Adding ability for BaseChannel to use PacketTransportInterface.

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 228 matching lines @@
   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_, signaling_thread, media_engine_.get(),
       media_channel, content_name, rtcp_mux_required, srtp_required);
   voice_channel->SetCryptoOptions(crypto_options_);
 
-  if (!voice_channel->Init_w(rtp_transport, rtcp_transport)) {
+  if (!voice_channel->Init_w(rtp_transport, rtcp_transport, 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 50 matching lines @@
   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_, signaling_thread, media_channel,
       content_name, rtcp_mux_required, srtp_required);
   video_channel->SetCryptoOptions(crypto_options_);
-  if (!video_channel->Init_w(rtp_transport, rtcp_transport)) {
+  if (!video_channel->Init_w(rtp_transport, rtcp_transport, 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 52 matching lines @@
   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_, signaling_thread, media_channel,
       content_name, rtcp_mux_required, srtp_required);
   data_channel->SetCryptoOptions(crypto_options_);
-  if (!data_channel->Init_w(rtp_transport, rtcp_transport)) {
+  if (!data_channel->Init_w(rtp_transport, rtcp_transport, 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