Modify PeerConnection for end-to-end QuicDataChannel usage

webrtc/api/webrtcsession.h

 /*
  *  Copyright 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 15 matching lines @@
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/pc/mediasession.h"
 
 namespace cricket {
 
 class ChannelManager;
 class DataChannel;
+class QuicTransportChannel;
 class StatsReport;
 class VideoCapturer;
 class VideoChannel;
 class VoiceChannel;
 
 }  // namespace cricket
 
 namespace webrtc {
 
 class IceRestartAnswerLatch;
@@ skipping 260 matching lines @@
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
+  cricket::QuicTransportChannel* quic_transport_channel() const;

[pthatcher1, 2016/03/30 20:34:49]

Doesn't this need USE_QUIC or HAVE_QUIC as well (and the class reference above)?

[Taylor Brandstetter, 2016/04/01 23:23:42]

It's possible that this is NULL when it's needed. So you need an equivalent of
DataChannelProviderInterface in order to tell a QuicDataChannel when the
underlying transport channel is created or destroyed.

 
   // Called when voice_channel_, video_channel_ and data_channel_ are created
   // and destroyed. As a result of, for example, setting a new description.
   sigslot::signal0<> SignalVoiceChannelCreated;
   sigslot::signal0<> SignalVoiceChannelDestroyed;
   sigslot::signal0<> SignalVideoChannelCreated;
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
   // Called when the whole session is destroyed.
@@ skipping 187 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_