Remove the dependency of TransportChannel and TransportChannelImpl.

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 189 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,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required,
     const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateVoiceChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
            bundle_transport_name, rtcp_mux_required, srtp_required, options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required,
     const AudioOptions& options) {
   RTC_DCHECK(initialized_);
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   RTC_DCHECK(nullptr != media_controller);
 
@@ skipping 33 matching lines @@
       voice_channels_.end(), voice_channel);
   RTC_DCHECK(it != voice_channels_.end());
   if (it == voice_channels_.end())
     return;
   voice_channels_.erase(it);
   delete voice_channel;
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateVideoChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
            bundle_transport_name, rtcp_mux_required, srtp_required, options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required,
     const VideoOptions& options) {
   RTC_DCHECK(initialized_);
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   RTC_DCHECK(nullptr != media_controller);
   VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
@@ skipping 33 matching lines @@
   RTC_DCHECK(it != video_channels_.end());
   if (it == video_channels_.end())
     return;
 
   video_channels_.erase(it);
   delete video_channel;
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel(
     webrtc::MediaControllerInterface* media_controller,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required) {
   return worker_thread_->Invoke<RtpDataChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateRtpDataChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
            bundle_transport_name, rtcp_mux_required, srtp_required));
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
     webrtc::MediaControllerInterface* media_controller,
-    TransportChannel* rtp_transport,
-    TransportChannel* rtcp_transport,
+    DtlsTransportInternal* rtp_transport,
+    DtlsTransportInternal* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp_mux_required,
     bool srtp_required) {
   // This is ok to alloc from a thread other than the worker thread.
   RTC_DCHECK(initialized_);
   MediaConfig config;
   if (media_controller) {
     config = media_controller->config();
@@ skipping 47 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