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

webrtc/modules/rtp_rtcp/source/rtp_receiver_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.
  */
 
@@ skipping 176 matching lines @@
                rtp_header->header.timestamp);
   rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs;
   num_energy_ = rtp_header->type.Audio.numEnergy;
   if (rtp_header->type.Audio.numEnergy > 0 &&
       rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
     memcpy(current_remote_energy_,
            rtp_header->type.Audio.arrOfEnergy,
            rtp_header->type.Audio.numEnergy);
   }
 
+  if (first_packet_received_()) {
+    LOG(LS_INFO) << "Received first audio RTP packet";
+  }
+
   return ParseAudioCodecSpecific(rtp_header,
                                  payload,
                                  payload_length,
                                  specific_payload.Audio,
                                  is_red);
 }
 
 int RTPReceiverAudio::GetPayloadTypeFrequency() const {
   CriticalSectionScoped lock(crit_sect_.get());
   if (last_received_g722_) {
@@ skipping 166 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc