Modified PeerConnection and WebRtcSession 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 12 matching lines @@
 #include "webrtc/api/statstypes.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #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"
 
+#ifdef HAVE_QUIC
+#include "webrtc/api/quicdatatransport.h"
+#endif  // HAVE_QUIC
+
 namespace cricket {
 
 class ChannelManager;
 class DataChannel;
 class StatsReport;
 class VideoChannel;
 class VoiceChannel;
 
+#ifdef HAVE_QUIC
+class QuicTransportChannel;
+#endif  // HAVE_QUIC
+
 }  // namespace cricket
 
 namespace webrtc {
 
 class IceRestartAnswerLatch;
 class JsepIceCandidate;
 class MediaStreamSignaling;
 class WebRtcSessionDescriptionFactory;
 
 extern const char kBundleWithoutRtcpMux[];
@@ skipping 86 matching lines @@
   };
 
   WebRtcSession(webrtc::MediaControllerInterface* media_controller,
                 rtc::Thread* network_thread,
                 rtc::Thread* worker_thread,
                 rtc::Thread* signaling_thread,
                 cricket::PortAllocator* port_allocator);
   virtual ~WebRtcSession();
 
   // These are const to allow them to be called from const methods.
+  rtc::Thread* network_thread() const { return network_thread_; }
   rtc::Thread* worker_thread() const { return worker_thread_; }
   rtc::Thread* signaling_thread() const { return signaling_thread_; }
 
   // The ID of this session.
   const std::string& id() const { return sid_; }
 
   bool Initialize(
       const PeerConnectionFactoryInterface::Options& options,
       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
       const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
@@ skipping 135 matching lines @@
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
   // Called when the whole session is destroyed.
   sigslot::signal0<> SignalDestroyed;
 
   // Called when a valid data channel OPEN message is received.
   // std::string represents the data channel label.
   sigslot::signal2<const std::string&, const InternalDataChannelInit&>
       SignalDataChannelOpenMessage;
+#ifdef HAVE_QUIC
+  QuicDataTransport* quic_data_transport() {
+    return quic_data_transport_.get();
+  }
+#endif  // HAVE_QUIC
 
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
     kOffer,
     kPrAnswer,
     kAnswer,
   };
 
@@ skipping 124 matching lines @@
   // Reports stats for all transports in use.
   void ReportTransportStats();
 
   // Gather the usage of IPv4/IPv6 as best connection.
   void ReportBestConnectionState(const cricket::TransportStats& stats);
 
   void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
   void OnSentPacket_w(const rtc::SentPacket& sent_packet);
 
+  const std::string GetTransportName(const std::string& content_name);
+
+  rtc::Thread* const network_thread_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const signaling_thread_;
 
   State state_ = STATE_INIT;
   Error error_ = ERROR_NONE;
   std::string error_desc_;
 
   const std::string sid_;
   bool initial_offerer_ = false;
 
@@ skipping 31 matching lines @@
 
   // Declares the bundle policy for the WebRTCSession.
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   bool received_first_video_packet_ = false;
   bool received_first_audio_packet_ = false;
 
+#ifdef HAVE_QUIC
+  std::unique_ptr<QuicDataTransport> quic_data_transport_;
+#endif  // HAVE_QUIC
+
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_