Clean up, indentation, try to handle latency problems in iax/pulseaudio

src/audio/audiortp.cpp

@@ -452,19 +452,8 @@ AudioRtpRTX::run () {
         TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
     }
 
-    // _debug("stop stream for audiortp loop\n");
     audiolayer->stopStream();
     _debug("- ARTP Action: Stop\n");
-  //} catch(std::exception &e) {
-    //_start.post();
-    //_debug("! ARTP: Stop %s\n", e.what());
-    //throw;
-  //} catch(...) {
-    //_start.post();
-    //_debugException("* ARTP Action: Stop");
-    //throw;
-  //}
-    
 }
 
 

src/audio/pulselayer.cpp

@@ -198,7 +198,7 @@ PulseLayer::getMic(void *buffer, int toCopy)
     void 
 PulseLayer::startStream (void) 
 {
-  _micRingBuffer.flush();
+    flushMic();
     _debug("Start stream\n");
     pa_threaded_mainloop_lock(m);
     pa_stream_cork( record->pulseStream(), NULL, NULL, NULL);

src/iaxvoiplink.cpp

@@ -429,7 +429,6 @@ IAXVoIPLink::answer(const CallID& id)
     call->setConnectionState(Call::Connected);
     // Start audio
     audiolayer->startStream();
-    audiolayer->flushMic();
 
     return true;
 }
@@ -701,7 +700,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
                 }
 
                 Manager::instance().peerAnsweredCall(id);
-                audiolayer->flushMic();
                 // start audio here?
             } else {
                 // deja connecté ?
@@ -805,7 +803,7 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
         // resample
         nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample_);
 
-        //audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+        /* Write the data to the mic ring buffer */
         audiolayer->putMain (spkrDataConverted , nbInt16 * sizeof(SFLDataFormat));
 
     } else {