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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 59 matching lines @@
 
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(NetworkState state) override;
 
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
 
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
@@ skipping 313 matching lines @@
     }
   }
   {
     ReadLockScoped write_lock(*receive_crit_);
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
 }
 
+void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  channel_group_->OnSentPacket(sent_packet);
+}
+
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (config_.voice_engine == nullptr || sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
   // Find existing audio stream.
   const auto it = sync_stream_mapping_.find(sync_group);
   if (it != sync_stream_mapping_.end()) {
     sync_audio_stream = it->second;
@@ skipping 108 matching lines @@
     size_t length,
     const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc