From 00fc6a2124fe7c2eab32012feb985c0ab3b47f9f Mon Sep 17 00:00:00 2001
From: asavard <asavard@asavard-KT378AA-A2L-a6552f.(none)>
Date: Wed, 27 Oct 2010 11:37:12 -0400
Subject: [PATCH] [#4367] Implement RTP main loop
---
.../audio/audiortp/AudioRtpRecordHandler.cpp | 48 +++---
.../src/audio/audiortp/AudioRtpSession.cpp | 146 +++++++++++-------
.../src/audio/audiortp/AudioRtpSession.h | 8 +-
3 files changed, 117 insertions(+), 85 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index aa452d9308..7995247659 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -49,7 +49,6 @@ AudioRtpRecord::AudioRtpRecord(ManagerImpl *manager) : _audioCodec(NULL)
, _codecSampleRate(0)
, _audioLayerFrameSize(0)
, _codecFrameSize(0)
- , _converterSamplingRate(0)
, _micFadeInComplete(false)
, _spkrFadeInComplete(false)
, _micAmplFactor(initFadeinFactor)
@@ -149,11 +148,6 @@ SamplerateConverter *AudioRtpRecord::getConverter() const
return _converter;
}
-int AudioRtpRecord::getConverterSamplingRate() const
-{
- return _converterSamplingRate;
-}
-
EventQueue *AudioRtpRecord::getEventQueue()
{
return &_eventQueue;
@@ -254,10 +248,6 @@ void AudioRtpRecord::setConverter(SamplerateConverter *_converter)
this->_converter = _converter;
}
-void AudioRtpRecord::setConverterSamplingRate(int _converterSamplingRate)
-{
- this->_converterSamplingRate = _converterSamplingRate;
-}
void AudioRtpRecord::setEventQueue(EventQueue _eventQueue)
{
@@ -355,9 +345,6 @@ void AudioRtpRecordHandler::initBuffers()
// Set sampling rate, main buffer choose the highest one
Manager::instance().getAudioDriver()->getMainBuffer()->setInternalSamplingRate (getCodecSampleRate());
- // may be different than one already set
- // _converterSamplingRate = Manager::instance().getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
-
// initialize SampleRate converter using AudioLayer's sampling rate
// (internal buffers initialized with maximal sampling rate and frame size)
_audioRtpRecord.setConverter(new SamplerateConverter (getAudioLayerSampleRate(), getAudioLayerFrameSize()));
@@ -400,7 +387,7 @@ void AudioRtpRecordHandler::putDtmfEvent(int digit)
int AudioRtpRecordHandler::processDataEncode(void)
{
- _debug("AudioProcessEncode");
+ _debug("Process data encode");
AudioCodec *audioCodec = getAudioCodec();
AudioLayer *audioLayer = Manager::instance().getAudioDriver();
@@ -413,21 +400,33 @@ int AudioRtpRecordHandler::processDataEncode(void)
assert(audioLayer);
int mainBufferSampleRate = audioLayer->getMainBuffer()->getInternalSamplingRate();
+
// compute codec framesize in ms
float fixedCodecFramesize = computeCodecFrameSize(getCodecFrameSize(), getCodecSampleRate());
+
// compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
int maxBytesToGet = computeNbByteAudioLayer(fixedCodecFramesize);
+
// available bytes inside ringbuffer
int availBytesFromMic = audioLayer->getMainBuffer()->availForGet(_ca->getCallId());
+ _debug(" avail byte from mic %d", availBytesFromMic);
+
+ if(!(availBytesFromMic > 320))
+ return 0;
+
// set available byte to maxByteToGet
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+
if(bytesAvail == 0){
memset(micDataEncoded, 0, sizeof (SFLDataFormat));
return getCodecFrameSize();
}
// Get bytes from micRingBuffer to data_from_mic
- int nbSample = audioLayer->getMainBuffer()->getData(micData, bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+ // int nbSample = audioLayer->getMainBuffer()->getData(micData, bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+ int nbSample = audioLayer->getMainBuffer()->getData(micData, 320, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+
+ // process mic fade in
if(!_audioRtpRecord.getMicFadeInComplete())
_audioRtpRecord.setMicFadeInComplete(fadeIn(micData, nbSample, _audioRtpRecord.getMicAmplFactor()));
@@ -443,13 +442,13 @@ int AudioRtpRecordHandler::processDataEncode(void)
int nbSampleUp = nbSample;
nbSample = _audioRtpRecord.getConverter()->downsampleData(micData, micDataConverted, _audioRtpRecord.getCodecSampleRate(), mainBufferSampleRate, nbSampleUp);
- if(Manager::instance().audioPreference.getNoiseReduce())
- _audioRtpRecord.getNoiseReductionProcess()->processAudio(micDataConverted, nbSample * sizeof (SFLDataFormat));
+ // if(Manager::instance().audioPreference.getNoiseReduce())
+ // _audioRtpRecord.getNoiseReductionProcess()->processAudio(micDataConverted, nbSample * sizeof (SFLDataFormat));
compSize = audioCodec->codecEncode(micDataEncoded, micDataConverted, nbSample * sizeof (SFLDataFormat));
}else{
- if(Manager::instance().audioPreference.getNoiseReduce())
- _audioRtpRecord.getNoiseReductionProcess()->processAudio(micData, nbSample * sizeof (SFLDataFormat));
+ // if(Manager::instance().audioPreference.getNoiseReduce())
+ // _audioRtpRecord.getNoiseReductionProcess()->processAudio(micData, nbSample * sizeof (SFLDataFormat));
// no resampling required
compSize = audioCodec->codecEncode(micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
@@ -460,8 +459,6 @@ int AudioRtpRecordHandler::processDataEncode(void)
void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned int size)
{
- _debug("AudioProcessDecode");
-
AudioCodec *audioCodec = getAudioCodec();
AudioLayer *audioLayer = Manager::instance().getAudioDriver();
@@ -471,7 +468,7 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned
SFLDataFormat *spkrDataDecoded = _audioRtpRecord.getSpkrDataConverted();
SFLDataFormat *spkrDataConverted = _audioRtpRecord.getSpkrDataDecoded();
- int mainBufferSampleRate = Manager::instance().getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+ int mainBufferSampleRate = audioLayer->getMainBuffer()->getInternalSamplingRate();
// Return the size of data in bytes
int expandedSize = audioCodec->codecDecode (spkrDataDecoded , spkrData , size);
@@ -491,17 +488,18 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned
nbSample = _audioRtpRecord.getConverter()->upsampleData (spkrDataDecoded, spkrDataConverted, _audioRtpRecord.getCodecSampleRate(), mainBufferSampleRate, nbSampleDown);
// put data in audio layer, size in byte
- Manager::instance().getAudioDriver()->getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
+ audioLayer->getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
} else {
// put data in audio layer, size in byte
- Manager::instance().getAudioDriver()->getMainBuffer()->putData (spkrDataDecoded, expandedSize, 100, _ca->getCallId());
+ audioLayer->getMainBuffer()->putData (spkrDataDecoded, expandedSize, 100, _ca->getCallId());
}
}
bool AudioRtpRecordHandler::fadeIn(SFLDataFormat *audio, int size, SFLDataFormat *factor)
{
+ /*
// apply amplitude factor;
while(size){
size--;
@@ -515,6 +513,8 @@ bool AudioRtpRecordHandler::fadeIn(SFLDataFormat *audio, int size, SFLDataFormat
return true;
return false;
+ */
+ return true;
}
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 21d4c6ccbb..d80dde79f0 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -41,6 +41,11 @@
namespace sfl
{
+inline uint32
+timeval2microtimeout(const timeval& t)
+{ return ((t.tv_sec * 1000000ul) + t.tv_usec); }
+
+
AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
// ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
AudioRtpRecordHandler(manager, sipcall),
@@ -216,24 +221,33 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
getEventQueue()->pop_front();
}
}
-/*
-bool onRTPPacketRecv (ost::IncomingRTPPkt&)
+
+bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
{
- _debug ("AudioRtpSession: onRTPPacketRecv");
+ //_debug ("AudioRtpSession: onRTPPacketRecv");
+
+ receiveSpeakerData ();
return true;
}
-*/
+
void AudioRtpSession::sendMicData()
{
- _debug("sendMicData");
+ // _debug("============== sendMicData ===============");
+
+ int compSize = processDataEncode();
+
+ // If no data, return
+ if(!compSize)
+ return;
// Increment timestamp for outgoing packet
_timestamp += _timestampIncrement;
- int compSize = processDataEncode();
+ // _debug(" compSize: %d", compSize);
+ // _debug(" timestamp: %d", _timestamp);
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
@@ -242,7 +256,6 @@ void AudioRtpSession::sendMicData()
void AudioRtpSession::receiveSpeakerData ()
{
- _debug("receiveSpkrData");
const ost::AppDataUnit* adu = NULL;
@@ -274,14 +287,31 @@ void AudioRtpSession::receiveSpeakerData ()
delete adu;
}
+void AudioRtpSession::notifyIncomingCall()
+{
+ // Notify (with a beep) an incoming call when there is already a call
+ if (Manager::instance().incomingCallWaiting() > 0) {
+ _countNotificationTime += _time->getSecond();
+ int countTimeModulo = _countNotificationTime % 5000;
+
+ // _debug("countNotificationTime: %d\n", countNotificationTime);
+ // _debug("countTimeModulo: %d\n", countTimeModulo);
+ if ( (countTimeModulo - _countNotificationTime) < 0) {
+ Manager::instance().notificationIncomingCall();
+ }
+ _countNotificationTime = countTimeModulo;
+ }
+}
+
int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
{
_debug ("AudioRtpSession: Starting main thread");
- initNoiseSuppress();
+ // initNoiseSuppress();
setSessionTimeouts();
setSessionMedia (audiocodec);
initBuffers();
+ enableStack();
int ret = start (_mainloopSemaphore);
return ret;
}
@@ -312,58 +342,54 @@ void AudioRtpSession::run ()
_debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
- // while (!static_cast<ost::Thread *>(this)->testCancel()) {
- while (!testCancel()) {
-
- _debug("audio");
-
- // 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 ();
-
-
-
- // Notify (with a beep) an incoming call when there is already a call
- if (Manager::instance().incomingCallWaiting() > 0) {
- _countNotificationTime += _time->getSecond();
- int countTimeModulo = _countNotificationTime % 5000;
-
- // _debug("countNotificationTime: %d\n", countNotificationTime);
- // _debug("countTimeModulo: %d\n", countTimeModulo);
- if ( (countTimeModulo - _countNotificationTime) < 0) {
- Manager::instance().notificationIncomingCall();
- }
-
- _countNotificationTime = countTimeModulo;
- }
-
- _manager->getAudioLayerMutex()->leave();
-
- // Let's wait for the next transmit cycle
- Thread::sleep (TimerPort::getTimer());
- TimerPort::incTimer (threadSleep);
- }
+ uint32 timeout = 0;
+ while ( isActive() ) {
+ if ( timeout < 1000 ){ // !(timeout/1000)
+ timeout = getSchedulingTimeout();
+ }
+
+ _manager->getAudioLayerMutex()->enter();
+
+ // Send session
+ if (getEventQueueSize() > 0) {
+ sendDtmfEvent (getEventQueue()->front());
+ } else {
+ sendMicData ();
+ }
+
+ // Recv session
+ // TODO should not be called here anymore
+// receiveSpeakerData ();
+
+ notifyIncomingCall();
+
+ _manager->getAudioLayerMutex()->leave();
+
+ setCancel(cancelDeferred);
+ controlReceptionService();
+ controlTransmissionService();
+ setCancel(cancelImmediate);
+ uint32 maxWait = timeval2microtimeout(getRTCPCheckInterval());
+ // make sure the scheduling timeout is
+ // <= the check interval for RTCP
+ // packets
+ timeout = (timeout > maxWait)? maxWait : timeout;
+ if ( timeout < 1000 ) { // !(timeout/1000)
+ setCancel(cancelDeferred);
+ dispatchDataPacket();
+ setCancel(cancelImmediate);
+ timerTick();
+ } else {
+ if ( isPendingData(timeout/1000) ) {
+ setCancel(cancelDeferred);
+ if (isActive()) { // take in only if active
+ takeInDataPacket();
+ }
+ setCancel(cancelImmediate);
+ }
+ timeout = 0;
+ }
+ }
_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 665dfe9cff..151a303178 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -73,6 +73,8 @@ class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public Aud
// Thread associated method
virtual void run ();
+ virtual bool onRTPPacketRecv (ost::IncomingRTPPkt&);
+
int startRtpThread (AudioCodec*);
/**
@@ -105,7 +107,6 @@ class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public Aud
*/
void setDestinationIpAddress (void);
-
/**
* Send encoded data to peer
*/
@@ -116,6 +117,11 @@ class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public Aud
*/
void receiveSpeakerData ();
+ /**
+ * Notificatify user with a beep for incoming calls
+ */
+ void notifyIncomingCall();
+
ost::Time * _time;
// This semaphore is not used
--
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