[#3443] Fix G722 codec timestamp incrementation

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

@@ -50,6 +50,9 @@
 #include <ccrtp/rtp.h>
 #include <cc++/numbers.h> // ost::Time
 
+// Frequency (in packet number)
+#define RTP_TIMESTAMP_RESET_FREQ 100
+
 namespace sfl {
 
     static const int schedulingTimeout = 100000;
@@ -206,6 +209,17 @@ namespace sfl {
               */
              int _timestamp;
 
+	     /**
+	      * Timestamp incrementation value based on codec period length (framesize) 
+	      * except for G722 which require a 8 kHz incrementation.
+	      */
+	     int _timestampIncrement;
+
+	     /**
+	      * Timestamp reset freqeuncy specified in number of packet sent
+	      */
+	     short _timestampCount;
+
              /**
               * Time counter used to trigger incoming call notification
               */
@@ -238,6 +252,8 @@ namespace sfl {
      _manager(manager),
      _converterSamplingRate(0),
      _timestamp(0),
+     _timestampIncrement(0),
+     _timestampCount(0),
      _countNotificationTime(0),
      _ca (sipcall)
     {
@@ -329,18 +345,26 @@ namespace sfl {
         _codecSampleRate = _audiocodec->getClockRate();
         _codecFrameSize = _audiocodec->getFrameSize();
 
+	// G722 requires timestamp to be incremented at 8 kHz
+	if (_audiocodec->getPayload() == 9)
+	    _timestampIncrement = 160;
+	else
+	    _timestampIncrement = _codecFrameSize;
+	  
+
         _debug("RTP: Codec sampling rate: %d", _codecSampleRate);
         _debug("RTP: Codec frame size: %d", _codecFrameSize);
+	_debug("RTP: RTP timestamp increment: %d", _timestampIncrement);
 
-        //TODO: figure out why this is necessary.
+        // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
         if (_audiocodec->getPayload() == 9) {
-            _debug ("RTP: Setting payload format to G722");
+            _debug ("RTP: Setting G722 payload format");
             static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
         } else if (_audiocodec->hasDynamicPayload()) {
-            _debug ("RTP: Setting a dynamic payload format");
+            _debug ("RTP: Setting dynamic payload format");
             static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
         } else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
-            _debug ("RTP: Setting a static payload format");
+            _debug ("RTP: Setting static payload format");
             static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
         }
     }
@@ -567,7 +591,7 @@ namespace sfl {
         //   4. send it
 
         // Increment timestamp for outgoing packet
-        _timestamp += _codecFrameSize;
+        _timestamp += _timestampIncrement;
 
         if (!_audiolayer) {
             _debug ("No audiolayer available for MIC");
@@ -668,14 +692,19 @@ namespace sfl {
 
         while (!testCancel()) {
 
-        	// ost::MutexLock lock(*(_manager->getAudioLayerMutex()));
+	    // Reset timestamp to make sure the timing information are up to date
+	    if(_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
+	        _timestamp = static_cast<D*>(this)->getCurrentTimestamp();
+		_timestampCount = 0;
+	    }
+	    _timestampCount++;
+
 	  
 	    _manager->getAudioLayerMutex()->enter();
 
 	    // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
 	    _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
 
-
 	    sessionWaiting = static_cast<D*>(this)->isWaiting();
 
             // Send session