Delete MediaController class, move Call ownership to PeerConnection.

webrtc/ortc/rtptransportcontrolleradapter.cc

Index: webrtc/ortc/rtptransportcontrolleradapter.cc
diff --git a/webrtc/ortc/rtptransportcontrolleradapter.cc b/webrtc/ortc/rtptransportcontrolleradapter.cc
index ad5d8b5d68f813f890e9f99e1d9b3f001f6583b2..b482e475fdd508f58da4753888cf6140bb9e580b 100644
--- a/webrtc/ortc/rtptransportcontrolleradapter.cc
+++ b/webrtc/ortc/rtptransportcontrolleradapter.cc
@@ -121,6 +121,10 @@ RtpTransportControllerAdapter::~RtpTransportControllerAdapter() {
     // been destroyed. This isn't safe (see error log above).
     DestroyVideoChannel();
   }
+  // Call must be destroyed on the worker thread.
+  worker_thread_->Invoke<void>(
+      RTC_FROM_HERE,
+      rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
 }
 
 RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
@@ -584,14 +588,11 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
     rtc::Thread* worker_thread)
     : signaling_thread_(signaling_thread),
       worker_thread_(worker_thread),
-      media_controller_(MediaControllerInterface::Create(config,
-                                                         worker_thread,
-                                                         channel_manager,
-                                                         event_log)) {
+      media_config_(config),
+      channel_manager_(channel_manager),
+      event_log_(event_log) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
-  RTC_DCHECK(channel_manager);
-  // MediaControllerInterface::Create should never fail.
-  RTC_DCHECK(media_controller_);
+  RTC_DCHECK(channel_manager_);
   // Add "dummy" codecs to the descriptions, because the media engines
   // currently reject empty lists of codecs. Note that these codecs will never
   // actually be used, because when parameters are set, the dummy codecs will
@@ -603,6 +604,33 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
   remote_audio_description_.AddCodec(dummy_audio);
   local_video_description_.AddCodec(dummy_video);
   remote_video_description_.AddCodec(dummy_video);
+
+  worker_thread_->Invoke<void>(
+      RTC_FROM_HERE,
+      rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
+}
+
+// TODO(nisse): Duplicates corresponding method in PeerConnection (used
+// to be in MediaController).

[stefan-webrtc, 2017/05/05 08:47:05]

I think this TODO should say when these duplicates can be removed, and perhaps
why they're needed?

[stefan-webrtc, 2017/05/05 08:53:50]

Talked offline, and duplicates are needed since this is an ORTC object and isn't
used by PC. 

[nisse-webrtc, 2017/05/05 09:02:37]

I'm afraid I'm not at all familiar with the new ortc-related code. Taylor, do
you see any good way to reduce duplication?

One possibility might be make these bandwidth values the default values in the
definition of the config struct? (For a followup, I'll now try to get this cl
landed).

[Taylor Brandstetter, 2017/05/05 16:32:01]

I wouldn't worry about reducing this duplication. Eventually this whole class
will be deleted and it will naturally go away; see the comment in the header
file. Speaking of which, that comment should be updated to remove "wraps a
MediaController".

+void RtpTransportControllerAdapter::Init_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(!call_);
+
+  const int kMinBandwidthBps = 30000;
+  const int kStartBandwidthBps = 300000;
+  const int kMaxBandwidthBps = 2000000;
+
+  webrtc::Call::Config call_config(event_log_);
+  call_config.audio_state = channel_manager_->media_engine()->GetAudioState();
+  call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
+  call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
+  call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
+
+  call_.reset(webrtc::Call::Create(call_config));
+}
+
+void RtpTransportControllerAdapter::Close_w() {
+  call_.reset();
 }
 
 RTCError RtpTransportControllerAdapter::AttachAudioSender(
@@ -818,8 +846,8 @@ void RtpTransportControllerAdapter::OnVideoReceiverDestroyed() {
 }
 
 void RtpTransportControllerAdapter::CreateVoiceChannel() {
-  voice_channel_ = media_controller_->channel_manager()->CreateVoiceChannel(
-      media_controller_.get(),
+  voice_channel_ = channel_manager_->CreateVoiceChannel(
+      call_.get(), media_config_,
       inner_audio_transport_->GetRtpPacketTransport()->GetInternal(),
       inner_audio_transport_->GetRtcpPacketTransport()
           ? inner_audio_transport_->GetRtcpPacketTransport()->GetInternal()
@@ -830,8 +858,8 @@ void RtpTransportControllerAdapter::CreateVoiceChannel() {
 }
 
 void RtpTransportControllerAdapter::CreateVideoChannel() {
-  video_channel_ = media_controller_->channel_manager()->CreateVideoChannel(
-      media_controller_.get(),
+  video_channel_ = channel_manager_->CreateVideoChannel(
+      call_.get(), media_config_,
       inner_video_transport_->GetRtpPacketTransport()->GetInternal(),
       inner_video_transport_->GetRtcpPacketTransport()
           ? inner_video_transport_->GetRtcpPacketTransport()->GetInternal()
@@ -843,14 +871,14 @@ void RtpTransportControllerAdapter::CreateVideoChannel() {
 
 void RtpTransportControllerAdapter::DestroyVoiceChannel() {
   RTC_DCHECK(voice_channel_);
-  media_controller_->channel_manager()->DestroyVoiceChannel(voice_channel_);
+  channel_manager_->DestroyVoiceChannel(voice_channel_);
   voice_channel_ = nullptr;
   inner_audio_transport_ = nullptr;
 }
 
 void RtpTransportControllerAdapter::DestroyVideoChannel() {
   RTC_DCHECK(video_channel_);
-  media_controller_->channel_manager()->DestroyVideoChannel(video_channel_);
+  channel_manager_->DestroyVideoChannel(video_channel_);
   video_channel_ = nullptr;
   inner_video_transport_ = nullptr;
 }