Wire up transport sequence number / send time callbacks to webrtc via libjingle.

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 543 matching lines @@
   video_codecs_ = GetSupportedCodecs();
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
                          kRtpTimestampOffsetHeaderExtensionDefaultId));
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
                          kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
                          kRtpVideoRotationHeaderExtensionDefaultId));
+  rtp_header_extensions_.push_back(
+      RtpHeaderExtension(kRtpTransportSequenceNumberHeaderExtension,
+                         kRtpTransportSequenceNumberHeaderExtensionDefaultId));
 }
 
 WebRtcVideoEngine2::~WebRtcVideoEngine2() {
   LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
 }
 
 void WebRtcVideoEngine2::Init() {
   LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
   initialized_ = true;
 }
@@ skipping 894 matching lines @@
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   if (call_->Receiver()->DeliverPacket(
           webrtc::MediaType::VIDEO,
           reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
           webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
   }
 }
 
+void WebRtcVideoChannel2::OnPacketSent(const rtc::SentPacket& packet_sent) {
+  webrtc::SentPacket webrtc_packet_sent(packet_sent.transport_sequence_number,
+                                        packet_sent.send_time_ms);
+
+  call_->OnSentPacket(webrtc_packet_sent);

[pthatcher1, 2015/09/25 23:24:57]

Why not just propogate from the TransportChannel to Call_ directly (maybe going
through BaseChannel)?  Why go through the MediaChannel?

[stefan-webrtc, 2015/09/28 12:10:49]

Hm, this is probably an artifact of my lack of overview of libjingle code. I
basically followed what was being done with OnReadyToSend().

[pthatcher1, 2015/09/28 23:58:40]

Not your fault, but OnReadyToSend needs to be cleaned up.  It's not a very good
model for copying.  

[stefan-webrtc, 2015/10/02 13:29:12]

Done.

+}
+
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
   RTC_DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
@@ skipping 1249 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO