diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
deleted file mode 100644
index 26615888a789d5b863c2a3e9f284fca381e2b458..0000000000000000000000000000000000000000
--- a/sflphone-common/src/audio/audiortp.cpp
+++ /dev/null
@@ -1,685 +0,0 @@
-/*
- * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@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.
- */
-
-#include <cstdio>
-#include <cstdlib>
-#include <ccrtp/rtp.h>
-#include <assert.h>
-#include <cstring>
-#include <math.h>
-#include <dlfcn.h>
-#include <iostream>
-#include <sstream>
-
-#include "../global.h"
-#include "../manager.h"
-#include "codecDescriptor.h"
-#include "audiortp.h"
-#include "audiolayer.h"
-#include "ringbuffer.h"
-#include "../user_cfg.h"
-#include "../sipcall.h"
-
-
-int AudioRtpRTX::count_rtp = 0;
-
-////////////////////////////////////////////////////////////////////////////////
-// AudioRtp
-////////////////////////////////////////////////////////////////////////////////
-AudioRtp::AudioRtp() :_RTXThread (0), _symmetric(), _rtpMutex()
-{
-
-}
-
-AudioRtp::~AudioRtp (void)
-{
- ost::MutexLock m (_rtpMutex);
-
- if (_RTXThread != _RTXThread) {
- delete _RTXThread;
- _RTXThread = 0;
- }
-
-}
-
-void
-AudioRtp::createNewSession (SIPCall *ca)
-{
-
- ost::MutexLock m (_rtpMutex);
-
- _debug ("AudioRtp::Create new rtp session");
-
- // something should stop the thread before...
-
- if (_RTXThread != 0) {
- _debug ("**********************************************************");
- _debug ("! ARTP Failure: Thread already exists..., stopping it");
- _debug ("**********************************************************");
- delete _RTXThread;
- _RTXThread = 0;
- }
-
- // Start RTP Send/Receive threads
- _symmetric = Manager::instance().getConfigInt (SIGNALISATION,SYMMETRIC) ? true : false;
-
- _RTXThread = new AudioRtpRTX (ca, _symmetric);
-
-}
-
-int
-AudioRtp::start (void)
-{
- ost::MutexLock m (_rtpMutex);
-
- if (_RTXThread == 0) {
- _debug ("! ARTP Failure: Cannot start audiortp thread since not yet created");
- throw AudioRtpException();
- }
-
-
- try {
- if (_RTXThread->start() != 0) {
- _debug ("! ARTP Failure: unable to start RTX Thread");
- return -1;
- }
- } catch (...) {
- _debugException ("! ARTP Failure: when trying to start a thread");
- throw;
- }
-
- return 0;
-}
-
-
-bool
-AudioRtp::closeRtpSession ()
-{
-
- ost::MutexLock m (_rtpMutex);
- // This will make RTP threads finish.
- _debug ("AudioRtp::Stopping rtp session");
-
- try {
- if (_RTXThread != 0) {
- delete _RTXThread;
- _RTXThread = 0;
- }
- } catch (...) {
- _debugException ("! ARTP Exception: when stopping audiortp");
- throw;
- }
-
- _debug ("AudioRtp::Audio rtp stopped");
-
- return true;
-}
-
-/*
-void
-AudioRtp::setRecording()
-{
-
- _debug ("AudioRtp::setRecording");
- _RTXThread->_ca->setRecording();
-
-}
-*/
-
-////////////////////////////////////////////////////////////////////////////////
-// AudioRtpRTX Class //
-////////////////////////////////////////////////////////////////////////////////
-AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time (new ost::Time()), _ca (sipcall), _sessionSend (NULL), _sessionRecv (NULL), _session (NULL),
- _sym (sym), micData (NULL), micDataConverted (NULL), micDataEncoded (NULL), spkrDataDecoded (NULL), spkrDataConverted (NULL),
- converter (NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec (NULL)
-{
-
- setCancel (cancelDefault);
- // AudioRtpRTX should be close if we change sample rate
- // TODO: Change bind address according to user settings.
- // TODO: this should be the local ip not the external (router) IP
- std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
- ost::InetHostAddress local_ip (localipConfig.c_str());
-
- _debug ("%i", _ca->getLocalAudioPort());
- _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
- // _session = new ost::RTPSessionBase(local_ip, _ca->getLocalAudioPort());
- _sessionRecv = NULL;
- _sessionSend = NULL;
-
- //mic, we receive from soundcard in stereo, and we send encoded
- //encoding before sending
- _audiolayer = Manager::instance().getAudioDriver();
- _layerFrameSize = _audiolayer->getFrameSize(); // in ms
- _layerSampleRate = _audiolayer->getSampleRate();
-
- // initBuffers();
-
- // initAudioRtpSession();
-
- _payloadIsSet = false;
- _remoteIpIsSet = false;
-
-
- count_rtp++;
- // open files
- std::string s_input;
- std::string s_output;
-
- // convert count into string
- std::stringstream out;
- out << count_rtp;
-
- s_input = "/home/alexandresavard/Desktop/buffer_record/rtp_input_";
- s_input.append (out.str());
-
- s_output = "/home/alexandresavard/Desktop/buffer_record/rtp_output_";
- s_output.append (out.str());
-
- rtp_input_rec = new std::fstream();
- rtp_output_rec = new std::fstream();
-
- rtp_input_rec->open (s_input.c_str(), std::fstream::out);
- rtp_output_rec->open (s_output.c_str(), std::fstream::out);
-
-}
-
-AudioRtpRTX::~AudioRtpRTX ()
-{
-
- ost::MutexLock m (_rtpRtxMutex);
-
- _debug ("Delete AudioRtpRTX instance in callid %s", _ca->getCallId().c_str());
-
- try {
- this->terminate();
- } catch (...) {
- _debugException ("! ARTP: Thread destructor didn't terminate correctly");
- throw;
- }
-
- _debug ("Remove audio stream for call id %s", _ca->getCallId().c_str());
-
- _audiolayer->getMainBuffer()->unBindAll (_ca->getCallId());
- // Manager::instance().removeStream(_ca->getCallId());
-
- _debug ("DELETE print micData address %p", micData);
- delete [] micData;
- micData = NULL;
- delete [] micDataConverted;
- micDataConverted = NULL;
- delete [] micDataEncoded;
- micDataEncoded = NULL;
-
- delete [] spkrDataDecoded;
- spkrDataDecoded = NULL;
- delete [] spkrDataConverted;
- spkrDataConverted = NULL;
-
- delete time;
- time = NULL;
-
- delete converter;
- converter = NULL;
-
- _ca = 0;
- // _session->terminate();
-
- delete _session;
- _session = NULL;
-
- _debug ("AudioRtpRTX instance deleted");
-
- rtp_input_rec->close();
- rtp_output_rec->close();
-
-}
-
-
-void
-AudioRtpRTX::initBuffers()
-{
- ost::MutexLock m (_rtpRtxMutex);
-
- _debug ("AudioRtpRTX::initBuffers Init RTP buffers for %s", _ca->getCallId().c_str());
-
- converter = new SamplerateConverter (_layerSampleRate , _layerFrameSize);
-
- nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
-
-
- _debug ("AudioRtpRTX::initBuffers NBSAMPLEMAX %i", nbSamplesMax);
-
- micData = new SFLDataFormat[nbSamplesMax];
- _debug ("CREATE print micData address %p", micData);
- micDataConverted = new SFLDataFormat[nbSamplesMax];
- micDataEncoded = new unsigned char[nbSamplesMax];
-
- spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
-
- Manager::instance().addStream (_ca->getCallId());
- // _audiolayer->getMainBuffer()->bindCallID(_ca->getCallId());
-}
-
-
-void
-AudioRtpRTX::initAudioRtpSession (void)
-{
-
- try {
-
- _session->setSchedulingTimeout (100000);
- _session->setExpireTimeout (1000000);
-
-
- } catch (...) {
- _debugException ("! ARTP Failure: initialisation failed");
- throw;
- }
-
-}
-
-void
-AudioRtpRTX::setRtpSessionMedia (void)
-{
-
- if (_ca == 0) {
- _debug (" !ARTP: No call, can't init RTP media");
- return;
- }
-
- AudioCodecType pl = (AudioCodecType) _ca->getLocalSDP()->get_session_media()->getPayload();
-
- _audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (pl);
-
- if (_audiocodec == NULL) {
- _debug (" !ARTP: No audiocodec, can't init RTP media");
- return;
- }
-
- _debug ("Init audio RTP session: codec payload %i", _audiocodec->getPayload());
-
- // _audioCodecInstance = *_audiocodec;
-
- _codecSampleRate = _audiocodec->getClockRate();
-
- _codecFrameSize = _audiocodec->getFrameSize();
-
- if (_audiocodec->getPayload() == 9) {
- _payloadIsSet = _session->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else if (_audiocodec->hasDynamicPayload()) {
- _payloadIsSet = _session->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
- _payloadIsSet = _session->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
- }
-
-
-
-}
-
-void
-AudioRtpRTX::setRtpSessionRemoteIp (void)
-{
-
- if (!_remoteIpIsSet) {
-
- _debug ("++++++++++++++++++++++++++ SET IP ADDRESS ++++++++++++++++++++++++++++");
-
- if (_ca == 0) {
- _debug (" !ARTP: No call, can't init RTP media ");
- return;
- }
-
- ost::InetHostAddress remote_ip (_ca->getLocalSDP()->get_remote_ip().c_str());
-
- _debug ("Init audio RTP session: remote ip %s", _ca->getLocalSDP()->get_remote_ip().data());
-
- if (!remote_ip) {
- _debug (" !ARTP Thread Error: Target IP address [%s] is not correct!", _ca->getLocalSDP()->get_remote_ip().data());
- return;
- }
-
- _debug ("++++Address: %s, audioport: %d", _ca->getLocalSDP()->get_remote_ip().c_str(), _ca->getLocalSDP()->get_remote_audio_port());
-
- _debug ("++++Audioport: %d", (int) _ca->getLocalSDP()->get_remote_audio_port());
-
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getLocalSDP()->get_remote_audio_port())) {
- _debug (" !ARTP Thread Error: can't add destination to session!");
- return;
- }
-
- _remoteIpIsSet = true;
- } else {
- _debug ("+++++++++++++++++++++++ IP ADDRESS ALREADY SET ++++++++++++++++++++++++");
- }
-
-}
-
-
-
-float
-AudioRtpRTX::computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate)
-{
- return ( (float) codecSamplePerFrame * 1000.0) / (float) codecClockRate;
-}
-
-int
-AudioRtpRTX::computeNbByteAudioLayer (float codecFrameSize)
-{
- return (int) ( (float) _layerSampleRate * codecFrameSize * (float) sizeof (SFLDataFormat) / 1000.0);
-}
-
-
-int
-AudioRtpRTX::processDataEncode()
-{
-
- // 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 = _audiolayer->getMainBuffer()->availForGet (_ca->getCallId());
-
- // _debug("AudioRtpRTX::processDataEncode() callid: %s availBytesFromMic %i", _ca->getCallId().c_str(), availBytesFromMic);
-
- // _debug("AudioRtpRTX::processDataEncode: availBytesFromMic: %i", availBytesFromMic);
- // set available byte to maxByteToGet
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- // _debug("bytesAvail %i", bytesAvail);
-
- if (bytesAvail == 0)
- return 0;
-
- // _debug("AudioRtpRTX::processDataEncode: bytesAvail: %i", bytesAvail);
- // Get bytes from micRingBuffer to data_from_mic
- int nbSample = _audiolayer->getMainBuffer()->getData (micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
-
- rtp_output_rec->write ( (char*) micData, bytesAvail);
-
- // _debug("AudioRtpRTX::processDataEncode: nbSample: %i", nbSample);
-
- // nb bytes to be sent over RTP
- int compSize = 0;
-
- // test if resampling is required
- if (_audiocodec->getClockRate() != _layerSampleRate) {
-
- int nb_sample_up = nbSample;
- //_debug("_nbSample audiolayer->getMic(): %i ", nbSample);
-
- // Store the length of the mic buffer in samples for recording
- _nSamplesMic = nbSample;
-
- nbSample = reSampleData (micData , micDataConverted, _audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
-
- compSize = _audiocodec->codecEncode (micDataEncoded, micDataConverted, nbSample*sizeof (int16));
-
- } else {
- // no resampling required
- compSize = _audiocodec->codecEncode (micDataEncoded, micData, nbSample*sizeof (int16));
- }
-
- return compSize;
-}
-
-
-void
-AudioRtpRTX::processDataDecode (unsigned char* spkrData, unsigned int size, int& countTime)
-{
- if (_audiocodec != NULL) {
-
- // 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);
-
- // test if resampling is required
-
- if (_audiocodec->getClockRate() != _layerSampleRate) {
-
- // Do sample rate conversion
- int nb_sample_down = nbSample;
- nbSample = reSampleData (spkrDataDecoded, spkrDataConverted, _codecSampleRate, nb_sample_down, UP_SAMPLING);
-
- rtp_input_rec->write ( (char*) spkrDataConverted, nbSample * sizeof (SFLDataFormat));
-
- // Store the number of samples for recording
- _nSamplesSpkr = nbSample;
-
-
-
- // put data in audio layer, size in byte
- _audiolayer->getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
-
- } else {
-
- // Stor the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- // put data in audio layer, size in byte
- _audiolayer->getMainBuffer()->putData (spkrDataDecoded, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
- }
-
- // Notify (with a beep) an incoming call when there is already a call
- countTime += time->getSecond();
-
- if (Manager::instance().incomingCallWaiting() > 0) {
- countTime = countTime % 500; // more often...
-
- if (countTime == 0) {
- Manager::instance().notificationIncomingCall();
- }
- }
-
- } else {
- countTime += time->getSecond();
- }
-}
-
-void
-AudioRtpRTX::sendSessionFromMic (int timestamp)
-{
- // STEP:
- // 1. get data from mic
- // 2. convert it to int16 - good sample, good rate
- // 3. encode it
- // 4. send it
-
- timestamp += time->getSecond();
- // no call, so we do nothing
-
- if (_ca==0) {
- _debug (" !ARTP: No call associated (mic)");
- return;
- }
-
- // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- if (!_audiolayer) {
- _debug (" !ARTP: No audiolayer available for MIC");
- return;
- }
-
- if (!_audiocodec) {
- _debug (" !ARTP: No audiocodec available for MIC");
- return;
- }
-
-
-
- int compSize = processDataEncode();
-
- // putData put the data on RTP queue, sendImmediate bypass this queue
- // _debug("AudioRtpRTX::sendSessionFromMic: timestamp: %i, compsize: %i", timestamp, compSize);
-
- if ( (compSize != 0) && (micDataEncoded != NULL))
- _session->putData (timestamp, micDataEncoded, compSize);
-
- // _session->sendImmediate(timestamp, micDataEncoded, compSize);
-
-
-}
-
-
-void
-AudioRtpRTX::receiveSessionForSpkr (int& countTime)
-{
-
- if (_ca == 0) {
- return;
- }
-
- if (!_audiolayer) {
- _debug (" !ARTP: No audiolayer available for SPEAKER");
- return;
- }
-
- if (!_audiocodec) {
- _debug (" !ARTP: No audiocodec available for SPEAKER");
- return;
- }
-
- const ost::AppDataUnit* adu = NULL;
-
- // int is_waiting = _session->isWaiting();
-
- // if (is_waiting != 0)
- adu = _session->getData (_session->getFirstTimestamp());
-
- // else
- // return;
-
-
-
- if (adu == NULL)
- return;
-
- unsigned char* spkrData = (unsigned char*) adu->getData(); // data in char
-
- unsigned int size = adu->getSize(); // size in char
-
- processDataDecode (spkrData, size, countTime);
-
- if (adu != NULL) {
- delete adu;
- adu = NULL;
- }
-
-}
-
-
-int
-AudioRtpRTX::reSampleData (SFLDataFormat *input, SFLDataFormat *output, int sampleRate_codec, int nbSamples, int status)
-{
- if (status==UP_SAMPLING) {
- return converter->upsampleData (input, output, sampleRate_codec , _layerSampleRate , nbSamples);
- } else if (status==DOWN_SAMPLING) {
- return converter->downsampleData (micData , micDataConverted , sampleRate_codec , _layerSampleRate , nbSamples);
- } else
-
- return 0;
-}
-
-
-
-void
-AudioRtpRTX::run ()
-{
-
- int sessionWaiting;
-
- initBuffers();
- initAudioRtpSession();
- setRtpSessionRemoteIp();
- setRtpSessionMedia();
-
- int timestep = _codecFrameSize;
-
- int countTime = 0; // for receive
-
- int threadSleep = 0;
-
- if (_codecSampleRate != 0)
- threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
- else
- threadSleep = _layerFrameSize;
-
- TimerPort::setTimer (threadSleep);
-
- _audiolayer->startStream();
-
- _audiolayer->getMainBuffer()->flush (_ca->getCallId());
-
- _session->startRunning();
-
- int timestamp = _session->getCurrentTimestamp(); // for mic
-
- _debug ("- ARTP Action: Start call %s",_ca->getCallId().c_str());
-
- while (!testCancel()) {
-
-
- // _debug("Main while loop for call: %s", _ca->getCallId().c_str());
- // Send session
- sessionWaiting = _session->isWaiting();
-
- sendSessionFromMic (timestamp);
- timestamp += timestep;
- // timestamp = _session->getCurrentTimestamp();
-
- // Recv session
- receiveSessionForSpkr (countTime);
-
- // Let's wait for the next transmit cycle
-
-
- if (sessionWaiting == 1) {
- // Record mic and speaker during conversation
- _ca->recAudio.recData (spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
- } else {
- // Record mic only while leaving a message
- _ca->recAudio.recData (micData,_nSamplesMic);
- }
-
-
- // Let's wait for the next transmit cycle
- Thread::sleep (TimerPort::getTimer());
-
-
- // TimerPort::incTimer(20); // 'frameSize' ms
- TimerPort::incTimer (threadSleep);
-
- }
-
- // _audiolayer->stopStream();
- _debug ("- ARTP Action: Stop call %s",_ca->getCallId().c_str());
-
-
-}
-
-
-// EOF
diff --git a/sflphone-common/src/audio/audiortp.h b/sflphone-common/src/audio/audiortp.h
deleted file mode 100644
index f9dc1671a9297b15cd79bab4a537b456340a370d..0000000000000000000000000000000000000000
--- a/sflphone-common/src/audio/audiortp.h
+++ /dev/null
@@ -1,299 +0,0 @@
-/*
- * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@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.
- */
-
-#ifndef __AUDIO_RTP_H__
-#define __AUDIO_RTP_H__
-
-#include <cstdio>
-#include <cstdlib>
-#include <iostream>
-#include <ccrtp/rtp.h>
-#include <cc++/numbers.h>
-
-#include "../global.h"
-// #include "plug-in/audiorecorder/audiorecord.h"
-#include "../samplerateconverter.h"
-#include "codecDescriptor.h"
-
-#include <fstream>
-
-#define UP_SAMPLING 0
-#define DOWN_SAMPLING 1
-
-
-
-class AudioRtpException: public std::exception
-{
- virtual const char* what() const throw()
- {
- return "AudioRtpException occured";
- }
-};
-
-
-/**
- * @file audiortp.h
- * @brief Manage the real-time data transport in a SIP call
- */
-
-
-class SIPCall;
-
-///////////////////////////////////////////////////////////////////////////////
-// Two pair of sockets
-///////////////////////////////////////////////////////////////////////////////
-class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
- public:
- /**
- * Constructor
- * @param sipcall The pointer on the SIP call
- * @param sym Tells whether or not the voip links are symmetric
- */
- AudioRtpRTX (SIPCall* sipcall, bool sym);
-
- /**
- * Destructor
- */
- ~AudioRtpRTX();
-
- /** For incoming call notification */
- ost::Time *time;
-
- /** Thread associated method */
- virtual void run ();
-
- /** A SIP call */
- SIPCall* _ca;
-
- /**
- * Update RTP session media info as received from SDP negociation
- */
- void setRtpSessionMedia(void);
-
- /**
- * Update RTP session remote ip destination as received from sip transaction
- */
- void setRtpSessionRemoteIp(void);
-
-
- friend class RtpTest;
-
- private:
-
- // copy constructor
- AudioRtpRTX(const AudioRtpRTX& rh);
-
- // assignment operator
- AudioRtpRTX& operator=(const AudioRtpRTX& rh);
-
- /** RTP session to send data */
- ost::RTPSession *_sessionSend;
-
- /** RTP session to receive data */
- ost::RTPSession *_sessionRecv;
-
- /** RTP symmetric session ( receive and send data in the same session ) */
- ost::SymmetricRTPSession *_session;
-
- /** Semaphore */
- // ost::Semaphore _start;
-
- /** Is the session symmetric or not */
- bool _sym;
-
- /** 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;
-
- /** audio layer */
- AudioLayer *_audiolayer;
-
- /** 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;
-
-
-
- bool _payloadIsSet;
- bool _remoteIpIsSet;
-
- /**
- * Init the RTP session. Create either symmetric or double sessions to manage data transport
- * Set the payloads according to the manager preferences
- */
- void initAudioRtpSession(void);
-
- /**
- * Return the lenth the codec frame in ms
- */
- float computeCodecFrameSize(int codecSamplePerFrame, int codecClockRate);
-
- /**
- * Compute nb of byte to get coresponding to X ms at audio layer frame size (44.1 khz)
- */
- int computeNbByteAudioLayer(float codecFrameSize);
-
-
- int processDataEncode();
-
-
- void processDataDecode(unsigned char* spkrData, unsigned int size, int& countTime);
-
-
- /**
- * Get the data from the mic, encode it and send it through the RTP session
- * @param timestamp To manage time and synchronizing
- */
- void sendSessionFromMic(int timestamp);
-
- /**
- * Get the data from the RTP packets, decode it and send it to the sound card
- * @param countTime To manage time and synchronizing
- */
- void receiveSessionForSpkr(int& countTime);
-
- /**
- * Init the buffers used for processing sound data
- */
- void initBuffers(void);
-
- /**
- * Call the appropriate function, up or downsampling
- * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
- * @param nbSamples Number of samples to process
- * @param status Type of resampling
- * UPSAMPLING
- * DOWNSAMPLING
- * @return int The number of samples after process
- */
- int reSampleData(SFLDataFormat *input, SFLDataFormat *output,int sampleRate_codec, int nbSamples, int status);
-
- /** The audio codec used during the session */
- AudioCodec *_audiocodec;
- // AudioCodec _audioCodecInstance;
-
- /** Mutex */
- ost::Mutex _rtpRtxMutex;
-
- public:
-
- std::fstream *rtp_input_rec;
- std::fstream *rtp_output_rec;
-
- static int count_rtp;
-
-
-};
-
-///////////////////////////////////////////////////////////////////////////////
-// Main class rtp
-///////////////////////////////////////////////////////////////////////////////
-class AudioRtp {
- public:
- /**
- * Constructor
- */
- AudioRtp();
-
- /**
- * Destructor
- */
- ~AudioRtp();
-
- /**
- * Create a brand new RTP session by calling the AudioRtpRTX constructor
- * @param ca A pointer on a SIP call
- */
- void createNewSession (SIPCall *ca);
-
- /**
- * Start the AudioRtpRTX thread created with createNewSession
- */
- int start(void);
-
- /**
- * Close a RTP session and kills the remaining threads
- */
- bool closeRtpSession( void );
-
- /**
- * Start recording
- */
- // void setRecording ();
-
- friend class RtpTest;
-
- /**
- *
- */
- inline AudioRtpRTX * getAudioRtpRtxThread(void) { return _RTXThread; }
- private:
- // copy constructor
- AudioRtp(const AudioRtp& rh);
-
- // assignment operator
- AudioRtp& operator=(const AudioRtp& rh);
-
- /** The RTP thread */
- AudioRtpRTX* _RTXThread;
-
- /** Symmetric session or not */
- bool _symmetric;
-
- /** Mutex */
- ost::Mutex _rtpMutex;
-
-};
-
-#endif // __AUDIO_RTP_H__
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index cb3480979ef3ec910d2cbc07588b68dc96966410..09756c4adff42163459b71e15d7f3cec7fc80c21 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -22,7 +22,6 @@
#include "AudioZrtpSession.h"
#include "AudioSrtpSession.h"
#include "AudioSymmetricRtpSession.h"
-
#include "manager.h"
#include "account.h"
#include "sip/sipcall.h"
@@ -122,7 +121,7 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
}
}
-void AudioRtpFactory::start (void)
+void AudioRtpFactory::start (AudioCodec* audiocodec)
{
if (_rtpSession == NULL) {
throw AudioRtpFactoryException ("RTP: Error: _rtpSession was null when trying to start audio thread");
@@ -131,7 +130,7 @@ void AudioRtpFactory::start (void)
switch (_rtpSessionType) {
case Sdes:
- if (static_cast<AudioSrtpSession *> (_rtpSession)->startRtpThread() != 0) {
+ if (static_cast<AudioSrtpSession *> (_rtpSession)->startRtpThread(audiocodec) != 0) {
throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSRtpSession thread");
}
break;
@@ -139,7 +138,7 @@ void AudioRtpFactory::start (void)
case Symmetric:
_debug ("Starting symmetric rtp thread");
- if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread() != 0) {
+ if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread(audiocodec) != 0) {
throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSymmetricRtpSession thread");
}
@@ -147,7 +146,7 @@ void AudioRtpFactory::start (void)
case Zrtp:
- if (static_cast<AudioZrtpSession *> (_rtpSession)->startRtpThread() != 0) {
+ if (static_cast<AudioZrtpSession *> (_rtpSession)->startRtpThread(audiocodec) != 0) {
throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioZrtpSession thread");
}
break;
@@ -210,6 +209,15 @@ void AudioRtpFactory::updateDestinationIpAddress (void)
}
}
+sfl::AudioSymmetricRtpSession * AudioRtpFactory::getAudioSymetricRtpSession()
+{
+ if ( (_rtpSessionType == Symmetric) && (_rtpSessionType != NULL)) {
+ return static_cast<AudioSymmetricRtpSession *> (_rtpSession);
+ } else {
+ throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in getAudioSymetricRtpSession");
+ }
+}
+
sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
{
if ( (_rtpSessionType == Zrtp) && (_rtpSessionType != NULL)) {
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index a1c93d2cdcfe9b78186188d45cbe442d2fc8879b..de16a31fe610b5ae2e27420f451d47bb87d0b7a9 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -23,6 +23,7 @@
#include <cc++/thread.h>
#include "sip/SdesNegotiator.h"
+#include "audio/codecs/audiocodec.h"
class SdesNegotiator;
class SIPCall;
@@ -30,6 +31,7 @@ class SIPCall;
namespace sfl {
class AudioZrtpSession;
class AudioSrtpSession;
+ class AudioSymmetricRtpSession;
}
namespace sfl {
@@ -76,7 +78,7 @@ namespace sfl {
* file. initAudioRtpSession must have been called prior to that.
* @param None
*/
- void start();
+ void start(AudioCodec*);
/**
* Stop the audio rtp thread of the type specified in the configuration
@@ -98,6 +100,8 @@ namespace sfl {
*/
inline void * getAudioRtpSession(void) { return _rtpSession; }
+ AudioSymmetricRtpSession * getAudioSymetricRtpSession();
+
/**
* @param None
* @return The internal audio rtp session type
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index cb0578beaa7a12d70b5efd781fd3fd29b07afcdb..9213eefad2698f2a266d60b27e54a6f0c187a86c 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -75,7 +75,7 @@ namespace sfl {
// Thread associated method
virtual void run ();
- int startRtpThread();
+ int startRtpThread(AudioCodec*);
/**
* Used mostly when receiving a reinvite
@@ -92,23 +92,24 @@ namespace sfl {
*/
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:
void initBuffers(void);
void setSessionTimeouts(void);
- void setSessionMedia(void);
+ void setSessionMedia(AudioCodec*);
void setDestinationIpAddress(void);
int processDataEncode(void);
void processDataDecode(unsigned char * spkrData, unsigned int size);
-
- 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);
- }
void sendMicData();
void receiveSpeakerData ();
@@ -206,7 +207,6 @@ namespace sfl {
protected:
SIPCall * _ca;
-
};
template <typename D>
@@ -307,17 +307,9 @@ namespace sfl {
}
template <typename D>
- void AudioRtpSession<D>::setSessionMedia(void)
+ void AudioRtpSession<D>::setSessionMedia(AudioCodec* audiocodec)
{
- assert(_ca);
-
- AudioCodecType pl = (AudioCodecType)_ca->getLocalSDP()->get_session_media()->getPayload();
- _audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl);
-
- if (_audiocodec == NULL) {
- _debug ("No audiocodec, can't init RTP media");
- throw AudioRtpSessionException();
- }
+ _audiocodec = audiocodec;
_debug ("Init audio RTP session: codec payload %i", _audiocodec->getPayload());
@@ -617,23 +609,20 @@ namespace sfl {
}
template <typename D>
- int AudioRtpSession<D>::startRtpThread ()
+ int AudioRtpSession<D>::startRtpThread (AudioCodec* audiocodec)
{
_debug("Starting main thread");
+ setSessionTimeouts();
+ setSessionMedia(audiocodec);
+ initBuffers();
return start(_mainloopSemaphore);
}
template <typename D>
void AudioRtpSession<D>::run ()
{
-
- setSessionTimeouts();
- setSessionMedia();
-
- initBuffers();
-
- // Timestamp must be initialized randomly
- _timestamp = static_cast<D*>(this)->getCurrentTimestamp();
+ // Timestamp must be initialized randomly
+ _timestamp = static_cast<D*>(this)->getCurrentTimestamp();
int sessionWaiting;
int threadSleep = 0;
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 022cdcbe0c2dd16cc5b97e6115570756b9cb7bd6..4d1e25a850a9f7fef5de7f0bf1aaed5395927bd3 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -3299,10 +3299,19 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
_debug("Sdes not initialized for this call\n");
}
+ assert(call->getLocalSDP());
+ assert(call->getLocalSDP()->get_session_media());
+
+ AudioCodecType pl = (AudioCodecType)call->getLocalSDP()->get_session_media()->getPayload();
+ AudioCodec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec(pl);
+
+ if (audiocodec == NULL)
+ _error ("SIP: No audiocodec found");
+
try {
call->setAudioStart (true);
- call->getAudioRtp()->start();
+ call->getAudioRtp()->start(audiocodec);
} catch (exception& rtpException) {
_debug ("%s", rtpException.what());
}