diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index c1c8e00f1e135548ba1552b9cd0816b63f597c33..7eb496a7112b9b14d35136de66f4c9d2dd647d75 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -109,7 +109,7 @@ AudioRtp::closeRtpSession () {
_debugException("! ARTP Exception: when stopping audiortp\n");
throw;
}
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
// audiolayer->stopStream();
_debug("AudioRtp::Audio rtp stopped\n");
@@ -150,16 +150,22 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _
// 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());
- if (!_sym) {
- _sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _session = NULL;
- } else {
- _debug ("%i\n", _ca->getLocalAudioPort());
- _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
- _sessionRecv = NULL;
- _sessionSend = NULL;
- }
+
+ _debug ("%i\n", _ca->getLocalAudioPort());
+ _session = new ost::SymmetricRTPSession (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();
+
}
AudioRtpRTX::~AudioRtpRTX () {
@@ -175,13 +181,10 @@ AudioRtpRTX::~AudioRtpRTX () {
throw;
}
_ca = 0;
- if (!_sym) {
- delete _sessionRecv; _sessionRecv = NULL;
- delete _sessionSend; _sessionSend = NULL;
- } else {
- _debug("Delete ost::RTPSession in AudioRtpRTX\n");
- delete _session; _session = NULL;
- }
+
+ _debug("Delete ost::RTPSession in AudioRtpRTX\n");
+ delete _session; _session = NULL;
+
_debug("Just killed AudioRtpRTX rtp sessions\n");
@@ -241,16 +244,10 @@ AudioRtpRTX::initAudioRtpSession (void)
}
*/
- if (!_sym) {
- _sessionRecv->setSchedulingTimeout (10000);
- _sessionRecv->setExpireTimeout(1000000);
-
- _sessionSend->setSchedulingTimeout(10000);
- _sessionSend->setExpireTimeout(1000000);
- } else {
- _session->setSchedulingTimeout(10000);
- _session->setExpireTimeout(1000000);
- }
+
+ _session->setSchedulingTimeout(10000);
+ _session->setExpireTimeout(1000000);
+
// setRtpSessionRemoteIp();
@@ -305,84 +302,59 @@ void
AudioRtpRTX::setRtpSessionMedia(void)
{
- _debug("setRtpSessionMedia()\n");
-
- if (_ca == 0) { return; }
+ _debug("**************************** setRtpSessionMedia() *****************************\n");
-
_audiocodec = _ca->getLocalSDP()->get_session_media ();
- _debug("AudioRtp: _audiocodec->getPayload() %i\n", _audiocodec->getPayload());
+ if (_ca == 0) { _debug(" !ARTP: No call, can't init RTP media\n"); return; }
+
+ if (_audiocodec == NULL) { _debug(" !ARTP: No audiocodec, can't init RTP media\n"); return; }
+
+ _debug("Init audio RTP session: codec payload %i\n", _audiocodec->getPayload());
if (_audiocodec == NULL) { return; }
_codecSampleRate = _audiocodec->getClockRate();
_codecFrameSize = _audiocodec->getFrameSize();
- if (!_sym) {
-
- bool payloadIsSet = false;
- if (_audiocodec) {
- if (_audiocodec->hasDynamicPayload()) {
- payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else {
- payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
- payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
- }
- }
- _sessionSend->setMark(true);
- } else {
+
- bool payloadIsSet = false;
- if (_audiocodec) {
- if (_audiocodec->hasDynamicPayload()) {
- payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
- } else {
- payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
- }
- }
+ bool payloadIsSet = false;
+
+ if (_audiocodec->hasDynamicPayload()) {
+ payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
+ }
+ else
+ {
+ payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
}
+
+
}
void
AudioRtpRTX::setRtpSessionRemoteIp(void)
{
- _debug("setRtpSessionRemoteIp()\n");
+ _debug("**************************** setRtpSessionRemoteIp() *****************************\n");
- if (_ca == 0) {
- _debug("Return no call \n");
- return;
- }
+ if (_ca == 0) { _debug(" !ARTP: No call, can't init RTP media \n"); return; }
ost::InetHostAddress remote_ip(_ca->getLocalSDP()->get_remote_ip().c_str());
- _debug("Init audio RTP session %s\n", _ca->getLocalSDP()->get_remote_ip().data());
- if (!remote_ip) {
- _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getLocalSDP()->get_remote_ip().data());
+ _debug("Init audio RTP session: remote ip %s\n", _ca->getLocalSDP()->get_remote_ip().data());
+ if (!remote_ip)
+ {
+ _debug(" !ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getLocalSDP()->get_remote_ip().data());
return;
}
-
- if (!_sym) {
-
- if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getLocalSDP()->get_remote_audio_port()) ) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getLocalSDP()->get_remote_audio_port());
- return;
- }
- if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getLocalSDP()->get_remote_audio_port())) {
- _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getLocalSDP()->get_remote_audio_port());
- return;
- }
-
- _sessionSend->setMark(true);
- } else {
-
- _debug("AudioRtp: _ca->getLocalSDP()->get_remote_audio_port() %i\n", (unsigned short)_ca->getLocalSDP()->get_remote_audio_port());
- if (!_session->addDestination (remote_ip, (unsigned short)_ca->getLocalSDP()->get_remote_audio_port() )) {
- return;
- }
+ if (!_session->addDestination (remote_ip, (unsigned short)_ca->getLocalSDP()->get_remote_audio_port() ))
+ {
+ return;
}
+
+
}
@@ -401,7 +373,7 @@ AudioRtpRTX::computeNbByteAudioLayer(float codecFrameSize)
int
-AudioRtpRTX::processDataEncode(AudioLayer* audiolayer)
+AudioRtpRTX::processDataEncode()
{
// compute codec framesize in ms
@@ -411,7 +383,7 @@ AudioRtpRTX::processDataEncode(AudioLayer* audiolayer)
int maxBytesToGet = computeNbByteAudioLayer(fixed_codec_framesize);
// available bytes inside ringbuffer
- int availBytesFromMic = audiolayer->canGetMic();
+ int availBytesFromMic = _audiolayer->canGetMic();
// set available byte to maxByteToGet
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
@@ -420,7 +392,7 @@ AudioRtpRTX::processDataEncode(AudioLayer* audiolayer)
return 0;
// Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
+ int nbSample = _audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
// nb bytes to be sent over RTP
int compSize = 0;
@@ -453,7 +425,7 @@ AudioRtpRTX::processDataEncode(AudioLayer* audiolayer)
void
-AudioRtpRTX::processDataDecode(AudioLayer* audiolayer, unsigned char* spkrData, unsigned int size, int& countTime)
+AudioRtpRTX::processDataDecode(unsigned char* spkrData, unsigned int size, int& countTime)
{
if (_audiocodec != NULL) {
@@ -475,7 +447,7 @@ AudioRtpRTX::processDataDecode(AudioLayer* audiolayer, unsigned char* spkrData,
_nSamplesSpkr = nbSample;
// put data in audio layer, size in byte
- audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
+ _audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
} else {
@@ -483,7 +455,7 @@ AudioRtpRTX::processDataDecode(AudioLayer* audiolayer, unsigned char* spkrData,
_nSamplesSpkr = nbSample;
// put data in audio layer, size in byte
- audiolayer->putMain (spkrDataDecoded, nbSample * sizeof(SFLDataFormat));
+ _audiolayer->putMain (spkrDataDecoded, nbSample * sizeof(SFLDataFormat));
}
// Notify (with a beep) an incoming call when there is already a call
@@ -514,21 +486,21 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
// no call, so we do nothing
if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; }
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }
+ // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ if (!_audiolayer) { _debug(" !ARTP: No audiolayer available for MIC\n"); return; }
- if (!_audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
+ if (!_audiocodec) { _debug(" !ARTP: No audiocodec available for MIC\n"); return; }
- int compSize = processDataEncode(audiolayer);
+ int compSize = processDataEncode();
// putData put the data on RTP queue, sendImmediate bypass this queue
if (!_sym) {
- // _sessionSend->putData(timestamp, micDataEncoded, compSize);
- _sessionSend->sendImmediate(timestamp, micDataEncoded, compSize);
+ _sessionSend->putData(timestamp, micDataEncoded, compSize);
+ // _sessionSend->sendImmediate(timestamp, micDataEncoded, compSize);
} else {
- // _session->putData(timestamp, micDataEncoded, compSize);
- _session->sendImmediate(timestamp, micDataEncoded, compSize);
+ _session->putData(timestamp, micDataEncoded, compSize);
+ // _session->sendImmediate(timestamp, micDataEncoded, compSize);
}
}
@@ -541,11 +513,13 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
if (_ca == 0) { return; }
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- if (!audiolayer) { return; }
+ // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- const ost::AppDataUnit* adu = NULL;
+ if (!_audiolayer) { _debug(" !ARTP: No audiolayer available for SPEAKER\n"); return; }
+
+ if (!_audiocodec) { _debug(" !ARTP: No audiocodec available for SPEAKER\n"); return; }
+ const ost::AppDataUnit* adu = NULL;
if (!_sym) {
adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
@@ -560,7 +534,7 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
unsigned char* spkrData = (unsigned char*)adu->getData(); // data in char
unsigned int size = adu->getSize(); // size in char
- processDataDecode(audiolayer, spkrData, size, countTime);
+ processDataDecode(spkrData, size, countTime);
}
@@ -583,38 +557,22 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
void
AudioRtpRTX::run () {
- //mic, we receive from soundcard in stereo, and we send encoded
- //encoding before sending
- AudioLayer *audiolayer = Manager::instance().getAudioDriver();
- _layerFrameSize = audiolayer->getFrameSize(); // en ms
- _layerSampleRate = audiolayer->getSampleRate();
-
- initBuffers();
- int step;
-
- int sessionWaiting;
-
- //try {
-
- // Init the session
- initAudioRtpSession();
+ /*
+ //mic, we receive from soundcard in stereo, and we send encoded
+ //encoding before sending
+ AudioLayer *audiolayer = Manager::instance().getAudioDriver();
+ _layerFrameSize = audiolayer->getFrameSize(); // en ms
+ _layerSampleRate = audiolayer->getSampleRate();
+ */
+ int sessionWaiting;
+
+ _session->startRunning();
+ //_debug("Session is now: %d active\n", _session->isActive());
- // step = (int) (_layerFrameSize * _codecSampleRate / 1000);
- step = _codecFrameSize;
- // start running the packet queue scheduler.
- //_debug("AudioRTP Thread started\n");
- if (!_sym) {
- _sessionRecv->startRunning();
- _sessionSend->startRunning();
- } else {
- _session->startRunning();
- //_debug("Session is now: %d active\n", _session->isActive());
- }
+ int timestep = _codecFrameSize;
int timestamp = 0; // for mic
- // step = (int) (_layerFrameSize * _codecSampleRate / 1000);
- step = _codecFrameSize;
int countTime = 0; // for receive
@@ -622,11 +580,11 @@ AudioRtpRTX::run () {
if (_codecSampleRate != 0)
threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
else
- threadSleep = _layerFrameSize;
+ threadSleep = _layerFrameSize;
TimerPort::setTimer(threadSleep);
- audiolayer->startStream();
+ _audiolayer->startStream();
_start.post();
_debug("- ARTP Action: Start call %s\n",_ca->getCallId().c_str());
while (!testCancel()) {
@@ -643,7 +601,7 @@ AudioRtpRTX::run () {
sessionWaiting = _session->isWaiting();
sendSessionFromMic(timestamp);
- timestamp += step;
+ timestamp += timestep;
////////////////////////////
// Recv session
diff --git a/sflphone-common/src/audio/audiortp.h b/sflphone-common/src/audio/audiortp.h
index 17bfdb6569773dccbe4e51139d88266ee22308b2..56345de0c99d70b840fa5f15700dd7d22c140745 100644
--- a/sflphone-common/src/audio/audiortp.h
+++ b/sflphone-common/src/audio/audiortp.h
@@ -133,6 +133,9 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/** Sample rate converter object */
SamplerateConverter* converter;
+ /** audio layer */
+ AudioLayer *_audiolayer;
+
/** Variables to process audio stream: sample rate for playing sound (typically 44100HZ) */
int _layerSampleRate;
@@ -178,10 +181,10 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
int computeNbByteAudioLayer(float codecFrameSize);
- int processDataEncode(AudioLayer* audiolayer);
+ int processDataEncode();
- void processDataDecode(AudioLayer* audiolayer, unsigned char* spkrData, unsigned int size, int& countTime);
+ void processDataDecode(unsigned char* spkrData, unsigned int size, int& countTime);
/**
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index 30080fad7921d8dfa7fe65ddb1d06d19b6e76806..567ec7f181bca74ca1a39500d99b4103ac250f56 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -2097,9 +2097,13 @@ void call_on_media_update ( pjsip_inv_session *inv, pj_status_t status )
if (call->getState() != Call::Hold)
{
+ _debug("Set parameters in RTP session\n");
link->_audiortp->getRTX()->setRtpSessionMedia();
link->_audiortp->getRTX()->setRtpSessionRemoteIp();
}
+ else {
+ _debug("Didn't set RTP parameters since call is on hold\n");
+ }
}