocodec->hasDynamicPayload()) {
- _debug ("RTP: Setting dynamic payload format");
- static_cast<D*> (this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
- _debug ("RTP: Setting static payload format");
- static_cast<D*> (this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
- }
-
-}
-
-template <typename D>
-void AudioRtpSession<D>::setDestinationIpAddress (void)
-{
- if (_ca == NULL) {
- _error ("RTP: Sipcall is gone.");
- throw AudioRtpSessionException();
- }
-
- _info ("RTP: Setting IP address for the RTP session");
-
- // Store remote ip in case we would need to forget current destination
- _remote_ip = ost::InetHostAddress (_ca->getLocalSDP()->get_remote_ip().c_str());
-
- if (!_remote_ip) {
- _warn ("RTP: Target IP address (%s) is not correct!",
- _ca->getLocalSDP()->get_remote_ip().data());
- return;
- }
-
- // Store remote port in case we would need to forget current destination
- _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
-
- _info ("RTP: New remote address for session: %s:%d",
- _ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
-
- if (! static_cast<D*> (this)->addDestination (_remote_ip, _remote_port)) {
- _warn ("RTP: Can't add new destination to session!");
- return;
- }
-}
-
-template <typename D>
-void AudioRtpSession<D>::updateDestinationIpAddress (void)
-{
- // Destination address are stored in a list in ccrtp
- // This method remove the current destination entry
-
- if (!static_cast<D*> (this)->forgetDestination (_remote_ip, _remote_port, _remote_port+1))
- _warn ("RTP: Could not remove previous destination");
-
- // new destination is stored in call
- // we just need to recall this method
- setDestinationIpAddress();
-}
-
-template<typename D>
-void AudioRtpSession<D>::putDtmfEvent (int digit)
-{
-
- sfl::DtmfEvent *dtmf = new sfl::DtmfEvent();
-
- dtmf->payload.event = digit;
- dtmf->payload.ebit = false; // end of event bit
- dtmf->payload.rbit = false; // reserved bit
- dtmf->payload.duration = 1; // duration for this event
- dtmf->newevent = true;
- dtmf->length = 1000;
-
- _eventQueue.push_back (dtmf);
-
- _debug ("RTP: Put Dtmf Event %d", _eventQueue.size());
-
-}
-
-template<typename D>
-void AudioRtpSession<D>::sendDtmfEvent (sfl::DtmfEvent *dtmf)
-{
- _debug ("RTP: Send Dtmf %d", _eventQueue.size());
-
- _timestamp += 160;
-
- // discard equivalent size of audio
- processDataEncode();
-
- // change Payload type for DTMF payload
- static_cast<D*> (this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) 101, 8000));
-
- // Set marker in case this is a new Event
- if (dtmf->newevent)
- static_cast<D*> (this)->setMark (true);
-
- static_cast<D*> (this)->putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
-
- // This is no more a new event
- if (dtmf->newevent) {
- dtmf->newevent = false;
- static_cast<D*> (this)->setMark (false);
- }
-
- // get back the payload to audio
- static_cast<D*> (this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
-
- // decrease length remaining to process for this event
- dtmf->length -= 160;
-
- dtmf->payload.duration += 1;
-
- // next packet is going to be the last one
- if ( (dtmf->length - 160) < 160)
- dtmf->payload.ebit = true;
-
- if (dtmf->length < 160) {
- delete dtmf;
- _eventQueue.pop_front();
- }
-}
-
-template <typename D>
-bool onRTPPacketRecv (ost::IncomingRTPPkt&)
-{
- _debug ("AudioRtpSession: onRTPPacketRecv");
-
- return true;
-}
-
-
-template <typename D>
-int AudioRtpSession<D>::processDataEncode (void)
-{
- assert (_audiocodec);
- assert (_audiolayer);
-
- int _mainBufferSampleRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
- // compute codec framesize in ms
- float fixed_codec_framesize = computeCodecFrameSize (_audiocodec->getFrameSize(), _audiocodec->getClockRate());
-
- // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
- int maxBytesToGet = computeNbByteAudioLayer (fixed_codec_framesize);
-
- // available bytes inside ringbuffer
- int availBytesFromMic = _manager->getAudioDriver()->getMainBuffer()->availForGet (_ca->getCallId());
-
- // set available byte to maxByteToGet
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
- if (bytesAvail == 0) {
- memset (_micDataEncoded, 0, sizeof (SFLDataFormat));
- return _audiocodec->getFrameSize();
- }
-
- // Get bytes from micRingBuffer to data_from_mic
- int nbSample = _manager->getAudioDriver()->getMainBuffer()->getData (_micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
-
- if (!_micFadeInComplete)
- _micFadeInComplete = fadeIn (_micData, nbSample, &_micAmplFactor);
-
- if (nbSample == 0)
- return nbSample;
-
- // nb bytes to be sent over RTP
- int compSize = 0;
-
- // test if resampling is required
- if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
- int nb_sample_up = nbSample;
-
- _nSamplesMic = nbSample;
-
- nbSample = _converter->downsampleData (_micData , _micDataConverted , _audiocodec->getClockRate(), _mainBufferSampleRate, nb_sample_up);
-
- if (_manager->audioPreference.getNoiseReduce())
- _audioProcess->processAudio (_micDataConverted, nbSample*sizeof (SFLDataFormat));
-
- compSize = _audiocodec->codecEncode (_micDataEncoded, _micDataConverted, nbSample*sizeof (SFLDataFormat));
-
- } else {
-
- _nSamplesMic = nbSample;
-
- if (_manager->audioPreference.getNoiseReduce())
- _audioProcess->processAudio (_micData, nbSample*sizeof (SFLDataFormat));
-
- // no resampling required
- compSize = _audiocodec->codecEncode (_micDataEncoded, _micData, nbSample*sizeof (SFLDataFormat));
-
- }
-
- return compSize;
-}
-
-template <typename D>
-void AudioRtpSession<D>::processDataDecode (unsigned char * spkrData, unsigned int size)
-{
- if (_audiocodec != NULL) {
-
-
- int _mainBufferSampleRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
- // Return the size of data in bytes
- int expandedSize = _audiocodec->codecDecode (_spkrDataDecoded , spkrData , size);
-
- // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
- int nbSample = expandedSize / sizeof (SFLDataFormat);
-
- if (!_spkrFadeInComplete)
- _spkrFadeInComplete = fadeIn (_spkrDataDecoded, nbSample, &_spkrAmplFactor);
-
- // test if resampling is required
- if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
-
- // Do sample rate conversion
- int nb_sample_down = nbSample;
-
- nbSample = _converter->upsampleData (_spkrDataDecoded, _spkrDataConverted, _codecSampleRate, _mainBufferSampleRate, nb_sample_down);
-
- // Store the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- // put data in audio layer, size in byte
- _manager->getAudioDriver()->getMainBuffer()->putData (_spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
-
-
- } else {
- // Store the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- // put data in audio layer, size in byte
- _manager->getAudioDriver()->getMainBuffer()->putData (_spkrDataDecoded, expandedSize, 100, _ca->getCallId());
- }
-
- // Notify (with a beep) an incoming call when there is already a call
- if (_manager->incomingCallWaiting() > 0) {
- _countNotificationTime += _time->getSecond();
- int countTimeModulo = _countNotificationTime % 5000;
-
- // _debug("countNotificationTime: %d\n", countNotificationTime);
- // _debug("countTimeModulo: %d\n", countTimeModulo);
- if ( (countTimeModulo - _countNotificationTime) < 0) {
- _manager->notificationIncomingCall();
- }
-
- _countNotificationTime = countTimeModulo;
- }
-
- }
-}
-
-template <typename D>
-void AudioRtpSession<D>::sendMicData()
-{
- // STEP:
- // 1. get data from mic
- // 2. convert it to int16 - good sample, good rate
- // 3. encode it
- // 4. send it
-
- // Increment timestamp for outgoing packet
- _timestamp += _timestampIncrement;
-
- if (!_audiolayer) {
- _debug ("No audiolayer available for MIC");
- return;
- }
-
- if (!_audiocodec) {
- _debug ("No audiocodec available for MIC");
- return;
- }
-
- int compSize = processDataEncode();
-
- // putData put the data on RTP queue, sendImmediate bypass this queue
- static_cast<D*> (this)->putData (_timestamp, _micDataEncoded, compSize);
-}
-
-
-template <typename D>
-void AudioRtpSession<D>::receiveSpeakerData ()
-{
-
- if (!_audiolayer) {
- _debug ("No audiolayer available for speaker");
- return;
- }
-
- if (!_audiocodec) {
- _debug ("No audiocodec available for speaker");
- return;
- }
-
- const ost::AppDataUnit* adu = NULL;
-
- int packetTimestamp = static_cast<D*> (this)->getFirstTimestamp();
-
- adu = static_cast<D*> (this)->getData (packetTimestamp);
-
- if (!adu) {
- return;
- }
-
- unsigned char* spkrDataIn = NULL;
- unsigned int size = 0;
-
- if (adu) {
-
- spkrDataIn = (unsigned char*) adu->getData(); // data in char
- size = adu->getSize(); // size in char
-
- } else {
- _debug ("No RTP packet available !!!!!!!!!!!!!!!!!!!!!!!\n");
- }
-
- // DTMF over RTP, size must be over 4 in order to process it as voice data
- if (size > 4) {
- processDataDecode (spkrDataIn, size);
- }
-
- delete adu;
-}
-
-template <typename D>
-bool AudioRtpSession<D>::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
-{
-
- // apply amplitude factor;
- while (size) {
- size--;
- audio[size] /= *factor;
- }
-
- // decrease factor
- *factor /= FADEIN_STEP_SIZE;
-
- // if factor reach 0, thsi function should no be called anymore
- if (*factor == 0)
- return true;
-
- return false;
-}
-
-template <typename D>
-int AudioRtpSession<D>::startRtpThread (AudioCodec* audiocodec)
-{
- _debug ("RTP: Starting main thread");
- setSessionTimeouts();
- setSessionMedia (audiocodec);
- initBuffers();
- return start (_mainloopSemaphore);
-}
-
-template <typename D>
-void AudioRtpSession<D>::run ()
-{
-
- // Timestamp must be initialized randomly
- _timestamp = static_cast<D*> (this)->getCurrentTimestamp();
-
- int threadSleep = 0;
-
- if (_codecSampleRate != 0) {
- threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
- } else {
- threadSleep = _layerFrameSize;
- }
-
- TimerPort::setTimer (threadSleep);
-
- if (_audiolayer == NULL) {
- _error ("RTP: Error: Audiolayer is null, cannot start the audio stream");
- throw AudioRtpSessionException();
- }
-
- // Set recording sampling rate
- _ca->setRecordingSmplRate (_audiocodec->getClockRate());
-
- // init noise reduction process
- _noiseSuppress = new NoiseSuppress (_codecFrameSize, _audiocodec->getClockRate());
- _audioProcess = new AudioProcessing (_noiseSuppress);
-
- // Start audio stream (if not started) AND flush all buffers (main and urgent)
- _manager->getAudioDriver()->startStream();
- static_cast<D*> (this)->startRunning();
-
- _debug ("RTP: Entering mainloop for call %s",_ca->getCallId().c_str());
-
- _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
- while (!testCancel()) {
-
- // Reset timestamp to make sure the timing information are up to date
- if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
- _timestamp = static_cast<D*> (this)->getCurrentTimestamp();
- _timestampCount = 0;
- }
-
- _timestampCount++;
-
-
- _manager->getAudioLayerMutex()->enter();
-
- // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
- _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
- // Send session
- if (_eventQueue.size() > 0) {
- sendDtmfEvent (_eventQueue.front());
- } else {
- sendMicData ();
- }
-
- // Recv session
- receiveSpeakerData ();
-
- _manager->getAudioLayerMutex()->leave();
-
- // Let's wait for the next transmit cycle
- Thread::sleep (TimerPort::getTimer());
- TimerPort::incTimer (threadSleep);
- }
-
- _debug ("RTP: Left main loop for call%s", _ca->getCallId().c_str());
-}
-
}
#endif // __AUDIO_RTP_SESSION_H__
diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
index 281cff8d9cdd87e60925aec13a902716f3c071c2..0b24c4ec4b17de87f31e1cc16beca2b726d1033d 100644
--- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
@@ -74,7 +74,7 @@ class SrtpException: public std::exception
}
};
-class AudioSrtpSession : public ost::SymmetricRTPSession, public AudioRtpSession<AudioSrtpSession>
+class AudioSrtpSession : public AudioRtpSession
{
public:
diff --git a/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
index a2275e3b508da1727d365fb03c2bfb10ff8bee80..330c8122f261e157673d8fedc38f47ae06e980ab 100644
--- a/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
@@ -37,13 +37,11 @@ class ManagerImpl;
namespace sfl
{
-class AudioSymmetricRtpSession : public ost::SymmetricRTPSession, public AudioRtpSession<AudioSymmetricRtpSession>
+class AudioSymmetricRtpSession : public AudioRtpSession
{
public:
- AudioSymmetricRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
- ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
- AudioRtpSession<AudioSymmetricRtpSession> (manager, sipcall) {
- _debug ("AudioSymmetricRtpSession initialized\n");
+ AudioSymmetricRtpSession (ManagerImpl * manager, SIPCall * sipcall) : AudioRtpSession (manager, sipcall) {
+
}
};
}
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
index 6632bc2a61c63f87223b32b8447e13c6f5e4febe..c450b934121688a8e8aeec28ba9896322cf986b4 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
@@ -43,14 +43,47 @@ namespace sfl
{
AudioZrtpSession::AudioZrtpSession (ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename) :
- ost::SymmetricZRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
- AudioRtpSession<AudioZrtpSession> (manager, sipcall),
- _zidFilename (zidFilename)
+ ost::SymmetricZRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
+ , _zidFilename (zidFilename)
+ , _time (new ost::Time())
+ , _mainloopSemaphore (0)
+ , _manager (manager)
+ , _timestamp (0)
+ , _timestampIncrement (0)
+ , _timestampCount (0)
+ , _countNotificationTime (0)
+ , _ca (sipcall)
{
_debug ("AudioZrtpSession initialized");
initializeZid();
+
+ setCancel (cancelDefault);
+
+ assert (_ca);
+
+ _info ("AudioZrtpSession: Local audio port %i will be used", _ca->getLocalAudioPort());
+}
+
+AudioZrtpSession::~AudioZrtpSession()
+{
+ _debug ("AudioZrtpSession: Delete AudioRtpSession instance");
+
+ try {
+ terminate();
+ } catch (...) {
+ _debugException ("AudioZrtpSession: Thread destructor didn't terminate correctly");
+ throw;
+ }
+
+ _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
+
+ if(_time)
+ delete _time;
+ _time = NULL;
}
+
+
void AudioZrtpSession::initializeZid (void)
{
@@ -95,4 +128,256 @@ void AudioZrtpSession::initializeZid (void)
return;
}
+
+void AudioRtpSession::setSessionTimeouts (void)
+{
+ setSchedulingTimeout (sfl::schedulingTimeout);
+ setExpireTimeout (sfl::expireTimeout);
+}
+
+void AudioRtpSession::setSessionMedia (AudioCodec* audioCodec)
+{
+ // set internal codec info for this session
+ setRtpMedia(audioCodec);
+
+ // store codec info locally
+ int payloadType = getCodecPayloadType();
+ int frameSize = getCodecFrameSize();
+ int smplRate = getCodecSampleRate();
+ bool dynamic = getHasDynamicPayload();
+
+ // G722 requires timestamp to be incremented at 8 kHz
+ if (payloadType == 9)
+ _timestampIncrement = 160;
+ else
+ _timestampIncrement = frameSize;
+
+ _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
+ _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
+ _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+
+ // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
+ if (payloadType == 9) {
+ _debug ("AudioRtpSession: Setting G722 payload format");
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
+ } else if (dynamic) {
+ _debug ("AudioRtpSession: Setting dynamic payload format");
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
+ } else if (dynamic && payloadType != 9) {
+ _debug ("AudioRtpSession: Setting static payload format");
+ setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
+ }
+
+}
+
+void AudioRtpSession::setDestinationIpAddress (void)
+{
+ if (_ca == NULL) {
+ _error ("AudioRtpSession: Sipcall is gone.");
+ throw AudioRtpSessionException();
+ }
+
+ _info ("AudioRtpSession: Setting IP address for the RTP session");
+
+ // Store remote ip in case we would need to forget current destination
+ _remote_ip = ost::InetHostAddress (_ca->getLocalSDP()->get_remote_ip().c_str());
+
+ if (!_remote_ip) {
+ _warn ("AudioRtpSession: Target IP address (%s) is not correct!",
+ _ca->getLocalSDP()->get_remote_ip().data());
+ return;
+ }
+
+ // Store remote port in case we would need to forget current destination
+ _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
+
+ _info ("AudioRtpSession: New remote address for session: %s:%d",
+ _ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
+
+ if (!addDestination (_remote_ip, _remote_port)) {
+ _warn ("AudioRtpSession: Can't add new destination to session!");
+ return;
+ }
+}
+
+void AudioRtpSession::updateDestinationIpAddress (void)
+{
+ // Destination address are stored in a list in ccrtp
+ // This method remove the current destination entry
+
+ if (!forgetDestination (_remote_ip, _remote_port, _remote_port+1))
+ _warn ("AudioRtpSession: Could not remove previous destination");
+
+ // new destination is stored in call
+ // we just need to recall this method
+ setDestinationIpAddress();
+}
+
+void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
+{
+ _debug ("AudioRtpSession: Send Dtmf %d", getEventQueueSize());
+
+ _timestamp += 160;
+
+ // discard equivalent size of audio
+ processDataEncode();
+
+ // change Payload type for DTMF payload
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) 101, 8000));
+
+ // Set marker in case this is a new Event
+ if (dtmf->newevent)
+ setMark (true);
+
+ putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+
+ // This is no more a new event
+ if (dtmf->newevent) {
+ dtmf->newevent = false;
+ setMark (false);
+ }
+
+ // get back the payload to audio
+ setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) getCodecPayloadType()));
+
+ // decrease length remaining to process for this event
+ dtmf->length -= 160;
+
+ dtmf->payload.duration += 1;
+
+ // next packet is going to be the last one
+ if ( (dtmf->length - 160) < 160)
+ dtmf->payload.ebit = true;
+
+ if (dtmf->length < 160) {
+ delete dtmf;
+ getEventQueue()->pop_front();
+ }
+}
+
+bool onRTPPacketRecv (ost::IncomingRTPPkt&)
+{
+ _debug ("AudioRtpSession: onRTPPacketRecv");
+
+ return true;
+}
+
+
+void AudioRtpSession::sendMicData()
+{
+ // Increment timestamp for outgoing packet
+ _timestamp += _timestampIncrement;
+
+ int compSize = processDataEncode();
+
+ // putData put the data on RTP queue, sendImmediate bypass this queue
+ putData (_timestamp, getEncodedData(), compSize);
+}
+
+
+void AudioRtpSession::receiveSpeakerData ()
+{
+ const ost::AppDataUnit* adu = NULL;
+
+ int packetTimestamp = getFirstTimestamp();
+
+ adu = getData (packetTimestamp);
+
+ if (!adu) {
+ return;
+ }
+
+ unsigned char* spkrDataIn = NULL;
+ unsigned int size = 0;
+
+ if (adu) {
+
+ spkrDataIn = (unsigned char*) adu->getData(); // data in char
+ size = adu->getSize(); // size in char
+
+ } else {
+ _debug ("AudioRtpSession: No RTP packet available");
+ }
+
+ // DTMF over RTP, size must be over 4 in order to process it as voice data
+ if (size > 4) {
+ processDataDecode (spkrDataIn, size);
+ }
+
+ delete adu;
+}
+
+
+int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
+{
+ _debug ("RTP: Starting main thread");
+ setSessionTimeouts();
+ setSessionMedia (audiocodec);
+ initBuffers();
+ return start (_mainloopSemaphore);
+}
+
+void AudioRtpSession::run ()
+{
+
+ // Timestamp must be initialized randomly
+ _timestamp = getCurrentTimestamp();
+
+ int threadSleep = 0;
+
+ if (getCodecSampleRate() != 0) {
+ threadSleep = (getCodecFrameSize() * 1000) / getCodecSampleRate();
+ } else {
+ // TODO should not be dependent of audio layer frame size
+ threadSleep = getAudioLayerFrameSize();
+ }
+
+ TimerPort::setTimer (threadSleep);
+
+ // Set recording sampling rate
+ _ca->setRecordingSmplRate (getCodecSampleRate());
+
+ // Start audio stream (if not started) AND flush all buffers (main and urgent)
+ _manager->getAudioDriver()->startStream();
+ startRunning();
+
+ _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
+
+ while (!testCancel()) {
+
+ // Reset timestamp to make sure the timing information are up to date
+ if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
+ _timestamp = getCurrentTimestamp();
+ _timestampCount = 0;
+ }
+
+ _timestampCount++;
+
+ _manager->getAudioLayerMutex()->enter();
+
+ // TODO should not be linked to audio layer here
+ // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
+ // _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
+ // Send session
+ if (getEventQueueSize() > 0) {
+ sendDtmfEvent (getEventQueue()->front());
+ } else {
+ sendMicData ();
+ }
+
+ // Recv session
+ // TODO should not be called here anymore
+ // receiveSpeakerData ();
+
+ _manager->getAudioLayerMutex()->leave();
+
+ // Let's wait for the next transmit cycle
+ Thread::sleep (TimerPort::getTimer());
+ TimerPort::incTimer (threadSleep);
+ }
+
+ _debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
+}
+
}
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
index 4eb68d364d8a6e057dc74de65bf2d13e9706dcc2..c50b08662aecfa90d42539487dcedccd62af2c7d 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
@@ -31,7 +31,7 @@
#include <libzrtpcpp/zrtpccrtp.h>
-#include "AudioRtpSession.h"
+#include "AudioRtpRecordHandler.h"
class ManagerImpl;
class SIPCall;
@@ -46,7 +46,7 @@ class ZrtpZidException: public std::exception
}
};
-class AudioZrtpSession : public ost::SymmetricZRTPSession, public AudioRtpSession<AudioZrtpSession>
+class AudioZrtpSession : public ost::TimerPort, public ost::SymmetricZRTPSession, public AudioRtpRecordHandler
{
public:
AudioZrtpSession (ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename);
@@ -54,6 +54,56 @@ class AudioZrtpSession : public ost::SymmetricZRTPSession, public AudioRtpSessio
private:
void initializeZid (void);
std::string _zidFilename;
+
+ ost::Time * _time;
+
+ // This semaphore is not used
+ // but is needed in order to avoid
+ // ambiguous compiling problem.
+ // It is set to 0, and since it is
+ // optional in ost::thread, then
+ // it amounts to the same as doing
+ // start() with no semaphore at all.
+ ost::Semaphore * _mainloopSemaphore;
+
+ // Main destination address for this rtp session.
+ // Stored in case or reINVITE, which may require to forget
+ // this destination and update a new one.
+ ost::InetHostAddress _remote_ip;
+
+
+ // Main destination port for this rtp session.
+ // Stored in case reINVITE, which may require to forget
+ // this destination and update a new one
+ unsigned short _remote_port;
+
+ /**
+ * Manager instance.
+ */
+ ManagerImpl * _manager;
+
+ /**
+ * Timestamp for this session
+ */
+ int _timestamp;
+
+ /**
+ * Timestamp incrementation value based on codec period length (framesize)
+ * except for G722 which require a 8 kHz incrementation.
+ */
+ int _timestampIncrement;
+
+ /**
+ * Timestamp reset freqeuncy specified in number of packet sent
+ */
+ short _timestampCount;
+
+ /**
+ * Time counter used to trigger incoming call notification
+ */
+ int _countNotificationTime;
+
+ SIPCall * _ca;
};
}
diff --git a/sflphone-common/src/audio/audiortp/Makefile.am b/sflphone-common/src/audio/audiortp/Makefile.am
index d2ab3aaec714d9065241752748284fe4ea59a0c2..f75a5973ea2feff0ebab76bbd6e5d3fb3922262f 100644
--- a/sflphone-common/src/audio/audiortp/Makefile.am
+++ b/sflphone-common/src/audio/audiortp/Makefile.am
@@ -3,12 +3,15 @@ include $(top_srcdir)/globals.mak
noinst_LTLIBRARIES = libaudiortp.la
libaudiortp_la_SOURCES = \
+ AudioRtpSession.cpp \
+ AudioRtpRecordHandler.cpp \
AudioRtpFactory.cpp \
AudioZrtpSession.cpp \
ZrtpSessionCallback.cpp \
AudioSrtpSession.cpp
noinst_HEADERS = \
+ AudioRtpRecordHandler.h \
AudioRtpFactory.h \
AudioRtpSession.h \
AudioSymmetricRtpSession.h \