[#2209] Add resampling changes for IAX

sflphone-common/src/audio/mainbuffer.cpp

@@ -43,9 +43,11 @@ void MainBuffer::setInternalSamplingRate(int sr)
 	// flushAllBuffers();
 	_internalSamplingRate = sr;
 
+	flushAllBuffers();
+
     }
 
-    flushAllBuffers();
+    // flushAllBuffers();
 }
 
 CallIDSet* MainBuffer::getCallIDSet(CallID call_id)

sflphone-common/src/iax/iaxvoiplink.cpp

@@ -127,6 +127,10 @@ IAXVoIPLink::init()
             _evThread->start();
 
             audiolayer = Manager::instance().getAudioDriver();
+
+	    // may be different than one already setted
+            converterSamplingRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
+	    
             break;
         }
 
@@ -313,12 +317,12 @@ IAXVoIPLink::sendAudioFromMic (void)
         return;
     }
 
-    
+    audiolayer->getMainBuffer()->setInternalSamplingRate(ac->getClockRate());
 
     // Send sound here
     if (audiolayer) {
 
-	
+	int _mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
 
         // we have to get 20ms of data from the mic *20/1000 = /50
         // rate/50 shall be lower than IAX__20S_48KHZ_MAX
@@ -347,11 +351,19 @@ IAXVoIPLink::sendAudioFromMic (void)
 
         // _debug("IAXVoIPLink::sendAudioFromMic : %i \n",nbSampleForRec_);
 
+
+	if (ac->getClockRate() && (ac->getClockRate() != _mainBufferSampleRate)) {
+
 	    // resample
-        nbSample_ = converter->downsampleData (micData , micDataConverted , (int) ac ->getClockRate() , (int) audiolayer->getSampleRate() , nbSample_);
+	    nbSample_ = converter->downsampleData (micData , micDataConverted , (int) ac->getClockRate(), _mainBufferSampleRate, nbSample_);
 	
 	    // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
 	    compSize = ac->codecEncode (micDataEncoded, micDataConverted , nbSample_*sizeof (int16));
+	} else {
+
+	    // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
+	    compSize = ac->codecEncode (micDataEncoded, micData, nbSample_*sizeof (int16));
+	}
 
 	// Send it out!
 	_mutexIAX.enterMutex();
@@ -872,7 +884,9 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
     int expandedSize, nbSample_;
     AudioCodec *ac;
 
+    ac = call->getCodecMap().getCodec (call -> getAudioCodec());
 
+    audiolayer->getMainBuffer()->setInternalSamplingRate(ac->getClockRate());
     
     // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
     // were triggered
@@ -885,6 +899,8 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
 
     if (audiolayer) {
 
+	int _mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
+
         // On-the-fly codec changing (normally, when we receive a full packet)
         // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
         // - subclass holds the voiceformat property.
@@ -893,7 +909,7 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
         }
 
         //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
-        ac = call->getCodecMap().getCodec (call -> getAudioCodec());
+        // ac = call->getCodecMap().getCodec (call -> getAudioCodec());
 
         data = (unsigned char*) event->data;
 
@@ -918,12 +934,22 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
 
         nbSample_ = nbInt16;
 
+	// test if resampling is required
+        if (ac->getClockRate() && (ac->getClockRate() != _mainBufferSampleRate)) {
+
 	    // resample
-        nbInt16 = converter->upsampleData (spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample_);
+	    nbInt16 = converter->upsampleData (spkrDataDecoded, spkrDataConverted, ac->getClockRate(), _mainBufferSampleRate, nbSample_);
 
 	    /* Write the data to the mic ring buffer */
 	    audiolayer->getMainBuffer()->putData (spkrDataConverted, nbInt16 * sizeof (SFLDataFormat), 100, call->getCallId());
 
+	} else {
+
+	    /* Write the data to the mic ring buffer */
+	    audiolayer->getMainBuffer()->putData (spkrDataDecoded, nbInt16 * sizeof (SFLDataFormat), 100, call->getCallId());
+	    
+	}
+
     } else {
         _debug ("IAX: incoming audio, but no sound card open");
     }

sflphone-common/src/iax/iaxvoiplink.h

@@ -288,6 +288,8 @@ class IAXVoIPLink : public VoIPLink
     /** Sample rate converter object */
     SamplerateConverter* converter;
 
+    int converterSamplingRate;
+    
     /** number of sample */
     int nbSample_;