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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
 #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"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
@@ skipping 301 matching lines @@
       dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
       memcpy(dataBuffer + rtpHeaderLength,
              payloadData + fragmentation->fragmentationOffset[0],
              fragmentation->fragmentationLength[0]);
 
       payloadSize = fragmentation->fragmentationLength[0];
     } else {
       memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
   }
+
   {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     _lastPayloadType = payloadType;
   }
   // Update audio level extension, if included.
   size_t packetSize = payloadSize + rtpHeaderLength;
   RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
   _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                (frameType == kAudioFrameSpeech),
                                audio_level_dbov);
   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 send_result = _rtpSender->SendToNetwork(
+      dataBuffer, payloadSize, rtpHeaderLength,
+      TickTime::MillisecondTimestamp(), kAllowRetransmission,
+      RtpPacketSender::kHighPriority);
+  if (first_packet_sent_()) {
+    LOG(LS_INFO) << "First audio RTP packet sent to pacer";
+  }
+  return send_result;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
 int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
   if (level_dBov > 127) {
     return -1;
   }
   CriticalSectionScoped cs(_sendAudioCritsect.get());
   _audioLevel_dBov = level_dBov;
   return 0;
@@ skipping 81 matching lines @@
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
         dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc