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

webrtc/modules/rtp_rtcp/source/rtp_sender_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 214 matching lines @@
   if (payloadSize == 0) {
     return -1;
   }
 
   rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       videoType, _rtpSender.MaxDataPayloadLength(),
       video_header ? &(video_header->codecHeader) : nullptr, frameType));
 
   StorageType storage;
   bool fec_enabled;
+  bool first_frame = first_frame_sent_();
   {
     CriticalSectionScoped cs(crit_.get());
     FecProtectionParams* fec_params =
         frameType == kVideoFrameKey ? &key_fec_params_ : &delta_fec_params_;
     producer_fec_.SetFecParameters(fec_params, 0);
     storage = packetizer->GetStorageType(_retransmissionSettings);
     fec_enabled = fec_enabled_;
   }
 
   // Register CVO rtp header extension at the first time when we receive a frame
   // with pending rotation.
   RTPSenderInterface::CVOMode cvo_mode = RTPSenderInterface::kCVONone;
   if (video_header && video_header->rotation != kVideoRotation_0) {
     cvo_mode = _rtpSender.ActivateCVORtpHeaderExtension();
   }
 
   uint16_t rtp_header_length = _rtpSender.RTPHeaderLength();
   size_t payload_bytes_to_send = payloadSize;
   const uint8_t* data = payloadData;
 
   // TODO(changbin): we currently don't support to configure the codec to
   // output multiple partitions for VP8. Should remove below check after the
   // issue is fixed.
   const RTPFragmentationHeader* frag =
       (videoType == kRtpVideoVp8) ? NULL : fragmentation;
 
   packetizer->SetPayloadData(data, payload_bytes_to_send, frag);
 
+  bool first = true;
   bool last = false;
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
       return -1;
     }
+
     // Write RTP header.
     // Set marker bit true if this is the last packet in frame.
     _rtpSender.BuildRTPheader(
         dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5:
     // The MTSI client shall add the payload bytes as defined in this clause
     // onto the last RTP packet in each group of packets which make up a key
     // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
@@ skipping 21 matching lines @@
     if (fec_enabled) {
       SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
                            rtp_header_length, _rtpSender.SequenceNumber(),
                            captureTimeStamp, capture_time_ms, storage,
                            packetizer->GetProtectionType() == kProtectedPacket);
     } else {
       SendVideoPacket(dataBuffer, payload_bytes_in_packet, rtp_header_length,
                       _rtpSender.SequenceNumber(), captureTimeStamp,
                       capture_time_ms, storage);
     }
+
+    if (first_frame) {
+      if (first) {
+        LOG(LS_INFO)
+            << "Sent first RTP packet of the first video frame (pre-pacer)";
+      }
+      if (last) {
+        LOG(LS_INFO)
+            << "Sent last RTP packet of the first video frame (pre-pacer)";
+      }
+    }
+    first = false;
   }
 
   TRACE_EVENT_ASYNC_END1(
       "webrtc", "Video", capture_time_ms, "timestamp", _rtpSender.Timestamp());
   return 0;
 }
 
 void RTPSenderVideo::ProcessBitrate() {
   _videoBitrate.Process();
   _fecOverheadRate.Process();
@@ skipping 11 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   CriticalSectionScoped cs(crit_.get());
   _retransmissionSettings = settings;
 }
 
 }  // namespace webrtc