[#1966] Add echocancellation when resampling not required

sflphone-common/src/audio/echocancel.cpp

@@ -66,7 +66,7 @@ EchoCancel::EchoCancel(int smplRate, int frameLength) : _samplingRate(smplRate),
   echoFile = new ofstream("echoData", ofstream::binary);
   spkrFile = new ofstream("spkrData", ofstream::binary);
   
-  micLevelData = new ofstream("micLevelData", ofstream::binary);
+  micLevelData = new ofstream("micLevelData", ofstream:binary);
   spkrLevelData = new ofstream("spkrLevelData", ofstream::binary);
   */
 
@@ -168,7 +168,7 @@ void EchoCancel::reset()
   memset(_delayLineAmplify, 0, MAX_DELAY_LINE_AMPL*sizeof(float));
 
   _amplDelayIndexIn = 0;
-  _amplDelayIndexOut = 0;
+  _amplDelayIndexOut = 100;
 
   _adaptDone = false;
   _adaptStarted = false;
@@ -204,6 +204,8 @@ void EchoCancel::reset()
   speex_preprocess_ctl(_noiseState, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL, &f);
 
   _spkrStoped = true;
+
+  _processedByte = 0;
 }
 
 void EchoCancel::putData(SFLDataFormat *inputData, int nbBytes) 
@@ -270,17 +272,15 @@ int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
   int spkrAvail = _spkrData->AvailForGet();
   int micAvail = _micData->AvailForGet();
 
-  _debug("EchoCancel: speaker avail %d, mic avail %d, processed: %d", spkrAvail/320, micAvail/320, _processedByte);
+  _debug("EchoCancel: speaker avail %d, mic avail %d, processed: %d", spkrAvail/320, micAvail/320, _processedByte/320);
 
   // Init number of frame processed
   int nbFrame = 0;
 
 
   // Get data from mic and speaker internal buffer
-  // while((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
-  if ((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
-
-    _debug("----------- ProcessData ----------");
+  while((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
+  // if ((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
 
     // get synchronized data
     _spkrData->Get(_tmpSpkr, byteSize);
@@ -318,8 +318,6 @@ int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
     ++nbFrame;
   }
 
-  
-
   // _event.reset();
 
   return nbFrame * _smplPerFrame;
@@ -345,7 +343,7 @@ void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDa
   // int tempspkrlevel[_nbSegmentPerFrame];
 
   for(int k = 0; k < _nbSegmentPerFrame; k++) {
-    /*
+
     updateEchoCancel(micData+(k*_smplPerSeg), spkrData+(k*_smplPerSeg));
 
     _spkrLevel = getMaxAmplitude(_avgSpkrLevelHist, _spkrHistoryLength);
@@ -354,9 +352,9 @@ void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDa
     // _debug("_spkrLevel: (max): %d", _spkrLevel);
     // _debug("_micLevel: (min): %d", _micLevel);
 
-    tempspkrlevel[k] = _spkrLevel;
-    tempmiclevel[k] = _micLevel;
-    */
+    // tempspkrlevel[k] = _spkrLevel;
+    // tempmiclevel[k] = _micLevel;
+
     /*
     if(_micLevel >= MIN_SIG_LEVEL) {
       if(_spkrLevel < MIN_SIG_LEVEL) {
@@ -384,10 +382,10 @@ void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDa
       }
     }
     */
-    /*
+
+
     if(_spkrLevel >= MIN_SIG_LEVEL) {
       _amplFactor = 0.0;
-      // _debug("------------------------------------------ echocancel");
     }
     else {
       increaseFactor(0.02);
@@ -398,8 +396,8 @@ void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDa
     _lastAmplFactor = _amplFactor;
     amplifySignal(micData+(k*_smplPerSeg), outputData+(k*_smplPerSeg), amplify);
 
-    */
-    amplifySignal(micData+(k*_smplPerSeg), outputData+(k*_smplPerSeg), 1.0);
+
+    // amplifySignal(micData+(k*_smplPerSeg), outputData+(k*_smplPerSeg), 1.0);
     
   }
 
@@ -503,13 +501,12 @@ int EchoCancel::getMaxAmplitude(int *data, int size) {
 
 void EchoCancel::amplifySignal(SFLDataFormat *micData, SFLDataFormat *outputData, float amplify) {
 
-  for(int i = 0; i < _smplPerSeg; i++)
-      outputData[i] = micData[i];
+  // for(int i = 0; i < _smplPerSeg; i++)
+  //  outputData[i] = micData[i];
 
   // Use delayed amplification factor due to sound card latency 
   // do not increment amplitude array if adaptation is not done 
   // if (_adaptDone) {
-  /*
   if (true) {
     for(int i = 0; i < _smplPerSeg; i++) {
         outputData[i] = (SFLDataFormat)(((float)micData[i])*_delayLineAmplify[_amplDelayIndexOut]);
@@ -529,7 +526,7 @@ void EchoCancel::amplifySignal(SFLDataFormat *micData, SFLDataFormat *outputData
 
   if(_amplDelayIndexIn >= MAX_DELAY_LINE_AMPL)
     _amplDelayIndexIn = 0;
-  */
+
 }
 
 

sflphone-common/src/audio/pulseaudio/audiostream.cpp

@@ -214,7 +214,7 @@ AudioStream::createStream (pa_context* c, std::string *deviceName)
     if (_streamType == PLAYBACK_STREAM) {
 
         attributes->maxlength = (uint32_t) -1;
-        attributes->tlength = pa_usec_to_bytes (20 * PA_USEC_PER_MSEC, &_sample_spec);
+        attributes->tlength = pa_usec_to_bytes (200 * PA_USEC_PER_MSEC, &_sample_spec);
         attributes->prebuf = 0;
         attributes->minreq = (uint32_t) -1;
 	
@@ -231,9 +231,9 @@ AudioStream::createStream (pa_context* c, std::string *deviceName)
     } else if (_streamType == CAPTURE_STREAM) {
 
         attributes->maxlength = (uint32_t) -1;
-	attributes->tlength = pa_usec_to_bytes (20 * PA_USEC_PER_MSEC, &_sample_spec);
+	attributes->tlength = pa_usec_to_bytes (200 * PA_USEC_PER_MSEC, &_sample_spec);
 	attributes->prebuf = 0;
-        attributes->fragsize = pa_usec_to_bytes (100 * PA_USEC_PER_MSEC, &_sample_spec);
+        attributes->fragsize = pa_usec_to_bytes (200 * PA_USEC_PER_MSEC, &_sample_spec);
 
 	pa_threaded_mainloop_lock(_mainloop);
 

sflphone-common/src/audio/pulseaudio/pulselayer.cpp

@@ -857,9 +857,6 @@ void PulseLayer::writeToSpeaker (void)
 		// to avoid such problem
 		AudioLayer::_echoCancel->setSamplingRate(_mainBufferSampleRate);
 
-		// Copy far-end signal in echo canceller to adapt filter coefficient
-		AudioLayer::_echoCanceller->putData(out, byteToGet);
-
                 // test if resampling is required
                 if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
 
@@ -883,6 +880,9 @@ void PulseLayer::writeToSpeaker (void)
 
                 }
 
+		// Copy far-end signal in echo canceller to adapt filter coefficient
+		AudioLayer::_echoCanceller->putData(out, byteToGet);
+
                 pa_xfree (out);
 
             } else {
@@ -939,7 +939,7 @@ void PulseLayer::readFromMic (void)
 
 	    // captureFile->write ((const char *)data, nbSample*sizeof(SFLDataFormat));
 
-            nbSample = _converter->downsampleData ( (SFLDataFormat*) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
+            nbSample = _converter->downsampleData ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
 
 	    // captureRsmplFile->write ((const char *)rsmpl_out, nbSample*sizeof(SFLDataFormat));
 
@@ -959,8 +959,11 @@ void PulseLayer::readFromMic (void)
 
         } else {
 
+	    // echo cancellation processing
+	  int sampleready = _echoCanceller->processAudio((SFLDataFormat *)data, echoCancelledMic, r);
+
             // no resampling required
-            getMainBuffer()->putData ( (void*) data, r, 100);
+            getMainBuffer()->putData (echoCancelledMic, sampleready*sizeof (SFLDataFormat), 100);
         }