Added log messages for important call setup events: first packet sent/received

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 5950dc75545fc713ed7e16df14b4d150bb80db2a..b9bf8e97e5bfe3405f01fe0b3c316eede9a20f65 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -36,7 +36,8 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver(
   return new RtpReceiverImpl(
       clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
       rtp_payload_registry,
-      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
+      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback),
+      false /*is_audio*/);
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
@@ -55,7 +56,8 @@ RtpReceiver* RtpReceiver::CreateAudioReceiver(
       clock, incoming_audio_feedback, incoming_messages_callback,
       rtp_payload_registry,
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
-                                               incoming_audio_feedback));
+                                               incoming_audio_feedback),
+      true /*is_audio*/);
 }
 
 RtpReceiverImpl::RtpReceiverImpl(
@@ -63,10 +65,12 @@ RtpReceiverImpl::RtpReceiverImpl(
     RtpAudioFeedback* incoming_audio_messages_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry,
-    RTPReceiverStrategy* rtp_media_receiver)
+    RTPReceiverStrategy* rtp_media_receiver,
+    bool is_audio)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
+      is_audio_(is_audio),
       cb_rtp_feedback_(incoming_messages_callback),
       critical_section_rtp_receiver_(
           CriticalSectionWrapper::CreateCriticalSection()),
@@ -185,6 +189,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(
 
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
+
   bool is_first_packet_in_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
@@ -205,9 +210,11 @@ bool RtpReceiverImpl::IncomingRtpPacket(
     return false;
   }
 
+  bool is_first_packet_in_first_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
+    is_first_packet_in_first_frame = (last_receive_time_ == 0);
     last_receive_time_ = clock_->TimeInMilliseconds();
     last_received_payload_length_ = payload_data_length;
 
@@ -219,6 +226,12 @@ bool RtpReceiverImpl::IncomingRtpPacket(
       last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
+
+  if (is_first_packet_in_first_frame) {

[mflodman, 2016/03/15 11:05:45]

I'd prefer to put this logic in RTPReceiver[Audio/Video]::ParseRtpPacket and
avoid adding is_audio_ to this class.

[pthatcher1, 2016/03/15 16:44:30]

I think it's strange that a parse method would modify state, but looking at the
code, it seems like these methods are more like "ProcessRtpPacket", since they
do more than just parse.

Note that we'll also have to add some state in the RTPRecieverAudio/Video to
keep track if it's seen a packet or not (like the last_receive_time_) is being
used here.

[skvlad, 2016/03/16 02:05:56]

Done.

[mflodman, 2016/03/16 14:30:54]

True, but I do prefer that over adding back a audio/video state in this class.

+    LOG(LS_INFO) << "Received first RTP packet of the first " <<
+      (is_audio_ ? "audio" : "video") << " frame";
+  }
+
   return true;
 }