renaming of jitter_buf in receive_buf, wallclock_ts (to be continued)

git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@308 8eb893ce-cfd4-0310-b710-fb5ebe64c474

core/AmMediaProcessor.cpp

@@ -178,6 +178,7 @@ void AmMediaProcessorThread::on_stop()
 void AmMediaProcessorThread::run()
 {
   struct timeval now,next_tick,diff,tick;
+  // wallclock time
   unsigned int ts = 0;
 
   tick.tv_sec  = 0;

core/AmRtpAudio.cpp

@@ -69,9 +69,9 @@ unsigned int AmRtpAudio::bytes2samples(unsigned int bytes) const
   return AmAudio::bytes2samples(bytes);
 }
 /* 
-   @param audio_buffer_ts [in]    the current ts in the audio buffer 
+   @param wallclock_ts [in]    the current ts in the audio buffer 
 */
-int AmRtpAudio::receive(unsigned int audio_buffer_ts) 
+int AmRtpAudio::receive(unsigned int wallclock_ts) 
 {
   int size;
   unsigned int rtp_ts;
@@ -97,7 +97,7 @@ int AmRtpAudio::receive(unsigned int audio_buffer_ts)
       return -1;
     }
 
-    playout_buffer->write(audio_buffer_ts, rtp_ts, (ShortSample*)((unsigned char *)samples), 
+    playout_buffer->write(wallclock_ts, rtp_ts, (ShortSample*)((unsigned char *)samples), 
 			  PCM16_B2S(size), begin_talk);
   }
   return size;

core/AmRtpAudio.h

@@ -72,7 +72,7 @@ public:
 
   void setCurrentPayload(int payload);
 
-  int receive(unsigned int audio_buffer_ts);
+  int receive(unsigned int wallclock_ts);
 
   void setSendOnly(bool so){
     send_only = so;

core/AmRtpStream.cpp

@@ -403,9 +403,9 @@ void AmRtpStream::bufferPacket(const AmRtpPacket* p)
   if (!receiving && !passive)
     return;
 
-  jitter_mut.lock();
-  jitter_buf[p->timestamp].copy(p);
-  jitter_mut.unlock();
+  receive_mut.lock();
+  receive_buf[p->timestamp].copy(p);
+  receive_mut.unlock();
 }
 
 void AmRtpStream::clearRTPTimeout(struct timeval* recv_time) {
@@ -421,26 +421,26 @@ int AmRtpStream::nextPacket(AmRtpPacket& p)
   struct timeval diff;
   gettimeofday(&now,NULL);
 
-  jitter_mut.lock();
+  receive_mut.lock();
   timersub(&now,&last_recv_time,&diff);
   if(AmConfig::DeadRtpTime && 
      (diff.tv_sec > 0) &&
      ((unsigned int)diff.tv_sec > AmConfig::DeadRtpTime)){
     WARN("RTP Timeout detected. Last received packet is too old.\n");
     DBG("diff.tv_sec = %i\n",(unsigned int)diff.tv_sec);
-    jitter_mut.unlock();
+    receive_mut.unlock();
     return RTP_TIMEOUT;
   }
 
-  if(jitter_buf.empty()){
-    jitter_mut.unlock();
+  if(receive_buf.empty()){
+    receive_mut.unlock();
     return RTP_EMPTY;
   }
 
-  AmRtpPacket& pp = jitter_buf.begin()->second;
+  AmRtpPacket& pp = receive_buf.begin()->second;
   p.copy(&pp);
-  jitter_buf.erase(jitter_buf.begin());
-  jitter_mut.unlock();
+  receive_buf.erase(receive_buf.begin());
+  receive_mut.unlock();
 
   return 1;
 }

core/AmRtpStream.h

@@ -58,7 +58,7 @@ struct amci_payload_t;
 class AmAudio;
 class AmSession;
 class SdpPayload;
-typedef map<unsigned int, AmRtpPacket, ts_less> JitterBuffer;
+typedef map<unsigned int, AmRtpPacket, ts_less> ReceiveBuffer;
 
 /**
  * \brief RTP implementation
@@ -121,8 +121,8 @@ protected:
   auto_ptr<const SdpPayload> telephone_event_pt;
 
 
-  JitterBuffer    jitter_buf;
-  AmMutex         jitter_mut;
+  ReceiveBuffer   receive_buf;
+  AmMutex         receive_mut;
 
   /* get next packet in buffer */
   int nextPacket(AmRtpPacket& p);