Recording now in audiortp::run() method

src/audio/audiortp.cpp

@@ -269,6 +269,10 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
     //_debug("get data from mic\n");
     int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
     int nb_sample_up = nbSample;
+    
+    // Store the length of the mic buffer in samples for recording
+    _nSamplesMic = nbSample;
+
     int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
 
     //_debug("resample data\n");
@@ -349,9 +353,9 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
 #ifdef DATAFORMAT_IS_FLOAT
 #else
 #endif
-      
-      // Record before sending to audio
-      recAudio.recData(spkrDataConverted,nbSample);
+    
+      // Stor the number of samples for recording
+      _nSamplesSpkr = nbSample;
         
       audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
 
@@ -432,6 +436,9 @@ AudioRtpRTX::run () {
       ////////////////////////////
       receiveSessionForSpkr(countTime);
       // Let's wait for the next transmit cycle
+
+      recAudio.recData(spkrDataConverted,_nSamplesSpkr);
+      
       Thread::sleep(TimerPort::getTimer());
       TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
     }

src/audio/audiortp.h

@@ -40,7 +40,6 @@
  * @brief Manage the real-time data transport in a SIP call
  */
 
-typedef unsigned char* samplePtr;
 
 class SIPCall;
 
@@ -119,6 +118,18 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
     /** Length of the sound frame we capture in ms(typically 20ms) */
     int _layerFrameSize; 
 
+    /** Speaker buffer length in samples once the data are resampled
+     *  (used for mixing and recording)
+     */
+    int _nSamplesSpkr;
+
+    /** Mic buffer length in samples once the data are resampled
+     *  (used for mixing and recording)
+     */
+    int _nSamplesMic;
+    
+  
+    
     /**
      * Init the RTP session. Create either symmetric or double sessions to manage data transport
      * Set the payloads according to the manager preferences

src/audio/pulselayer.cpp

@@ -93,10 +93,6 @@ PulseLayer::connectPulseAudioServer( void )
   //muteAudioApps(99);
   _debug("Context creation done\n");
 
-  //FILE_TYPE ft = FILE_WAV;
-  //SOUND_FORMAT sf = INT16;
-  //audioRec.setSndSamplingRate(44100);
-  //audioRec.openFile("SFLWavFile.wav",ft,sf);
 }
 
 void PulseLayer::context_state_callback( pa_context* c, void* user_data )
@@ -249,12 +245,6 @@ PulseLayer::startStream (void)
   pa_stream_cork( record->pulseStream(), NULL, NULL, NULL);
   pa_threaded_mainloop_unlock(m);
 
-  _debug("Opening the wave file\n");
-  FILE_TYPE ft = FILE_WAV;
-  SOUND_FORMAT sf = INT16;
-  audioRec.setSndSamplingRate(44100);
-  audioRec.openFile("SFLWavFile.wav",ft,sf); 
-
 }
 
   void 
@@ -264,9 +254,6 @@ PulseLayer::stopStream (void)
   pa_stream_flush( playback->pulseStream(), NULL, NULL );
   pa_stream_flush( record->pulseStream(), NULL, NULL );
   flushMic();
-
-  _debug("Closing the wav file\n");
-  audioRec.closeFile();
 }
 
 
@@ -305,7 +292,8 @@ PulseLayer::processData( void )
   }
 }
 
-void PulseLayer::writeToSpeaker( void )
+void
+PulseLayer::writeToSpeaker( void )
 {   
   /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
   int urgentAvail; 

src/audio/pulselayer.h

@@ -22,7 +22,6 @@
 
 #include "audiolayer.h"
 #include "audiostream.h"
-#include "plug-in/audiorecorder/audiorecord.h"
 
 #define PLAYBACK_STREAM_NAME	    "SFLphone out"
 #define CAPTURE_STREAM_NAME	    "SFLphone in"
@@ -256,10 +255,6 @@ class PulseLayer : public AudioLayer {
     int spkrVolume;
     int micVolume;
 
-    /**
-     * Audio device to record audio streams on disk
-     */ 
-    AudioRecord audioRec;
 
 public: 
     static int streamState;

src/audio/ringbuffer.cpp

@@ -82,7 +82,8 @@ RingBuffer::Put(void* buffer, int toCopy, unsigned short volume) {
       toCopy = (mBufferSize-4) - len;
 
    src = (samplePtr) buffer;
-
+   
+  
    copied = 0;
    pos = mEnd;
 

src/plug-in/audiorecorder/audiorecord.cpp

@@ -163,7 +163,7 @@ void AudioRecord::recData(SFLDataFormat* buffer, int nSamples) {
   // printf("AudioRecord : buffer: %x : ", buffer);
  
   if ( sndFormat_ == INT16 ) { // TODO change INT16 to SINT16
-    if ( fwrite(buffer, nSamples*sizeof(SFLDataFormat), 1, fp) != 1)
+    if ( fwrite(buffer, sizeof(SFLDataFormat), nSamples, fp) != nSamples)
       _debug("AudioRecord: Could not record data!\n");
     else {
       // printf("Buffer : %x \n",*buffer);