[#4367] Remove debug in rtp session

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

@@ -387,7 +387,6 @@ void AudioRtpRecordHandler::putDtmfEvent(int digit)
 
 int AudioRtpRecordHandler::processDataEncode(void)
 {
-	_debug("Process data encode");
 
 	AudioCodec *audioCodec = getAudioCodec();
 	AudioLayer *audioLayer = Manager::instance().getAudioDriver();
@@ -407,11 +406,8 @@ int AudioRtpRecordHandler::processDataEncode(void)
     // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
     int bytesToGet = computeNbByteAudioLayer(fixedCodecFramesize);
 
-    _debug("    byte to get %d", bytesToGet);
-
     // available bytes inside ringbuffer
     int availBytesFromMic = audioLayer->getMainBuffer()->availForGet(_ca->getCallId());
-    _debug("    avail byte from mic %d", availBytesFromMic);
 
     if(availBytesFromMic < bytesToGet)
     	return 0;

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

@@ -235,8 +235,6 @@ bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
 
 void AudioRtpSession::sendMicData()
 {
-    _debug("============== sendMicData ===============");
-
     int compSize = processDataEncode();
 
     // If no data, return
@@ -254,9 +252,6 @@ void AudioRtpSession::sendMicData()
     // Increment timestamp for outgoing packet
     _timestamp += _timestampIncrement;
 
-    _debug("    compSize: %d", compSize);
-    _debug("    timestamp: %d", _timestamp);
-
     // putData put the data on RTP queue, sendImmediate bypass this queue
     putData (_timestamp, getMicDataEncoded(), compSize);
 }
@@ -302,8 +297,6 @@ void AudioRtpSession::notifyIncomingCall()
 		_countNotificationTime += _time->getSecond();
 		int countTimeModulo = _countNotificationTime % 5000;
 
-		// _debug("countNotificationTime: %d\n", countNotificationTime);
-		// _debug("countTimeModulo: %d\n", countTimeModulo);
 		if ( (countTimeModulo - _countNotificationTime) < 0) {
 			Manager::instance().notificationIncomingCall();
 		}
@@ -346,7 +339,6 @@ void AudioRtpSession::run ()
 
     // Start audio stream (if not started) AND flush all buffers (main and urgent)
     _manager->getAudioDriver()->startStream();
-    // startRunning();
 
     _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());