*** empty log message ***

src/audio/audiolayer.cpp

@@ -411,6 +411,7 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
   // Additionally handle the mic's audio stream 
   micAvailPut = _micRingBuffer.AvailForPut();
   toPut = (micAvailPut <= (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBuffer * sizeof(SFLDataFormat);
+  //_debug("AL: Nb sample: %d char, [0]=%f [1]=%f [2]=%f\n", toPut, in[0], in[1], in[2]);
   _micRingBuffer.Put(in, toPut, micVolume);
 
   return paContinue;

src/audio/audiortp.cpp

@@ -234,12 +234,12 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
   //   3. encode it
   //   4. send it
   try {
-    if (_ca==0) { return; } // no call, so we do nothing
+    if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
     AudioLayer* audiolayer = Manager::instance().getAudioDriver();
-    if (!audiolayer) { return; }
+    if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }
 
     AudioCodec* audiocodec = _ca->getAudioCodec();
-    if (!audiocodec) { return; }
+    if (!audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
 
     // we have to get 20ms of data from the mic *20/1000 = /50
     // rate/50 shall be lower than RTP_20S_48KHZ_MAX
@@ -250,6 +250,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
 
     // take the lower
     int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+    //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
 
     // Get bytes from micRingBuffer to data_from_mic
     int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
@@ -289,8 +290,9 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
       //_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
       //_debug("has to fill: %d chars at %p\n", (RTP_20S_8KHZ_MAX-nbSample)*sizeof(int16), toSIP + nbSample);
       memset(toSIP + nbSample, 0, (RTP_20S_8KHZ_MAX-nbSample)*sizeof(int16));
-      //nbSample = RTP_20S_8KHZ_MAX;
+      nbSample = RTP_20S_8KHZ_MAX;
     }
+    //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
 
     // for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
     // codecEncode(char *dest, int16* src, size in bytes of the src)

src/global.h

@@ -58,8 +58,8 @@ typedef short int16;
   #define _debugInit(...)
 #endif
 
-#define SFLPHONED_VERSION "0.6.3"
-#define SFLPHONED_VERSIONNUM 0x000603
+#define SFLPHONED_VERSION "0.7.0"
+#define SFLPHONED_VERSIONNUM 0x000700
 
 #define PROGNAME         "sflphoned"
 #define PROGNAME_GLOBAL  "sflphone"