[#4847] Apply RTP modification to ZRTP session

sflphone-common/src/audio/audiortp/AudioRtpSession.cpp

@@ -255,9 +255,7 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
 
 bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
 {
-    // audioCodecMutex.enter();
     receiveSpeakerData ();
-    // audioCodecMutex.leave();
 
     return true;
 }
@@ -355,9 +353,6 @@ void AudioRtpSession::run ()
     // Set recording sampling rate
     _ca->setRecordingSmplRate (getCodecSampleRate());
 
-    // Start audio stream (if not started) AND flush all buffers (main and urgent)
-    // _manager->getAudioDriver()->startStream();
-
     _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
 
     uint32 timeout = 0;
@@ -394,7 +389,6 @@ void AudioRtpSession::run ()
             setCancel (cancelImmediate);
             timerTick();
         } else {
-
             if (isPendingData (timeout/1000)) {
                 setCancel (cancelDeferred);
 

sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp

 /*
  *  Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
+ *  Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
  *  Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
  *
  *  This program is free software; you can redistribute it and/or modify
@@ -89,7 +90,7 @@ AudioZrtpSession::~AudioZrtpSession()
         throw;
     }
 
-    _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
+    Manager::instance().getMainBuffer()->unBindAll (_ca->getCallId());
 
     if (_time)
         delete _time;
@@ -177,14 +178,11 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
     _debug ("AudioZrtpSession: RTP timestamp increment: %d", _timestampIncrement);
 
     // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
-    if (payloadType == g722PayloadType) {
-        _debug ("AudioZrtpSession: Setting G722 payload format");
-        setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
-    } else if (dynamic) {
-        _debug ("AudioZrtpSession: Setting dynamic payload format");
+    if (dynamic) {
+        _debug ("AudioRtpSession: Setting dynamic payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
-    } else if (dynamic && payloadType != g722PayloadType) {
-        _debug ("AudioZrtpSession: Setting static payload format");
+    } else {
+        _debug ("AudioRtpSession: Setting static payload format");
         setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
     }
 
@@ -214,13 +212,10 @@ void AudioZrtpSession::updateSessionMedia (AudioCodec *audioCodec)
     _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
 
     // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
-    if (payloadType == g722PayloadType) {
-        _debug ("AudioRtpSession: Setting G722 payload format");
-        setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
-    } else if (dynamic) {
+    if (dynamic) {
         _debug ("AudioRtpSession: Setting dynamic payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
-    } else if (dynamic && payloadType != g722PayloadType) {
+    } else {
         _debug ("AudioRtpSession: Setting static payload format");
         setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
     }
@@ -288,7 +283,8 @@ void AudioZrtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
     if (dtmf->newevent)
         setMark (true);
 
-    putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+    // putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+    sendImmediate (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
 
     // This is no more a new event
     if (dtmf->newevent) {
@@ -330,17 +326,12 @@ void AudioZrtpSession::sendMicData()
     if (!compSize)
         return;
 
-    // Reset timestamp to make sure the timing information are up to date
-    if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
-        _timestamp = getCurrentTimestamp();
-        _timestampCount = 0;
-    }
-
     // Increment timestamp for outgoing packet
     _timestamp += _timestampIncrement;
 
     // putData put the data on RTP queue, sendImmediate bypass this queue
     putData (_timestamp, getMicDataEncoded(), compSize);
+    sendImmediate (_timestamp, getMicDataEncoded(), compSize);
 }
 
 
@@ -388,43 +379,33 @@ void AudioZrtpSession::notifyIncomingCall()
 
 int AudioZrtpSession::startRtpThread (AudioCodec* audiocodec)
 {
+    if (_isStarted)
+        return 0;
+
     _debug ("AudioZrtpSession: Starting main thread");
+    _isStarted = true;
     setSessionTimeouts();
     setSessionMedia (audiocodec);
     initBuffers();
     initNoiseSuppress();
     enableStack();
-
     int ret = start (_mainloopSemaphore);
-
     return ret;
 }
 
-void AudioZrtpSession::run ()
+void AudioZrtpSession::stopRtpThread ()
 {
+    _debug ("AudioZrtpSession: Stoping main thread");
 
-    // Timestamp must be initialized randomly
-    _timestamp = getCurrentTimestamp();
-
-    /*
-    int threadSleep = 0;
-
-    if (getCodecSampleRate() != 0) {
-        threadSleep = (getCodecFrameSize() * 1000) / getCodecSampleRate();
-    } else {
-    	// TODO should not be dependent of audio layer frame size
-        threadSleep = getAudioLayerFrameSize();
-    }
+    disableStack();
+}
 
-    TimerPort::setTimer (threadSleep);
-    */
+void AudioZrtpSession::run ()
+{
 
     // Set recording sampling rate
     _ca->setRecordingSmplRate (getCodecSampleRate());
 
-    // Start audio stream (if not started) AND flush all buffers (main and urgent)
-    _manager->getAudioDriver()->startStream();
-
     _debug ("AudioZrtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
 
     uint32 timeout = 0;
@@ -435,8 +416,6 @@ void AudioZrtpSession::run ()
             timeout = getSchedulingTimeout();
         }
 
-        // _manager->getAudioLayerMutex()->enter();
-
         // Send session
         if (getEventQueueSize() > 0) {
             sendDtmfEvent (getEventQueue()->front());
@@ -447,8 +426,6 @@ void AudioZrtpSession::run ()
         // This also should be moved
         notifyIncomingCall();
 
-        // _manager->getAudioLayerMutex()->leave();
-
         setCancel (cancelDeferred);
         controlReceptionService();
         controlTransmissionService();
@@ -461,7 +438,7 @@ void AudioZrtpSession::run ()
 
         if (timeout < 1000) {   // !(timeout/1000)
             setCancel (cancelDeferred);
-            dispatchDataPacket();
+            // dispatchDataPacket();
             setCancel (cancelImmediate);
             timerTick();
         } else {

sflphone-common/src/audio/audiortp/AudioZrtpSession.h

 /*
  *  Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
+ *  Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
  *  Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
  *
  *  This program is free software; you can redistribute it and/or modify
@@ -65,7 +66,9 @@ class AudioZrtpSession : protected ost::Thread, public AudioRtpRecordHandler, pu
 
         virtual bool onRTPPacketRecv (ost::IncomingRTPPkt&);
 
-        int startRtpThread (AudioCodec*);
+        int startRtpThread (AudioCodec *);
+
+        void stopRtpThread (void);
 
         /**
          * Used mostly when receiving a reinvite
@@ -168,6 +171,8 @@ class AudioZrtpSession : protected ost::Thread, public AudioRtpRecordHandler, pu
         int _countNotificationTime;
 
         SIPCall * _ca;
+
+        bool _isStarted;
 };
 
 }