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

talk/app/webrtc/webrtcsession.cc

Index: talk/app/webrtc/webrtcsession.cc
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 2ab9a1e6969599983724f9ffdce7aebb91b5d418..83025b0cbd5478340f23f41eb451ed4962a00084 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -45,6 +45,7 @@
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
@@ -63,6 +64,8 @@ using cricket::STUN_PORT_TYPE;
 using cricket::RELAY_PORT_TYPE;
 using cricket::PRFLX_PORT_TYPE;
 
+using rtc::Bind;
+
 namespace webrtc {
 
 // Error messages
@@ -1764,19 +1767,23 @@ void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
 bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
   // Creating the media channels and transport proxies.
   const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
+  bool voice_channel_created = false;
   if (voice && !voice->rejected && !voice_channel_) {
     if (!CreateVoiceChannel(voice)) {
       LOG(LS_ERROR) << "Failed to create voice channel.";
       return false;
     }
+    voice_channel_created = true;
   }
 
   const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
+  bool video_channel_created = false;
   if (video && !video->rejected && !video_channel_) {
     if (!CreateVideoChannel(video)) {
       LOG(LS_ERROR) << "Failed to create video channel.";
       return false;
     }
+    video_channel_created = true;
   }
 
   const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
@@ -1814,6 +1821,17 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
     }
   }
 
+  if (voice_channel_created) {

[pthatcher1, 2015/10/08 20:27:55]

It's not necessary to put this after the bundle code, since bundle may be
inactive at this point and then activated much later.  I think it would be more
simple to remove the "created" bools and just put the
ConnectToSignalSentPacket_w in the CreateVideoChannel and CreateVoiceChannel
methods.

Furthermore, we need to do one of the following:

A.  Call ConnectToSignalSentPacket for the DataChannel.
B.  Call ConnectToSignalSentPacket at the end of EnableBundle for both the
VoiceChannel and the VideoChannel. 

Otherwise if audio and video got bundled over the data channel's transport, the
signal wouldn't be hooked up.  In fact, we should probably have a unit test for
that scenario.

[stefan-webrtc, 2015/10/09 14:18:44]

Right, makes sense.

I didn't figure out a good way of testing this though. I'm not quite sure how to
setup a data channel and then later add video and audio and have them be bundled
at a later stage.

If you could take a look at the test I wrote for the regular case and based on
that give me some suggestions, that would be great. The WebRtcSession code is
fairly unknown territory to me, so figuring out how to write such a test takes a
lot of time. :)

+    worker_thread()->Invoke<void>(
+        Bind(&MediaControllerInterface::ConnectToSignalSentPacket_w,
+             media_controller_.get(), voice_channel_->transport_channel()));
+  }
+  if (video_channel_created) {
+    worker_thread()->Invoke<void>(
+        Bind(&MediaControllerInterface::ConnectToSignalSentPacket_w,
+             media_controller_.get(), video_channel_->transport_channel()));
+  }
+
   return true;
 }