Delete MediaController class, move Call ownership to PeerConnection.

webrtc/pc/webrtcsession.cc

Index: webrtc/pc/webrtcsession.cc
diff --git a/webrtc/pc/webrtcsession.cc b/webrtc/pc/webrtcsession.cc
index c6046e905fd48e3a57b98315d49bebe091b56dfd..54ef1a923e7fb20cb37ffc9516238c3773d5da53 100644
--- a/webrtc/pc/webrtcsession.cc
+++ b/webrtc/pc/webrtcsession.cc
@@ -461,7 +461,9 @@ bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
 }
 
 WebRtcSession::WebRtcSession(
-    webrtc::MediaControllerInterface* media_controller,
+    cricket::ChannelManager* channel_manager,
+    const cricket::MediaConfig& media_config,
+    RtcEventLog* event_log,
     rtc::Thread* network_thread,
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
@@ -477,8 +479,9 @@ WebRtcSession::WebRtcSession(
       sid_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
       transport_controller_(std::move(transport_controller)),
       sctp_factory_(std::move(sctp_factory)),
-      media_controller_(media_controller),
-      channel_manager_(media_controller_->channel_manager()),
+      media_config_(media_config),
+      event_log_(event_log),
+      channel_manager_(channel_manager),
       ice_observer_(NULL),
       ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
       ice_connection_receiving_(true),
@@ -525,6 +528,11 @@ WebRtcSession::~WebRtcSession() {
   }
 #endif
 
+  // Make sure Call is destructed on the worker thread.
+  if (state() != STATE_CLOSED) {
+    worker_thread_->Invoke<void>(RTC_FROM_HERE,
+                                 rtc::Bind(&WebRtcSession::Close_w, this));
+  }
   LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
 }
 
@@ -620,9 +628,29 @@ bool WebRtcSession::Initialize(
     webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
   }
 
+  worker_thread_->Invoke<void>(RTC_FROM_HERE,
+                               rtc::Bind(&WebRtcSession::Init_w, this));
+
   return true;
 }
 
+void WebRtcSession::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 WebRtcSession::Close() {
   SetState(STATE_CLOSED);
   RemoveUnusedChannels(nullptr);
@@ -630,7 +658,14 @@ void WebRtcSession::Close() {
   RTC_DCHECK(!video_channel_);
   RTC_DCHECK(!rtp_data_channel_);
   RTC_DCHECK(!sctp_transport_);
-  media_controller_->Close();
+
+  worker_thread_->Invoke<void>(RTC_FROM_HERE,
+                               rtc::Bind(&WebRtcSession::Close_w, this));
+}
+
+void WebRtcSession::Close_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  call_.reset();
 }
 
 cricket::BaseChannel* WebRtcSession::GetChannel(
@@ -1757,7 +1792,7 @@ bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
   }
 
   voice_channel_.reset(channel_manager_->CreateVoiceChannel(
-      media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
+      call(), media_config_, rtp_dtls_transport, rtcp_dtls_transport,
       transport_controller_->signaling_thread(), content->name, SrtpRequired(),
       audio_options_));
   if (!voice_channel_) {
@@ -1799,7 +1834,7 @@ bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
   }
 
   video_channel_.reset(channel_manager_->CreateVideoChannel(
-      media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
+      call(), media_config_, rtp_dtls_transport, rtcp_dtls_transport,
       transport_controller_->signaling_thread(), content->name, SrtpRequired(),
       video_options_));
 
@@ -1864,7 +1899,7 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
     }
 
     rtp_data_channel_.reset(channel_manager_->CreateRtpDataChannel(
-        media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
+        media_config_, rtp_dtls_transport, rtcp_dtls_transport,
         transport_controller_->signaling_thread(), content->name,
         SrtpRequired()));
 
@@ -2313,7 +2348,7 @@ void WebRtcSession::ReportNegotiatedCiphers(
 
 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
-  media_controller_->call_w()->OnSentPacket(sent_packet);
+  call()->OnSentPacket(sent_packet);
 }
 
 const std::string WebRtcSession::GetTransportName(