[#790] Reduce number of start/stop streams call on pulse audio

sflphone-client-gnome/src/contacts/calltree.c

@@ -412,8 +412,6 @@ calltree_update_call (calltab_t* tab, callable_obj_t * c)
 void calltree_add_history_entry (callable_obj_t * c)
 {
 
-    printf("calltree_add_history_entry\n");
-
     if (dbus_get_history_enabled () == 0)
         return;
 
@@ -473,8 +471,6 @@ void calltree_add_history_entry (callable_obj_t * c)
 void calltree_add_call (calltab_t* tab, callable_obj_t * c)
 {
 
-    printf("calltree_add_call\n");
-
     if (tab == history)
     {
         calltree_add_history_entry (c);

sflphone-common/src/audio/audiortp.cpp

@@ -98,7 +98,7 @@ AudioRtp::closeRtpSession () {
         throw;
     }
     AudioLayer* audiolayer = Manager::instance().getAudioDriver();
-    audiolayer->stopStream();
+    // audiolayer->stopStream();
 
     return true;
 }
@@ -398,52 +398,6 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
 
     int compSize = processDataEncode(audiolayer);
 
-    /*
-    // compute codec framesize in ms
-    float fixed_codec_framesize = computeCodecFrameSize(_audiocodec->getFrameSize(), _audiocodec->getClockRate());
-
-    // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
-    int maxBytesToGet = computeNbByteAudioLayer(fixed_codec_framesize);
-    
-    // available bytes inside ringbuffer
-    int availBytesFromMic = audiolayer->canGetMic();
-
-    // set available byte to maxByteToGet
-    int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
-    if (bytesAvail == 0)
-      return;
-
-    // Get bytes from micRingBuffer to data_from_mic
-    int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
-
-    // nb bytes to be sent over RTP
-    int compSize = 0;
-
-    // test if resampling is required
-    if (_audiocodec->getClockRate() != _layerSampleRate) {
-
-        int nb_sample_up = nbSample;
-        // _debug("_nbSample audiolayer->getMic(): %i \n", nbSample);
-    
-        // Store the length of the mic buffer in samples for recording
-        _nSamplesMic = nbSample;
-
-
-        // int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000; 
-	 nbSample = reSampleData(micData , micDataConverted, _audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
-
-        compSize = _audiocodec->codecEncode( micDataEncoded, micDataConverted, nbSample*sizeof(int16));
-
-    } else {
-        // no resampling required
-
-        // int nbSamplesMax = _codecFrameSize;
-        compSize = _audiocodec->codecEncode( micDataEncoded, micData, nbSample*sizeof(int16));
-
-    }
-    */
-
     // putData put the data on RTP queue, sendImmediate bypass this queue
     if (!_sym) {
         // _sessionSend->putData(timestamp, micDataEncoded, compSize);
@@ -483,53 +437,9 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
     unsigned int size = adu->getSize(); // size in char
 
     processDataDecode(audiolayer, spkrData, size, countTime);
-    /*
-    if (_audiocodec != NULL) {
-
-        // Return the size of data in bytes 
-        int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
-
-        // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
-        int nbSample = expandedSize / sizeof(SFLDataFormat);
-
-        // test if resampling is required 
-        if (_audiocodec->getClockRate() != _layerSampleRate) {
-
-            // Do sample rate conversion
-            int nb_sample_down = nbSample;
-            nbSample = reSampleData(spkrDataDecoded , spkrDataConverted, _codecSampleRate , nb_sample_down, UP_SAMPLING);
-
-            // Store the number of samples for recording
-            _nSamplesSpkr = nbSample;
-
-	    // put data in audio layer, size in byte
-            audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
     
-        } else {
-
-            // Stor the number of samples for recording
-            _nSamplesSpkr = nbSample;
-
-	    // put data in audio layer, size in byte
-            audiolayer->putMain (spkrDataDecoded, nbSample * sizeof(SFLDataFormat));
-        }
-
-        // Notify (with a beep) an incoming call when there is already a call 
-        countTime += time->getSecond();
-        if (Manager::instance().incomingCallWaiting() > 0) {
-            countTime = countTime % 500; // more often...
-            if (countTime == 0) {
-                Manager::instance().notificationIncomingCall();
-            }
 }
 
-    } else {
-        countTime += time->getSecond();
-    }
-    delete adu; adu = NULL;
-    */
-
-}
 
     int 
     AudioRtpRTX::reSampleData(SFLDataFormat *input, SFLDataFormat *output, int sampleRate_codec, int nbSamples, int status)
@@ -634,7 +544,7 @@ AudioRtpRTX::run () {
       
     }
     
-    audiolayer->stopStream();
+    // audiolayer->stopStream();
     _debug("- ARTP Action: Stop call %s\n",_ca->getCallId().c_str());
   //} catch(std::exception &e) {
     //_start.post();

sflphone-common/src/audio/audiostream.cpp

@@ -49,10 +49,6 @@ AudioStream::connectStream()
 
   if(!_audiostream)
     _audiostream = createStream( _context );
-  else {
-    disconnectStream();
-    _audiostream = createStream( _context );
-  }
 
   return true;
 }

sflphone-common/src/audio/pulselayer.cpp

@@ -166,12 +166,15 @@ bool PulseLayer::createStreams( pa_context* c )
 
     pa_threaded_mainloop_signal(m , 0);
 
+    isCorked = true;
+
     return true;
 }
 
 
 bool PulseLayer::openDevice(int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED) 
 {
+
     _debug("PulseLayer::openDevice \n");
     _sampleRate = sampleRate;
     _frameSize = frameSize;	
@@ -225,6 +228,7 @@ int PulseLayer::getMic(void *buffer, int toCopy)
 void PulseLayer::startStream (void) 
 {
     // flushMic();
+    if (isCorked) {
 
         _urgentRingBuffer.flush();
         _micRingBuffer.flush();
@@ -238,11 +242,17 @@ void PulseLayer::startStream (void)
   
         pa_threaded_mainloop_unlock(m);
 
+	isCorked = false;
+    }
+
 }
 
     void 
 PulseLayer::stopStream (void) 
 {
+
+    if (!isCorked) {
+
         _debug("PulseLayer::Stop stream\n");
         pa_stream_flush( playback->pulseStream(), NULL, NULL );
 	pa_stream_flush( record->pulseStream(), NULL, NULL );
@@ -254,6 +264,10 @@ PulseLayer::stopStream (void)
 	pa_stream_cork( playback->pulseStream(), 1, NULL, NULL);
 	pa_stream_cork( record->pulseStream(), 1, NULL, NULL);
 
+	isCorked = true;
+
+    }
+	
 }
 
 

sflphone-common/src/audio/pulselayer.h

@@ -189,6 +189,8 @@ class PulseLayer : public AudioLayer {
      */
     AudioStream* record;
 
+    bool isCorked;
+
     int spkrVolume;
     int micVolume;
 

sflphone-common/src/managerimpl.cpp

@@ -326,6 +326,13 @@ ManagerImpl::hangupCall(const CallID& id)
 
     _debug("Stop audio stream\n");
     audiolayer = getAudioDriver();
+
+    int nbCalls = getCallList().size();
+
+    _debug("nbCalls %i \n", nbCalls);
+
+    // stop stream 
+    if (!(nbCalls > 1))
         audiolayer->stopStream();
 
     /* Direct IP to IP call */
@@ -958,6 +965,7 @@ void ManagerImpl::stopTone (bool stopAudio=true)
         if (audiolayer) audiolayer->stopStream();
     }
     
+
     _toneMutex.enterMutex();
     if (_telephoneTone != 0) {
         _telephoneTone->setCurrentTone(Tone::TONE_NULL);