Remove BaseChannel's dependency on TransportController.

webrtc/pc/channelmanager.cc

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 188 matching lines @@
   while (!video_channels_.empty()) {
     DestroyVideoChannel_w(video_channels_.back());
   }
   while (!voice_channels_.empty()) {
     DestroyVoiceChannel_w(voice_channels_.back());
   }
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
-      RTC_FROM_HERE, Bind(&ChannelManager::CreateVoiceChannel_w, this,
-                          media_controller, transport_controller, content_name,
-                          bundle_transport_name, rtcp, srtp_required, options));
+      RTC_FROM_HERE,
+      Bind(&ChannelManager::CreateVoiceChannel_w, this, media_controller,
+           rtp_transport, rtcp_transport, signaling_thread, content_name,
+           bundle_transport_name, rtcp, srtp_required, options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const AudioOptions& options) {
   ASSERT(initialized_);
   ASSERT(worker_thread_ == rtc::Thread::Current());
   ASSERT(nullptr != media_controller);
+
   VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (!media_channel)
     return nullptr;
 
   VoiceChannel* voice_channel = new VoiceChannel(
-      worker_thread_, network_thread_, media_engine_.get(), media_channel,
-      transport_controller, content_name, rtcp, srtp_required);
+      worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
+      media_channel, content_name, rtcp, srtp_required);
   voice_channel->SetCryptoOptions(crypto_options_);
-  if (!voice_channel->Init_w(bundle_transport_name)) {
+
+  if (!voice_channel->Init_w(rtp_transport, rtcp_transport)) {
     delete voice_channel;
     return nullptr;
   }
   voice_channels_.push_back(voice_channel);
   return voice_channel;
 }
 
 void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
   if (voice_channel) {
@@ skipping 12 matching lines @@
       voice_channels_.end(), voice_channel);
   ASSERT(it != voice_channels_.end());
   if (it == voice_channels_.end())
     return;
   voice_channels_.erase(it);
   delete voice_channel;
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
-      RTC_FROM_HERE, Bind(&ChannelManager::CreateVideoChannel_w, this,
-                          media_controller, transport_controller, content_name,
-                          bundle_transport_name, rtcp, srtp_required, options));
+      RTC_FROM_HERE,
+      Bind(&ChannelManager::CreateVideoChannel_w, this, media_controller,
+           rtp_transport, rtcp_transport, signaling_thread, content_name,
+           bundle_transport_name, rtcp, srtp_required, options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const VideoOptions& options) {
   ASSERT(initialized_);
   ASSERT(worker_thread_ == rtc::Thread::Current());
   ASSERT(nullptr != media_controller);
   VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (media_channel == NULL) {
     return NULL;
   }
 
   VideoChannel* video_channel =
-      new VideoChannel(worker_thread_, network_thread_, media_channel,
-                       transport_controller, content_name, rtcp, srtp_required);
+      new VideoChannel(worker_thread_, network_thread_, signaling_thread,
+                       media_channel, content_name, rtcp, srtp_required);
   video_channel->SetCryptoOptions(crypto_options_);
-  if (!video_channel->Init_w(bundle_transport_name)) {
+  if (!video_channel->Init_w(rtp_transport, rtcp_transport)) {
     delete video_channel;
     return NULL;
   }
   video_channels_.push_back(video_channel);
   return video_channel;
 }
 
 void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
   if (video_channel) {
@@ skipping 13 matching lines @@
   ASSERT(it != video_channels_.end());
   if (it == video_channels_.end())
     return;
 
   video_channels_.erase(it);
   delete video_channel;
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required) {
   return worker_thread_->Invoke<RtpDataChannel*>(
-      RTC_FROM_HERE, Bind(&ChannelManager::CreateRtpDataChannel_w, this,
-                          media_controller, transport_controller, content_name,
-                          bundle_transport_name, rtcp, srtp_required));
+      RTC_FROM_HERE,
+      Bind(&ChannelManager::CreateRtpDataChannel_w, this, media_controller,
+           rtp_transport, rtcp_transport, signaling_thread, content_name,
+           bundle_transport_name, rtcp, srtp_required));
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportController* transport_controller,
+    TransportChannel* rtp_transport,
+    TransportChannel* rtcp_transport,
+    rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required) {
   // This is ok to alloc from a thread other than the worker thread.
   ASSERT(initialized_);
   MediaConfig config;
   if (media_controller) {
     config = media_controller->config();
   }
   DataMediaChannel* media_channel = data_media_engine_->CreateChannel(config);
   if (!media_channel) {
     LOG(LS_WARNING) << "Failed to create RTP data channel.";
     return nullptr;
   }
 
-  RtpDataChannel* data_channel = new RtpDataChannel(
-      worker_thread_, network_thread_, media_channel, transport_controller,
-      content_name, rtcp, srtp_required);
+  RtpDataChannel* data_channel =
+      new RtpDataChannel(worker_thread_, network_thread_, signaling_thread,
+                         media_channel, content_name, rtcp, srtp_required);
   data_channel->SetCryptoOptions(crypto_options_);
-  if (!data_channel->Init_w(bundle_transport_name)) {
+  if (!data_channel->Init_w(rtp_transport, rtcp_transport)) {
     LOG(LS_WARNING) << "Failed to init data channel.";
     delete data_channel;
     return nullptr;
   }
   data_channels_.push_back(data_channel);
   return data_channel;
 }
 
 void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel");
@@ skipping 25 matching lines @@
                           media_engine_.get(), file, max_size_bytes));
 }
 
 void ChannelManager::StopAecDump() {
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
       Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
 }
 
 }  // namespace cricket