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

talk/session/media/channel.h

Index: talk/session/media/channel.h
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 969f907928c5f6649bcbc0c6382b6d688d2d6a6f..c4c3c5adc505f1ad5ea0294c5ca454ccda2ad37c 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -52,6 +52,10 @@
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
 
+namespace webrtc {
+class MediaControllerInterface;
+}  // namespace webrtc
+
 namespace cricket {
 
 struct CryptoParams;
@@ -79,6 +83,7 @@ class BaseChannel
  public:
   BaseChannel(rtc::Thread* thread,
               MediaChannel* channel,
+              webrtc::MediaControllerInterface* media_controller,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
@@ -199,9 +204,8 @@ class BaseChannel
 
   // NetworkInterface implementation, called by MediaEngine
   virtual bool SendPacket(rtc::Buffer* packet,
-                          rtc::DiffServCodePoint dscp);
-  virtual bool SendRtcp(rtc::Buffer* packet,
-                        rtc::DiffServCodePoint dscp);
+                          const rtc::PacketOptions& options);
+  virtual bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options);
 
   // From TransportChannel
   void OnWritableState(TransportChannel* channel);
@@ -214,8 +218,9 @@ class BaseChannel
 
   bool PacketIsRtcp(const TransportChannel* channel, const char* data,
                     size_t len);
-  bool SendPacket(bool rtcp, rtc::Buffer* packet,
-                  rtc::DiffServCodePoint dscp);
+  bool SendPacket(bool rtcp,
+                  rtc::Buffer* packet,
+                  const rtc::PacketOptions& options);
   virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
   void HandlePacket(bool rtcp, rtc::Buffer* packet,
                     const rtc::PacketTime& packet_time);
@@ -261,7 +266,7 @@ class BaseChannel
   // Helper method to get RTP Absoulute SendTime extension header id if
   // present in remote supported extensions list.
   void MaybeCacheRtpAbsSendTimeHeaderExtension(
-    const std::vector<RtpHeaderExtension>& extensions);
+      const std::vector<RtpHeaderExtension>& extensions);
 
   bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
                        bool* dtls,
@@ -296,6 +301,7 @@ class BaseChannel
   rtc::Thread* worker_thread_;
   TransportController* transport_controller_;
   MediaChannel* media_channel_;
+  webrtc::MediaControllerInterface* const media_controller_;
   std::vector<StreamParams> local_streams_;
   std::vector<StreamParams> remote_streams_;
 
@@ -330,6 +336,7 @@ class VoiceChannel : public BaseChannel {
   VoiceChannel(rtc::Thread* thread,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
+               webrtc::MediaControllerInterface* media_controller,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
@@ -432,6 +439,7 @@ class VideoChannel : public BaseChannel {
  public:
   VideoChannel(rtc::Thread* thread,
                VideoMediaChannel* channel,
+               webrtc::MediaControllerInterface* media_controller,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
@@ -471,8 +479,6 @@ class VideoChannel : public BaseChannel {
   bool SendIntraFrame();
   bool RequestIntraFrame();
 
-  // Configure sending media on the stream with SSRC |ssrc|
-  // If there is only one sending stream SSRC 0 can be used.
   bool SetVideoSend(uint32 ssrc, bool enable, const VideoOptions* options);
 
  private: