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

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.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 52 matching lines @@
 
   RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
   const size_t payload_data_length =
       payload_length - rtp_header->header.paddingLength;
 
   if (payload == NULL || payload_data_length == 0) {
     return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
                                                                            : -1;
   }
 
+  if (first_packet_received_()) {
+    LOG(LS_INFO) << "Received first video RTP packet";
+  }
+
   // We are not allowed to hold a critical section when calling below functions.
   rtc::scoped_ptr<RtpDepacketizer> depacketizer(
       RtpDepacketizer::Create(rtp_header->type.Video.codec));
   if (depacketizer.get() == NULL) {
     LOG(LS_ERROR) << "Failed to create depacketizer.";
     return -1;
   }
 
   rtp_header->type.Video.isFirstPacket = is_first_packet;
   RtpDepacketizer::ParsedPayload parsed_payload;
@@ skipping 30 matching lines @@
     RtpFeedback* callback,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
   // TODO(pbos): Remove as soon as audio can handle a changing payload type
   // without this callback.
   return 0;
 }
 
 }  // namespace webrtc