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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 2aa4961cdc77d98490c607ed23945617228bf760..850e9b1575958e74727dea458428f616c0da10ad 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -12,6 +12,7 @@
 
 #include <string.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -45,7 +46,8 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock,
       _cngSWBPayloadType(-1),
       _cngFBPayloadType(-1),
       _lastPayloadType(-1),
-      _audioLevel_dBov(0) {}
+      _audioLevel_dBov(0),
+      _firstPacketSent(false) {}
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
@@ -342,9 +344,15 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
       memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
   }
+
+  bool firstPacket = false;
   {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     _lastPayloadType = payloadType;
+    if (_firstPacketSent) {

[pthatcher1, 2016/03/16 04:55:27]

Should this be if(!_firstPacketSent) ?

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

Yes.

[skvlad, 2016/03/16 19:07:04]

Good catch, fixed!

+      firstPacket = true;
+      _firstPacketSent = true;
+    }
   }
   // Update audio level extension, if included.
   size_t packetSize = payloadSize + rtpHeaderLength;
@@ -357,10 +365,14 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                          _rtpSender->Timestamp(), "seqnum",
                          _rtpSender->SequenceNumber());
-  return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                   TickTime::MillisecondTimestamp(),
-                                   kAllowRetransmission,
-                                   RtpPacketSender::kHighPriority);
+  int32_t sendResult = _rtpSender->SendToNetwork(
+      dataBuffer, payloadSize, rtpHeaderLength,
+      TickTime::MillisecondTimestamp(), kAllowRetransmission,
+      RtpPacketSender::kHighPriority);
+  if (firstPacket) {
+    LOG(LS_INFO) << "First audio RTP packet sent";
+  }
+  return sendResult;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet