diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 9a58c7ec0a3d65f6b17a3f475400a7855cc1413f..319ba0d92626666004fe4da8aa5509e599129746 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -48,12 +48,12 @@ AudioRtp::AudioRtp() :_RTXThread(0), _symmetric(), _threadMutex()
}
AudioRtp::~AudioRtp (void) {
- delete _RTXThread; _RTXThread = 0;
+ delete _RTXThread; _RTXThread = 0;
}
int
AudioRtp::createNewSession (SIPCall *ca) {
-
+
ost::MutexLock m(_threadMutex);
// something should stop the thread before...
@@ -64,46 +64,46 @@ AudioRtp::createNewSession (SIPCall *ca) {
delete _RTXThread; _RTXThread = 0;
}
- // Start RTP Send/Receive threads
- _symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
- _RTXThread = new AudioRtpRTX (ca, _symmetric);
- try {
- if (_RTXThread->start() != 0) {
- _debug("! ARTP Failure: unable to start RTX Thread\n");
- return -1;
+ // Start RTP Send/Receive threads
+ _symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
+ _RTXThread = new AudioRtpRTX (ca, _symmetric);
+ try {
+ if (_RTXThread->start() != 0) {
+ _debug("! ARTP Failure: unable to start RTX Thread\n");
+ return -1;
+ }
+ } catch(...) {
+ _debugException("! ARTP Failure: when trying to start a thread");
+ throw;
}
- } catch(...) {
- _debugException("! ARTP Failure: when trying to start a thread");
- throw;
- }
- return 0;
+ return 0;
}
void
AudioRtp::closeRtpSession () {
- ost::MutexLock m(_threadMutex);
- // This will make RTP threads finish.
- _debug("Stopping AudioRTP\n");
- try {
-
- delete _RTXThread; _RTXThread = 0;
- } catch(...) {
- _debugException("! ARTP Exception: when stopping audiortp\n");
- throw;
- }
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- audiolayer->stopStream();
+ ost::MutexLock m(_threadMutex);
+ // This will make RTP threads finish.
+ _debug("Stopping AudioRTP\n");
+ try {
+
+ delete _RTXThread; _RTXThread = 0;
+ } catch(...) {
+ _debugException("! ARTP Exception: when stopping audiortp\n");
+ throw;
+ }
+ AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ audiolayer->stopStream();
}
void
AudioRtp::setRecording() {
-
- _debug("AudioRtp::setRecording\n");
- _RTXThread->_ca->setRecording();
+
+ _debug("AudioRtp::setRecording\n");
+ _RTXThread->_ca->setRecording();
}
@@ -113,152 +113,152 @@ AudioRtp::setRecording() {
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _ca(sipcall), _sessionSend(NULL), _sessionRecv(NULL), _session(NULL), _start(),
- _sym(sym), micData(NULL), micDataConverted(NULL), micDataEncoded(NULL), spkrDataDecoded(NULL), spkrDataConverted(NULL),
- converter(NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec(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());
- if (!_sym) {
- _sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _session = NULL;
- } else {
- _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
- _sessionRecv = NULL;
- _sessionSend = 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());
+ if (!_sym) {
+ _sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
+ _sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
+ _session = NULL;
+ } else {
+ _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
+ _sessionRecv = NULL;
+ _sessionSend = NULL;
+ }
}
AudioRtpRTX::~AudioRtpRTX () {
- _start.wait();
-
- try {
- this->terminate();
- } catch(...) {
- _debugException("! ARTP: Thread destructor didn't terminate correctly");
- throw;
- }
- _ca = 0;
- if (!_sym) {
- delete _sessionRecv; _sessionRecv = NULL;
- delete _sessionSend; _sessionSend = NULL;
- } else {
- delete _session; _session = NULL;
- }
-
- 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;
-
-}
-
- void
-AudioRtpRTX::initBuffers()
-{
- converter = new SamplerateConverter( _layerSampleRate , _layerFrameSize );
+ _start.wait();
- int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
+ try {
+ this->terminate();
+ } catch(...) {
+ _debugException("! ARTP: Thread destructor didn't terminate correctly");
+ throw;
+ }
+ _ca = 0;
+ if (!_sym) {
+ delete _sessionRecv; _sessionRecv = NULL;
+ delete _sessionSend; _sessionSend = NULL;
+ } else {
+ delete _session; _session = NULL;
+ }
- micData = new SFLDataFormat[nbSamplesMax];
- micDataConverted = new SFLDataFormat[nbSamplesMax];
- micDataEncoded = new unsigned char[nbSamplesMax];
+ delete [] micData; micData = NULL;
+ delete [] micDataConverted; micDataConverted = NULL;
+ delete [] micDataEncoded; micDataEncoded = NULL;
- spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
-}
+ delete [] spkrDataDecoded; spkrDataDecoded = NULL;
+ delete [] spkrDataConverted; spkrDataConverted = NULL;
- void
-AudioRtpRTX::initAudioRtpSession (void)
-{
- try {
- if (_ca == 0) { return; }
- _audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( _ca->getAudioCodec() );
- _codecSampleRate = _audiocodec->getClockRate();
-
- _debug("Init audio RTP session\n");
- ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
- if (!remote_ip) {
- _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
- return;
- }
+ delete time; time = NULL;
+ delete converter; converter = NULL;
- if (!_sym) {
- _sessionRecv->setSchedulingTimeout (10000);
- _sessionRecv->setExpireTimeout(1000000);
+}
- _sessionSend->setSchedulingTimeout(10000);
- _sessionSend->setExpireTimeout(1000000);
- } else {
- _session->setSchedulingTimeout(10000);
- _session->setExpireTimeout(1000000);
- }
+ void
+AudioRtpRTX::initBuffers()
+{
+ converter = new SamplerateConverter( _layerSampleRate , _layerFrameSize );
- if (!_sym) {
- if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
- return;
- }
- if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
- _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
- return;
- }
-
- 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 {
+ int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
- //_debug("AudioRTP Thread: Added session destination %s\n", remote_ip.getHostname() );
+ micData = new SFLDataFormat[nbSamplesMax];
+ micDataConverted = new SFLDataFormat[nbSamplesMax];
+ micDataEncoded = new unsigned char[nbSamplesMax];
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
- return;
- }
+ spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+}
- 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()));
- }
- }
+ void
+AudioRtpRTX::initAudioRtpSession (void)
+{
+ try {
+ if (_ca == 0) { return; }
+ _audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( _ca->getAudioCodec() );
+ _codecSampleRate = _audiocodec->getClockRate();
+
+ _debug("Init audio RTP session\n");
+ ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
+ if (!remote_ip) {
+ _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
+ return;
+ }
+
+
+ if (!_sym) {
+ _sessionRecv->setSchedulingTimeout (10000);
+ _sessionRecv->setExpireTimeout(1000000);
+
+ _sessionSend->setSchedulingTimeout(10000);
+ _sessionSend->setExpireTimeout(1000000);
+ } else {
+ _session->setSchedulingTimeout(10000);
+ _session->setExpireTimeout(1000000);
+ }
+
+ if (!_sym) {
+ if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
+ return;
+ }
+ if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
+ _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
+ return;
+ }
+
+ 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 {
+
+ //_debug("AudioRTP Thread: Added session destination %s\n", remote_ip.getHostname() );
+
+ if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
+ return;
+ }
+
+ 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()));
+ }
+ }
+ }
+
+
+ } catch(...) {
+ _debugException("! ARTP Failure: initialisation failed");
+ throw;
}
-
-
- } catch(...) {
- _debugException("! ARTP Failure: initialisation failed");
- throw;
- }
}
- void
+ 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
- //try {
+ // STEP:
+ // 1. get data from mic
+ // 2. convert it to int16 - good sample, good rate
+ // 3. encode it
+ // 4. send it
+ //try {
timestamp += time->getSecond();
if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
@@ -278,7 +278,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
//_debug("get data from mic\n");
int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
int nb_sample_up = nbSample;
-
+
// Store the length of the mic buffer in samples for recording
_nSamplesMic = nbSample;
@@ -288,32 +288,32 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
nbSample = reSampleData(_audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
- // fill end with 0...
- memset( micDataConverted + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
- nbSample = nbSamplesMax;
+ // fill end with 0...
+ memset( micDataConverted + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
+ nbSample = nbSamplesMax;
}
int compSize = _audiocodec->codecEncode( micDataEncoded , micDataConverted , nbSample*sizeof(int16));
// encode divise by two
// Send encoded audio sample over the network
if (compSize > nbSamplesMax) { _debug("! ARTP: %d should be %d\n", compSize, nbSamplesMax);}
if (!_sym) {
- _sessionSend->putData(timestamp, micDataEncoded, compSize);
+ _sessionSend->putData(timestamp, micDataEncoded, compSize);
} else {
- _session->putData(timestamp, micDataEncoded, compSize);
+ _session->putData(timestamp, micDataEncoded, compSize);
}
- /*} catch(...) {
- _debugException("! ARTP: sending failed");
- throw;
- }*/
+ /*} catch(...) {
+ _debugException("! ARTP: sending failed");
+ throw;
+ }*/
}
- void
+ void
AudioRtpRTX::receiveSessionForSpkr (int& countTime)
{
- if (_ca == 0) { return; }
- //try {
+ if (_ca == 0) { return; }
+ //try {
AudioLayer* audiolayer = Manager::instance().getAudioDriver();
if (!audiolayer) { return; }
@@ -321,13 +321,13 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
// Get audio data stream
if (!_sym) {
- adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
+ adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
} else {
- adu = _session->getData(_session->getFirstTimestamp());
+ adu = _session->getData(_session->getFirstTimestamp());
}
if (adu == NULL) {
- //_debug("No RTP audio stream\n");
- return;
+ //_debug("No RTP audio stream\n");
+ return;
}
//int payload = adu->getType(); // codec type
@@ -338,93 +338,93 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
unsigned int max = (unsigned int)(_codecSampleRate * _layerFrameSize / 1000);
if ( size > max ) {
- _debug("We have received from RTP a packet larger than expected: %d VS %d\n", size, max);
- _debug("The packet size has been cropped\n");
- size=max;
+ _debug("We have received from RTP a packet larger than expected: %d VS %d\n", size, max);
+ _debug("The packet size has been cropped\n");
+ size=max;
}
if (_audiocodec != NULL) {
-
-
- int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
- //buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
- int nbInt16 = expandedSize / sizeof(int16);
- //nbInt16 represents the number of samples we just decoded
- if ((unsigned int)nbInt16 > max) {
- _debug("We have decoded an RTP packet larger than expected: %d VS %d. Cropping.\n", nbInt16, max);
- nbInt16=max;
- }
- int nbSample = nbInt16;
-
- // Do sample rate conversion
- int nb_sample_down = nbSample;
- nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
+
+
+ int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
+ //buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
+ int nbInt16 = expandedSize / sizeof(int16);
+ //nbInt16 represents the number of samples we just decoded
+ if ((unsigned int)nbInt16 > max) {
+ _debug("We have decoded an RTP packet larger than expected: %d VS %d. Cropping.\n", nbInt16, max);
+ nbInt16=max;
+ }
+ int nbSample = nbInt16;
+
+ // Do sample rate conversion
+ int nb_sample_down = nbSample;
+ nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
#ifdef DATAFORMAT_IS_FLOAT
#else
#endif
-
- // Stor the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- //audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
- audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
-
-
- // 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();
- }
- }
+
+ // Stor the number of samples for recording
+ _nSamplesSpkr = nbSample;
+
+ //audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
+ audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
+
+
+ // 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();
+ countTime += time->getSecond();
}
delete adu; adu = NULL;
- //} catch(...) {
+ //} catch(...) {
//_debugException("! ARTP: receiving failed");
//throw;
- //}
+ //}
}
- int
+ int
AudioRtpRTX::reSampleData(int sampleRate_codec, int nbSamples, int status)
{
- if(status==UP_SAMPLING){
- return converter->upsampleData( spkrDataDecoded , spkrDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
- }
- else if(status==DOWN_SAMPLING){
- return converter->downsampleData( micData , micDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
- }
- else
- return 0;
+ if(status==UP_SAMPLING){
+ return converter->upsampleData( spkrDataDecoded , spkrDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
+ }
+ else if(status==DOWN_SAMPLING){
+ return converter->downsampleData( micData , micDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
+ }
+ else
+ return 0;
}
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;
-
- //try {
+ //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;
+
+ //try {
// Init the session
initAudioRtpSession();
step = (int) (_layerFrameSize * _codecSampleRate / 1000);
// start running the packet queue scheduler.
//_debug("AudioRTP Thread started\n");
if (!_sym) {
- _sessionRecv->startRunning();
- _sessionSend->startRunning();
+ _sessionRecv->startRunning();
+ _sessionSend->startRunning();
} else {
- _session->startRunning();
- //_debug("Session is now: %d active\n", _session->isActive());
+ _session->startRunning();
+ //_debug("Session is now: %d active\n", _session->isActive());
}
int timestamp = 0; // for mic
@@ -435,36 +435,25 @@ AudioRtpRTX::run () {
_start.post();
_debug("- ARTP Action: Start\n");
while (!testCancel()) {
- ////////////////////////////
- // Send session
- ////////////////////////////
- sendSessionFromMic(timestamp);
- timestamp += step;
- ////////////////////////////
- // Recv session
- ////////////////////////////
- receiveSessionForSpkr(countTime);
- // Let's wait for the next transmit cycle
-
- _ca->recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
-
- Thread::sleep(TimerPort::getTimer());
- TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
+ ////////////////////////////
+ // Send session
+ ////////////////////////////
+ sendSessionFromMic(timestamp);
+ timestamp += step;
+ ////////////////////////////
+ // Recv session
+ ////////////////////////////
+ receiveSessionForSpkr(countTime);
+ // Let's wait for the next transmit cycle
+
+ _ca->recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
+
+ Thread::sleep(TimerPort::getTimer());
+ TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
}
- // _debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
_debug("- ARTP Action: Stop\n");
- //} catch(std::exception &e) {
- //_start.post();
- //_debug("! ARTP: Stop %s\n", e.what());
- //throw;
- //} catch(...) {
- //_start.post();
- //_debugException("* ARTP Action: Stop");
- //throw;
- //}
-
}
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index a3566314fb0d4ad4a05f1b6ec961f0eac334d672..a8ae5937fa270ec0e251cc8babcf0838545cc8e9 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -25,20 +25,20 @@ int PulseLayer::streamState;
static void audioCallback ( pa_stream* s, size_t bytes, void* userdata )
{
- assert( s && bytes );
- assert( bytes > 0 );
- static_cast<PulseLayer*>(userdata)->processData();
+ assert( s && bytes );
+ assert( bytes > 0 );
+ static_cast<PulseLayer*>(userdata)->processData();
}
-
+
PulseLayer::PulseLayer(ManagerImpl* manager)
- : AudioLayer( manager , PULSEAUDIO )
- , context(NULL)
- , m(NULL)
- , playback()
- , record()
+ : AudioLayer( manager , PULSEAUDIO )
+ , context(NULL)
+ , m(NULL)
+ , playback()
+ , record()
{
- PulseLayer::streamState = 0;
- _debug("Pulse audio constructor: Create context\n");
+ PulseLayer::streamState = 0;
+ _debug("Pulse audio constructor: Create context\n");
}
// Destructor
@@ -50,324 +50,324 @@ PulseLayer::~PulseLayer (void)
pa_context_unref( context );
}
- void
+ void
PulseLayer::closeLayer( void )
{
- _debug(" Destroy pulselayer\n");
-
- playback->disconnect();
- record->disconnect();
-
- while(PulseLayer::streamState != 2)
- ;
- PulseLayer::streamState = 0;
+ _debug(" Destroy pulselayer\n");
+
+ playback->disconnect();
+ record->disconnect();
+
+ while(PulseLayer::streamState != 2)
+ ;
+ PulseLayer::streamState = 0;
- //TODO Remove this ugly hack
- sleep(2);
+ //TODO Remove this ugly hack
+ sleep(2);
}
- void
+ void
PulseLayer::connectPulseAudioServer( void )
{
- pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
+ pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
- pa_threaded_mainloop_lock( m );
+ pa_threaded_mainloop_lock( m );
- _debug("Connect the context to the server\n");
- pa_context_connect( context, NULL , flag , NULL );
+ _debug("Connect the context to the server\n");
+ pa_context_connect( context, NULL , flag , NULL );
- pa_context_set_state_callback(context, context_state_callback, this);
- pa_threaded_mainloop_wait( m );
+ pa_context_set_state_callback(context, context_state_callback, this);
+ pa_threaded_mainloop_wait( m );
- // Run the main loop
- if( pa_context_get_state( context ) != PA_CONTEXT_READY ){
- _debug("Error connecting to pulse audio server\n");
- pa_threaded_mainloop_unlock( m );
- }
+ // Run the main loop
+ if( pa_context_get_state( context ) != PA_CONTEXT_READY ){
+ _debug("Error connecting to pulse audio server\n");
+ pa_threaded_mainloop_unlock( m );
+ }
- pa_threaded_mainloop_unlock( m );
- //serverinfo();
- //muteAudioApps(99);
- _debug("Context creation done\n");
+ pa_threaded_mainloop_unlock( m );
+ //serverinfo();
+ //muteAudioApps(99);
+ _debug("Context creation done\n");
}
void PulseLayer::context_state_callback( pa_context* c, void* user_data )
{
- _debug("The state of the context changed\n");
- PulseLayer* pulse = (PulseLayer*)user_data;
- assert(c && pulse->m);
- switch(pa_context_get_state(c)){
- case PA_CONTEXT_CONNECTING:
- case PA_CONTEXT_AUTHORIZING:
- case PA_CONTEXT_SETTING_NAME:
- _debug("Waiting....\n");
- break;
- case PA_CONTEXT_READY:
- pulse->createStreams( c );
- _debug("Connection to PulseAudio server established\n");
- break;
- case PA_CONTEXT_TERMINATED:
- _debug("Context terminated\n");
- break;
- case PA_CONTEXT_FAILED:
- default:
- _debug(" Error : %s\n" , pa_strerror(pa_context_errno(c)));
- pulse->disconnectPulseAudioServer();
- exit(0);
- break;
- }
+ _debug("The state of the context changed\n");
+ PulseLayer* pulse = (PulseLayer*)user_data;
+ assert(c && pulse->m);
+ switch(pa_context_get_state(c)){
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ _debug("Waiting....\n");
+ break;
+ case PA_CONTEXT_READY:
+ pulse->createStreams( c );
+ _debug("Connection to PulseAudio server established\n");
+ break;
+ case PA_CONTEXT_TERMINATED:
+ _debug("Context terminated\n");
+ break;
+ case PA_CONTEXT_FAILED:
+ default:
+ _debug(" Error : %s\n" , pa_strerror(pa_context_errno(c)));
+ pulse->disconnectPulseAudioServer();
+ exit(0);
+ break;
+ }
}
void PulseLayer::disconnectPulseAudioServer( void )
{
- if( playback )
- delete playback; playback=NULL;
+ if( playback )
+ delete playback; playback=NULL;
- if( record )
- delete record; record=NULL;
+ if( record )
+ delete record; record=NULL;
}
- void
+ void
PulseLayer::createStreams( pa_context* c )
{
- playback = new AudioStream(c, PLAYBACK_STREAM, PLAYBACK_STREAM_NAME, _manager->getSpkrVolume());
- pa_stream_set_write_callback( playback->pulseStream() , audioCallback, this);
- //pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
- record = new AudioStream(c, CAPTURE_STREAM, CAPTURE_STREAM_NAME , _manager->getMicVolume());
- pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
- //pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
+ playback = new AudioStream(c, PLAYBACK_STREAM, PLAYBACK_STREAM_NAME, _manager->getSpkrVolume());
+ pa_stream_set_write_callback( playback->pulseStream() , audioCallback, this);
+ //pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
+ record = new AudioStream(c, CAPTURE_STREAM, CAPTURE_STREAM_NAME , _manager->getMicVolume());
+ pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
+ //pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
- pa_threaded_mainloop_signal(m , 0);
+ pa_threaded_mainloop_signal(m , 0);
}
- bool
+ bool
PulseLayer::openDevice(int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED)
{
- _sampleRate = sampleRate;
- _frameSize = frameSize;
+ _sampleRate = sampleRate;
+ _frameSize = frameSize;
- m = pa_threaded_mainloop_new();
- assert(m);
+ m = pa_threaded_mainloop_new();
+ assert(m);
- if( pa_threaded_mainloop_start( m ) < 0 ){
- _debug("Failed starting the mainloop\n");
- }
+ if( pa_threaded_mainloop_start( m ) < 0 ){
+ _debug("Failed starting the mainloop\n");
+ }
- // Instanciate a context
- if( !(context = pa_context_new( pa_threaded_mainloop_get_api( m ) , "SFLphone" )))
- _debug("Error while creating the context\n");
+ // Instanciate a context
+ if( !(context = pa_context_new( pa_threaded_mainloop_get_api( m ) , "SFLphone" )))
+ _debug("Error while creating the context\n");
- assert(context);
+ assert(context);
- connectPulseAudioServer();
+ connectPulseAudioServer();
- _debug("Connection Done!! \n");
- return true;
+ _debug("Connection Done!! \n");
+ return true;
}
- void
+ void
PulseLayer::closeCaptureStream( void )
{
}
- void
+ void
PulseLayer::closePlaybackStream( void )
{
}
- int
+ int
PulseLayer::canGetMic()
{
- if( record )
- return _micRingBuffer.AvailForGet();
- else
- return 0;
+ if( record )
+ return _micRingBuffer.AvailForGet();
+ else
+ return 0;
}
- int
+ int
PulseLayer::getMic(void *buffer, int toCopy)
{
- if( record ){
- return _micRingBuffer.Get(buffer, toCopy, 100);
- }
- else
- return 0;
+ if( record ){
+ return _micRingBuffer.Get(buffer, toCopy, 100);
+ }
+ else
+ return 0;
}
- void
+ void
PulseLayer::startStream (void)
{
- _micRingBuffer.flush();
- _debug("Start stream\n");
- pa_threaded_mainloop_lock(m);
- pa_stream_cork( record->pulseStream(), NULL, NULL, NULL);
- pa_threaded_mainloop_unlock(m);
+ flushMic();
+ _debug("Start stream\n");
+ pa_threaded_mainloop_lock(m);
+ pa_stream_cork( record->pulseStream(), NULL, NULL, NULL);
+ pa_threaded_mainloop_unlock(m);
}
- void
+ void
PulseLayer::stopStream (void)
{
- _debug("Stop stream\n");
- pa_stream_flush( playback->pulseStream(), NULL, NULL );
- pa_stream_flush( record->pulseStream(), NULL, NULL );
- flushMic();
- flushMain();
- flushUrgent();
+ _debug("Stop stream\n");
+ pa_stream_flush( playback->pulseStream(), NULL, NULL );
+ pa_stream_flush( record->pulseStream(), NULL, NULL );
+ flushMic();
+ flushMain();
+ flushUrgent();
}
- void
+ void
PulseLayer::underflow ( pa_stream* s UNUSED, void* userdata UNUSED )
{
- _debug("Buffer Underflow\n");
+ _debug("Buffer Underflow\n");
}
- void
+ void
PulseLayer::overflow ( pa_stream* s, void* userdata UNUSED )
{
- //PulseLayer* pulse = (PulseLayer*) userdata;
- pa_stream_drop( s );
- pa_stream_trigger( s, NULL, NULL);
+ //PulseLayer* pulse = (PulseLayer*) userdata;
+ pa_stream_drop( s );
+ pa_stream_trigger( s, NULL, NULL);
}
- void
+ void
PulseLayer::processData( void )
{
- // Handle the mic
- // We check if the stream is ready
- if( (record->pulseStream()) && pa_stream_get_state( record->pulseStream()) == PA_STREAM_READY)
- readFromMic();
+ // Handle the mic
+ // We check if the stream is ready
+ if( (record->pulseStream()) && pa_stream_get_state( record->pulseStream()) == PA_STREAM_READY)
+ readFromMic();
- // Handle the data for the speakers
- if( (playback->pulseStream()) && pa_stream_get_state( playback->pulseStream()) == PA_STREAM_READY){
+ // Handle the data for the speakers
+ if( (playback->pulseStream()) && pa_stream_get_state( playback->pulseStream()) == PA_STREAM_READY){
- // If the playback buffer is full, we don't overflow it; wait for it to have free space
- if( pa_stream_writable_size(playback->pulseStream()) == 0 )
- return;
+ // If the playback buffer is full, we don't overflow it; wait for it to have free space
+ if( pa_stream_writable_size(playback->pulseStream()) == 0 )
+ return;
- writeToSpeaker();
- }
+ writeToSpeaker();
+ }
}
-void
+ void
PulseLayer::writeToSpeaker( void )
{
- /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
- int urgentAvail;
- /** Bytes available in the regular ringbuffer ( reserved for voice ) */
- int normalAvail;
- int toGet;
- int toPlay;
-
- SFLDataFormat* out;// = (SFLDataFormat*)pa_xmalloc(framesPerBuffer);
- urgentAvail = _urgentRingBuffer.AvailForGet();
-
- if (urgentAvail > 0) {
- // Urgent data (dtmf, incoming call signal) come first.
- //_debug("Play urgent!: %i\e" , urgentAvail);
- toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
- out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
- _urgentRingBuffer.Get(out, toGet, 100);
- pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
- // Consume the regular one as well (same amount of bytes)
- _voiceRingBuffer.Discard(toGet);
- }
- else
- {
- AudioLoop* tone = _manager->getTelephoneTone();
- if ( tone != 0) {
- toGet = framesPerBuffer;
- out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
- tone->getNext(out, toGet , 100);
- pa_stream_write( playback->pulseStream() , out , toGet * sizeof(SFLDataFormat) , pa_xfree, 0 , PA_SEEK_RELATIVE);
- }
- if ( (tone=_manager->getTelephoneFile()) != 0 ) {
- toGet = framesPerBuffer;
- toPlay = ( (int)(toGet * sizeof(SFLDataFormat)) > framesPerBuffer )? framesPerBuffer : toGet * sizeof(SFLDataFormat) ;
- out = (SFLDataFormat*)pa_xmalloc(toPlay);
- tone->getNext(out, toPlay/2 , 100);
- pa_stream_write( playback->pulseStream() , out , toPlay , pa_xfree, 0 , PA_SEEK_RELATIVE) ;
- }
- else {
- out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
- normalAvail = _voiceRingBuffer.AvailForGet();
- toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
- if (toGet) {
- _voiceRingBuffer.Get(out, toGet, 100);
- _voiceRingBuffer.Discard(toGet);
- }
- else {
- bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
- }
- pa_stream_write( playback->pulseStream() , out , toGet , NULL, 0 , PA_SEEK_RELATIVE);
- pa_xfree(out);
+ /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
+ int urgentAvail;
+ /** Bytes available in the regular ringbuffer ( reserved for voice ) */
+ int normalAvail;
+ int toGet;
+ int toPlay;
+
+ SFLDataFormat* out;// = (SFLDataFormat*)pa_xmalloc(framesPerBuffer);
+ urgentAvail = _urgentRingBuffer.AvailForGet();
+
+ if (urgentAvail > 0) {
+ // Urgent data (dtmf, incoming call signal) come first.
+ //_debug("Play urgent!: %i\e" , urgentAvail);
+ toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
+ _urgentRingBuffer.Get(out, toGet, 100);
+ pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ // Consume the regular one as well (same amount of bytes)
+ _voiceRingBuffer.Discard(toGet);
+ }
+ else
+ {
+ AudioLoop* tone = _manager->getTelephoneTone();
+ if ( tone != 0) {
+ toGet = framesPerBuffer;
+ out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
+ tone->getNext(out, toGet , 100);
+ pa_stream_write( playback->pulseStream() , out , toGet * sizeof(SFLDataFormat) , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ }
+ if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ toGet = framesPerBuffer;
+ toPlay = ( (int)(toGet * sizeof(SFLDataFormat)) > framesPerBuffer )? framesPerBuffer : toGet * sizeof(SFLDataFormat) ;
+ out = (SFLDataFormat*)pa_xmalloc(toPlay);
+ tone->getNext(out, toPlay/2 , 100);
+ pa_stream_write( playback->pulseStream() , out , toPlay , pa_xfree, 0 , PA_SEEK_RELATIVE) ;
+ }
+ else {
+ out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
+ normalAvail = _voiceRingBuffer.AvailForGet();
+ toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ if (toGet) {
+ _voiceRingBuffer.Get(out, toGet, 100);
+ _voiceRingBuffer.Discard(toGet);
+ }
+ else {
+ bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
+ }
+ pa_stream_write( playback->pulseStream() , out , toGet , NULL, 0 , PA_SEEK_RELATIVE);
+ pa_xfree(out);
+ }
}
- }
}
-
+
void PulseLayer::readFromMic( void )
{
- const char* data;
- size_t r;
+ const char* data;
+ size_t r;
- if( pa_stream_peek( record->pulseStream() , (const void**)&data , &r ) < 0 || !data ){
- //_debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
- }
+ if( pa_stream_peek( record->pulseStream() , (const void**)&data , &r ) < 0 || !data ){
+ //_debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ }
- if( data != 0 ){
- _micRingBuffer.Put( (void*)data ,r, 100);
- }
+ if( data != 0 ){
+ _micRingBuffer.Put( (void*)data ,r, 100);
+ }
- if( pa_stream_drop( record->pulseStream() ) < 0 ) {
- //_debug("pa_stream_drop() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
- }
+ if( pa_stream_drop( record->pulseStream() ) < 0 ) {
+ //_debug("pa_stream_drop() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ }
}
static void retrieve_server_info(pa_context *c UNUSED, const pa_server_info *i, void *userdata UNUSED)
{
- _debug("Server Info: Process owner : %s\n" , i->user_name);
- _debug("\t\tServer name : %s - Server version = %s\n" , i->server_name, i->server_version);
- _debug("\t\tDefault sink name : %s\n" , i->default_sink_name);
- _debug("\t\tDefault source name : %s\n" , i->default_source_name);
+ _debug("Server Info: Process owner : %s\n" , i->user_name);
+ _debug("\t\tServer name : %s - Server version = %s\n" , i->server_name, i->server_version);
+ _debug("\t\tDefault sink name : %s\n" , i->default_sink_name);
+ _debug("\t\tDefault source name : %s\n" , i->default_source_name);
}
static void reduce_sink_list_cb(pa_context *c UNUSED, const pa_sink_input_info *i, int eol, void *userdata)
{
- PulseLayer* pulse = (PulseLayer*) userdata;
- if( !eol ){
- //_debug("Sink Info: index : %i\n" , i->index);
- //_debug("\t\tClient : %i\n" , i->client);
- //_debug("\t\tVolume : %i\n" , i->volume.values[0]);
- //_debug("\t\tChannels : %i\n" , i->volume.channels);
- if( strcmp( i->name , PLAYBACK_STREAM_NAME ) != 0)
- pulse->setSinkVolume( i->index , i->volume.channels, 10 );
- }
+ PulseLayer* pulse = (PulseLayer*) userdata;
+ if( !eol ){
+ //_debug("Sink Info: index : %i\n" , i->index);
+ //_debug("\t\tClient : %i\n" , i->client);
+ //_debug("\t\tVolume : %i\n" , i->volume.values[0]);
+ //_debug("\t\tChannels : %i\n" , i->volume.channels);
+ if( strcmp( i->name , PLAYBACK_STREAM_NAME ) != 0)
+ pulse->setSinkVolume( i->index , i->volume.channels, 10 );
+ }
}
static void restore_sink_list_cb(pa_context *c UNUSED, const pa_sink_input_info *i, int eol, void *userdata)
{
- PulseLayer* pulse = (PulseLayer*) userdata;
- if( !eol ){
- //_debug("Sink Info: index : %i\n" , i->index);
- //_debug("\t\tSink name : -%s-\n" , i->name);
- //_debug("\t\tClient : %i\n" , i->client);
- //_debug("\t\tVolume : %i\n" , i->volume.values[0]);
- //_debug("\t\tChannels : %i\n" , i->volume.channels);
- if( strcmp( i->name , PLAYBACK_STREAM_NAME ) != 0)
- pulse->setSinkVolume( i->index , i->volume.channels, 100);
- }
+ PulseLayer* pulse = (PulseLayer*) userdata;
+ if( !eol ){
+ //_debug("Sink Info: index : %i\n" , i->index);
+ //_debug("\t\tSink name : -%s-\n" , i->name);
+ //_debug("\t\tClient : %i\n" , i->client);
+ //_debug("\t\tVolume : %i\n" , i->volume.values[0]);
+ //_debug("\t\tChannels : %i\n" , i->volume.channels);
+ if( strcmp( i->name , PLAYBACK_STREAM_NAME ) != 0)
+ pulse->setSinkVolume( i->index , i->volume.channels, 100);
+ }
}
static void set_playback_volume_cb(pa_context *c UNUSED, const pa_sink_input_info *i, int eol, void *userdata)
{
PulseLayer* pulse;
int volume;
-
+
pulse = (PulseLayer*) userdata;
volume = pulse->getSpkrVolume();
@@ -381,7 +381,7 @@ static void set_capture_volume_cb(pa_context *c UNUSED, const pa_source_output_i
{
PulseLayer* pulse;
int volume;
-
+
pulse = (PulseLayer*) userdata;
volume = pulse->getMicVolume();
@@ -391,45 +391,45 @@ static void set_capture_volume_cb(pa_context *c UNUSED, const pa_source_output_i
}
}
- void
+ void
PulseLayer::reducePulseAppsVolume( void )
{
- pa_context_get_sink_input_info_list( context , reduce_sink_list_cb , this );
+ pa_context_get_sink_input_info_list( context , reduce_sink_list_cb , this );
}
- void
+ void
PulseLayer::restorePulseAppsVolume( void )
{
- pa_context_get_sink_input_info_list( context , restore_sink_list_cb , this );
+ pa_context_get_sink_input_info_list( context , restore_sink_list_cb , this );
}
- void
+ void
PulseLayer::serverinfo( void )
{
- pa_context_get_server_info( context , retrieve_server_info , NULL );
+ pa_context_get_server_info( context , retrieve_server_info , NULL );
}
void PulseLayer::setSinkVolume( int index, int channels, int volume )
{
-
+
pa_cvolume cvolume;
pa_volume_t vol = PA_VOLUME_NORM * ((double)volume / 100) ;
pa_cvolume_set( &cvolume , channels , vol);
- _debug("Set sink volume of index %i\n" , index);
- pa_context_set_sink_input_volume( context, index, &cvolume, NULL, NULL) ;
+ _debug("Set sink volume of index %i\n" , index);
+ pa_context_set_sink_input_volume( context, index, &cvolume, NULL, NULL) ;
}
void PulseLayer::setSourceVolume( int index, int channels, int volume )
{
-
+
pa_cvolume cvolume;
pa_volume_t vol = PA_VOLUME_NORM * ((double)volume / 100) ;
pa_cvolume_set( &cvolume , channels , vol);
- _debug("Set source volume of index %i\n" , index);
+ _debug("Set source volume of index %i\n" , index);
pa_context_set_source_volume_by_index(context, index, &cvolume, NULL, NULL);
}
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index e7f8ec57685ce46cadd8436c1067c48a3b81a73d..f5c65eea29615baf338a9346a120ab5c87d06fd1 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -429,7 +429,6 @@ IAXVoIPLink::answer(const CallID& id)
call->setConnectionState(Call::Connected);
// Start audio
audiolayer->startStream();
- audiolayer->flushMic();
return true;
}
@@ -701,7 +700,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
}
Manager::instance().peerAnsweredCall(id);
- audiolayer->flushMic();
// start audio here?
} else {
// deja connecté ?
@@ -805,7 +803,7 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
// resample
nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample_);
- //audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+ /* Write the data to the mic ring buffer */
audiolayer->putMain (spkrDataConverted , nbInt16 * sizeof(SFLDataFormat));
} else {