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

talk/app/webrtc/mediacontroller.cc

 /*
  * libjingle
  * Copyright 2015 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 12 matching lines @@
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/app/webrtc/mediacontroller.h"
 
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/call.h"
+#include "webrtc/p2p/base/transportchannel.h"
 
 namespace {
 
 const int kMinBandwidthBps = 30000;
 const int kStartBandwidthBps = 300000;
 const int kMaxBandwidthBps = 2000000;
 
-class MediaController : public webrtc::MediaControllerInterface {
+class MediaController : public webrtc::MediaControllerInterface,
+                        public sigslot::has_slots<> {
  public:
   MediaController(rtc::Thread* worker_thread,
                   webrtc::VoiceEngine* voice_engine)
       : worker_thread_(worker_thread) {
     RTC_DCHECK(nullptr != worker_thread);
     worker_thread_->Invoke<void>(
         rtc::Bind(&MediaController::Construct_w, this, voice_engine));
   }
   ~MediaController() override {
     worker_thread_->Invoke<void>(
         rtc::Bind(&MediaController::Destruct_w, this));
+    RTC_DCHECK(transport_channels_.empty());
   }
 
   webrtc::Call* call_w() override {
     RTC_DCHECK(worker_thread_->IsCurrent());
     return call_.get();
   }
 
+  void ConnectTransportChannel(
+      cricket::TransportChannel* transport_channel) override {

[pthatcher1, 2015/10/07 16:44:37]

Needs a RTC_DCHECK(worker_thread_->IsCurrent());

[stefan-webrtc, 2015/10/08 12:53:45]

Done.

+    if (transport_channels_.find(transport_channel) !=
+        transport_channels_.end())
+      return;
+    transport_channels_.insert(transport_channel);
+    transport_channel->SignalSentPacket.connect(this,
+                                                &MediaController::OnSentPacket);
+  }

[pthatcher1, 2015/10/07 16:44:37]

Can't you do this?

if (!transport_channels_.insert(transport_channel)->second) {
  return;
}
transport_channel->SignalSentPacket.connect(this,
&MediaController::OnSentPacket);

[stefan-webrtc, 2015/10/08 12:53:45]

Done.

+
+  void DisconnectTransportChannel(
+      cricket::TransportChannel* transport_channel) override {
+    RTC_DCHECK(transport_channels_.find(transport_channel) !=
+               transport_channels_.end());
+    transport_channels_.erase(transport_channel);
+    transport_channel->SignalSentPacket.disconnect(this);
+  }
+
  private:
   void Construct_w(webrtc::VoiceEngine* voice_engine)  {
     RTC_DCHECK(worker_thread_->IsCurrent());
     webrtc::Call::Config config;
     config.voice_engine = voice_engine;
     config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
     config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
     config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
     call_.reset(webrtc::Call::Create(config));
   }
   void Destruct_w() {
     RTC_DCHECK(worker_thread_->IsCurrent());
     call_.reset(nullptr);
   }
+  void OnSentPacket(cricket::TransportChannel* channel,
+                    const rtc::SentPacket& sent_packet) {
+    call_->OnSentPacket(sent_packet);
+  }
 
   rtc::Thread* worker_thread_;
   rtc::scoped_ptr<webrtc::Call> call_;
+  std::set<cricket::TransportChannel*> transport_channels_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
 };
 } // namespace {
 
 namespace webrtc {
 
 MediaControllerInterface* MediaControllerInterface::Create(
     rtc::Thread* worker_thread, webrtc::VoiceEngine* voice_engine) {
   return new MediaController(worker_thread, voice_engine);
 }
 } // namespace webrtc