[#1966] Add timing info in pulseaudio layer

sflphone-common/src/audio/echocancel.cpp

@@ -188,12 +188,14 @@ void EchoCancel::putData(SFLDataFormat *inputData, int nbBytes)
 {
   // std::cout << "putData nbBytes: " << nbBytes << std::endl;
 
+  /*
   if(_spkrStoped) {
       _debug("EchoCancel: Flush data");
       _micData->flushAll();
       _spkrData->flushAll();
       _spkrStoped = false;
   }
+  */
 
   // Put data in speaker ring buffer
   _spkrData->Put(inputData, nbBytes);
@@ -204,10 +206,11 @@ void EchoCancel::process(SFLDataFormat *data, int nbBytes) {}
 
 int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes)
 {
-
+  /*
   if(_spkrStoped) {
     return 0;
   }
+  */
 
   int byteSize = _smplPerFrame*sizeof(SFLDataFormat);
 
@@ -357,7 +360,7 @@ void EchoCancel::updateEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDat
   if(_adaptCnt == _micAdaptSize) {
     _debug("EchoCancel: Echo path adaptation completed");
     _adaptDone = true;
-    _amplDelayIndexOut = getMaximumIndex(_correlationArray, _correlationSize);
+    _amplDelayIndexOut = 0;// getMaximumIndex(_correlationArray, _correlationSize);
     _debug("EchoCancel: Echo length %d", _amplDelayIndexOut);
     spkrLearningData->write((const char *)_spkrAdaptArray, _spkrAdaptSize*sizeof(int));
     micLearningData->write((const char *)_micAdaptArray, _micAdaptSize*sizeof(int));

sflphone-common/src/audio/echocancel.h

@@ -35,7 +35,7 @@
 #define ECHO_LENGTH 50
 
 // Voice level threashold 
-#define MIN_SIG_LEVEL 100
+#define MIN_SIG_LEVEL 1000
 
 // Delay between mic and speaker
 #define DELAY_AMPLIFY 60

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

@@ -145,6 +145,13 @@ AudioStream::stream_state_callback (pa_stream* s, void* user_data)
 
         case PA_STREAM_READY:
             _info ("Audio: Stream successfully created, connected to %s", pa_stream_get_device_name (s));
+	    // pa_buffer_attr *buffattr = (pa_buffer_attr *)pa_xmalloc (sizeof(pa_buffer_attr));
+	    _debug("Audio: maxlength %u", pa_stream_get_buffer_attr(s)->maxlength);
+	    _debug("Audio: tlength %u", pa_stream_get_buffer_attr(s)->tlength);
+	    _debug("Audio: prebug %u", pa_stream_get_buffer_attr(s)->prebuf);
+	    _debug("Audio: minreq %u", pa_stream_get_buffer_attr(s)->minreq);
+	    _debug("Audio: fragsize %u", pa_stream_get_buffer_attr(s)->fragsize);
+	    // pa_xfree (buffattr);
             break;
 
         case PA_STREAM_UNCONNECTED:
@@ -210,7 +217,9 @@ AudioStream::createStream (pa_context* c, std::string *deviceName)
     } else if (_streamType == CAPTURE_STREAM) {
 
         attributes->maxlength = (uint32_t) -1;
-        attributes->fragsize = pa_usec_to_bytes (20 * PA_USEC_PER_MSEC, &_sample_spec);
+	attributes->tlength = pa_usec_to_bytes (20 * PA_USEC_PER_MSEC, &_sample_spec);
+	attributes->prebuf = 0;
+        attributes->fragsize = pa_usec_to_bytes (100 * PA_USEC_PER_MSEC, &_sample_spec);
 
 	pa_threaded_mainloop_lock(_mainloop);
 

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

@@ -66,19 +66,16 @@ static void pa_success_callback(pa_context *c, int success, void *userdata) {
 
 static void latency_update_callback(pa_stream *p, void *userdata) {
 
-  pa_usec_t r_usec;
-  pa_buffer_attr *buffattr;
+    pa_usec_t r_usec;
 
-  pa_stream_get_latency (p, &r_usec, NULL);	
+    pa_stream_get_latency (p, &r_usec, NULL);	
 
-  // buffattr = pa_stream_get_buffer_attr(p);   	
-  
     _debug("Audio: Stream letency update %0.0f ms for device %s", (float)r_usec/1000, pa_stream_get_device_name(p));
-  // _debug("Audio: maxlength %d", buffattr->maxlength);
-  // _debug("Audio: tlength %d", buffattr->tlength);
-  // _debug("Audio: prebug %d", buffattr->prebuf);
-  // _debug("Audio: minreq %d", buffattr->minreq);
-  // _debug("Audio: fragsize %d", buffattr->fragsize);
+    _debug("Audio: maxlength %u", pa_stream_get_buffer_attr(p)->maxlength);
+    _debug("Audio: tlength %u", pa_stream_get_buffer_attr(p)->tlength);
+    _debug("Audio: prebuf %u", pa_stream_get_buffer_attr(p)->prebuf);
+    _debug("Audio: minreq %u", pa_stream_get_buffer_attr(p)->minreq);
+    _debug("Audio: fragsize %u", pa_stream_get_buffer_attr(p)->fragsize);
   
 }
 
@@ -558,6 +555,7 @@ bool PulseLayer::createStreams (pa_context* c)
     pa_stream_set_latency_update_callback(record->pulseStream(), latency_update_callback, this);
     delete recordParam;
   
+    /*
     PulseLayerType * ringtoneParam = new PulseLayerType();
     ringtoneParam->context = c;
     ringtoneParam->type = RINGTONE_STREAM;
@@ -575,7 +573,7 @@ bool PulseLayer::createStreams (pa_context* c)
     pa_stream_set_write_callback(ringtone->pulseStream(), ringtone_callback, this);
     pa_stream_set_moved_callback(ringtone->pulseStream(), stream_moved_callback, this);
     delete ringtoneParam;
-
+    */
     pa_threaded_mainloop_signal (m , 0);
 
     flushMain();
@@ -943,6 +941,7 @@ void PulseLayer::readFromMic (void)
 	    // echo cancellation processing
 	    int sampleready = _echoCanceller->processAudio(rsmpl_out, echoCancelledMic, nbSample*sizeof(SFLDataFormat));
 
+	    _debug("sampleready: %d", sampleready);
             // getMainBuffer()->putData ( (void*) rsmpl_out, nbSample*sizeof (SFLDataFormat), 100);
 	    if(sampleready)
 	      getMainBuffer()->putData ( echoCancelledMic, sampleready*sizeof (SFLDataFormat), 100);