[#2209] Reinit converterSamplingRate in RTP sessions

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

@@ -81,8 +81,8 @@ namespace sfl {
             inline float computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate) {
                 return ( (float) codecSamplePerFrame * 1000.0) / (float) codecClockRate;
             }          
-            inline int computeNbByteAudioLayer (float codecFrameSize) {
-                return (int) ( (float) converterSamplingRate * codecFrameSize * (float) sizeof (SFLDataFormat) / 1000.0);
+            int computeNbByteAudioLayer (float codecFrameSize) {
+                return (int) ( ((float) converterSamplingRate * codecFrameSize * sizeof(SFLDataFormat))/ 1000.0);
             }
           
             void sendMicData(int timestamp);
@@ -225,9 +225,9 @@ namespace sfl {
 
 	// initialize SampleRate converter using AudioLayer's sampling rate
 	// (internal buffers initialized with maximal sampling rate and frame size)
-        _converter = new SamplerateConverter (_layerSampleRate, _layerFrameSize);
+        _converter = new SamplerateConverter(_layerSampleRate, _layerFrameSize);
 
-        int nbSamplesMax = (int) (_codecSampleRate * _layerFrameSize /1000)*2;
+        int nbSamplesMax = (int)(_codecSampleRate * _layerFrameSize /1000)*2;
         _micData = new SFLDataFormat[nbSamplesMax];
         _micDataConverted = new SFLDataFormat[nbSamplesMax];
         _micDataEncoded = new unsigned char[nbSamplesMax];
@@ -309,19 +309,31 @@ namespace sfl {
     {
         assert(_audiocodec);
         assert(_audiolayer);
-	
+
+	_debug("AudioRtpSession::processDataEncode %s\n", _ca->getCallId().c_str());
 
 	int _mainBufferSampleRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
-        
+
+	_debug("    _mainBufferSampleRate %i\n", _mainBufferSampleRate);
+
+	_debug("    converterSamplingRate %i\n", converterSamplingRate);
+
+	_debug("    _audiocodec->getFrameSize() %i, _audiocodec->getClockRate() %i\n", _audiocodec->getFrameSize(), _audiocodec->getClockRate());
+
         // compute codec framesize in ms
         float fixed_codec_framesize = computeCodecFrameSize (_audiocodec->getFrameSize(), _audiocodec->getClockRate());
+	_debug("    fixed_codec_framesize %f\n", fixed_codec_framesize);
 
         // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
         int maxBytesToGet = computeNbByteAudioLayer (fixed_codec_framesize);
 
+	_debug("    maxBytesToGet %i\n", maxBytesToGet);
+
         // available bytes inside ringbuffer
         int availBytesFromMic = _audiolayer->getMainBuffer()->availForGet(_ca->getCallId());
 
+	_debug("    availBytesFromMic %i\n", availBytesFromMic);
+
         // set available byte to maxByteToGet
         int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
 
@@ -331,6 +343,8 @@ namespace sfl {
         // Get bytes from micRingBuffer to data_from_mic
         int nbSample = _audiolayer->getMainBuffer()->getData(_micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
 
+	_debug("    nbSample %i\n", nbSample);
+
         // nb bytes to be sent over RTP
         int compSize = 0;
 
@@ -342,6 +356,8 @@ namespace sfl {
 
             compSize = _audiocodec->codecEncode (_micDataEncoded, _micDataConverted, nbSample*sizeof (int16));
 
+	    _debug("    nbSample(in resampling block) %i\n", nbSample); 
+
         } else {
             // no resampling required
             compSize = _audiocodec->codecEncode (_micDataEncoded, _micData, nbSample*sizeof (int16));
@@ -354,8 +370,11 @@ namespace sfl {
     void AudioRtpSession<D>::processDataDecode(unsigned char * spkrData, unsigned int size, int& countTime) 
     {
 
+	_debug("AudioRtpSession::processDataDecode %s\n", _ca->getCallId().c_str());
+
         if (_audiocodec != NULL) {
 
+
 	    int _mainBufferSampleRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
 
             // Return the size of data in bytes
@@ -364,6 +383,10 @@ namespace sfl {
             // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
             int nbSample = expandedSize / sizeof (SFLDataFormat);
 
+	    _debug("    nbSample %i\n", nbSample);
+
+	    _debug("    _mainBufferSampleRate %i\n", _mainBufferSampleRate);
+
             // test if resampling is required
             if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
 
@@ -375,6 +398,8 @@ namespace sfl {
                 // Store the number of samples for recording
                 _nSamplesSpkr = nbSample;
 
+		_debug("    nbSample (in resampling block) %i\n", nbSample);
+
                 // put data in audio layer, size in byte
 		_audiolayer->getMainBuffer()->putData (_spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, _ca->getCallId());
 
@@ -383,6 +408,7 @@ namespace sfl {
                 // Store the number of samples for recording
                 _nSamplesSpkr = nbSample;
 
+
                 // put data in audio layer, size in byte
                 _audiolayer->getMainBuffer()->putData (_spkrDataDecoded, expandedSize, 100, _ca->getCallId());
             }
@@ -503,6 +529,8 @@ namespace sfl {
 
         while (!testCancel()) {
 
+	    converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
+
             // Send session
             sessionWaiting = static_cast<D*>(this)->isWaiting();