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

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 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 27 matching lines @@
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/videocapturer.h"
 #include "talk/session/media/channel.h"
 #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"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 using cricket::ContentInfo;
 using cricket::ContentInfos;
 using cricket::MediaContentDescription;
 using cricket::SessionDescription;
 using cricket::TransportInfo;
 
 using cricket::LOCAL_PORT_TYPE;
 using cricket::STUN_PORT_TYPE;
 using cricket::RELAY_PORT_TYPE;
 using cricket::PRFLX_PORT_TYPE;
 
+using rtc::Bind;
+
 namespace webrtc {
 
 // Error messages
 const char kBundleWithoutRtcpMux[] = "RTCP-MUX must be enabled when BUNDLE "
                                      "is enabled.";
 const char kCreateChannelFailed[] = "Failed to create channels.";
 const char kInvalidCandidates[] = "Description contains invalid candidates.";
 const char kInvalidSdp[] = "Invalid session description.";
 const char kMlineMismatch[] =
     "Offer and answer descriptions m-lines are not matching. Rejecting answer.";
@@ skipping 1681 matching lines @@
     const std::string content_name = data_channel_->content_name();
     channel_manager_->DestroyDataChannel(data_channel_.release());
   }
 }
 
 // TODO(mallinath) - Add a correct error code if the channels are not created
 // due to BUNDLE is enabled but rtcp-mux is disabled.
 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);
   if (data_channel_type_ != cricket::DCT_NONE &&
       data && !data->rejected && !data_channel_) {
     if (!CreateDataChannel(data)) {
       LOG(LS_ERROR) << "Failed to create data channel.";
       return false;
     }
   }
@@ skipping 17 matching lines @@
     if (!bundle_group) {
       LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
       return false;
     }
     if (!EnableBundle(*bundle_group)) {
       LOG(LS_WARNING) << "max-bundle failed to enable bundling.";
       return false;
     }
   }
 
+  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;
 }
 
 bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
   voice_channel_.reset(channel_manager_->CreateVoiceChannel(
       media_controller_.get(), transport_controller(), content->name, true,
       audio_options_));
   if (!voice_channel_) {
     return false;
   }
@@ skipping 352 matching lines @@
   if (!srtp_cipher.empty()) {
     metrics_observer_->IncrementSparseEnumCounter(
         srtp_counter_type, rtc::GetSrtpCryptoSuiteFromName(srtp_cipher));
   }
   if (ssl_cipher) {
     metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type, ssl_cipher);
   }
 }
 
 }  // namespace webrtc