Refactor RtcDataChannelHandler.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/lazy_instance.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/metrics/histogram.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
+#include "base/thread_task_runner_handle.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
@@ skipping 793 matching lines @@
   if (!webrtc_channel) {
     DLOG(ERROR) << "Could not create native data channel.";
     return NULL;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateDataChannel(
         this, webrtc_channel.get(), PeerConnectionTracker::SOURCE_LOCAL);
 
   ++num_data_channels_created_;
 
-  return new RtcDataChannelHandler(webrtc_channel);
+  return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(),
+                                   webrtc_channel);
 }
 
 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     const blink::WebMediaStreamTrack& track) {
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
   DVLOG(1) << "createDTMFSender.";
 
   MediaStreamTrack* native_track = MediaStreamTrack::GetTrack(track);
   if (!native_track ||
       track.source().type() != blink::WebMediaStreamSource::TypeAudio) {
@@ skipping 176 matching lines @@
 
 void RTCPeerConnectionHandler::OnDataChannel(
     webrtc::DataChannelInterface* data_channel) {
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnDataChannel");
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateDataChannel(
         this, data_channel, PeerConnectionTracker::SOURCE_REMOTE);
 
   DVLOG(1) << "RTCPeerConnectionHandler::OnDataChannel "
            << data_channel->label();
-  client_->didAddRemoteDataChannel(new RtcDataChannelHandler(data_channel));
+  client_->didAddRemoteDataChannel(new RtcDataChannelHandler(
+      base::ThreadTaskRunnerHandle::Get(), data_channel));
 }
 
 void RTCPeerConnectionHandler::OnRenegotiationNeeded() {
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRenegotiationNeeded");
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackOnRenegotiationNeeded(this);
   client_->negotiationNeeded();
 }
 
 PeerConnectionTracker* RTCPeerConnectionHandler::peer_connection_tracker() {
   return peer_connection_tracker_;
 }
 
 webrtc::SessionDescriptionInterface*
 RTCPeerConnectionHandler::CreateNativeSessionDescription(
     const blink::WebRTCSessionDescription& description,
     webrtc::SdpParseError* error) {
   std::string sdp = base::UTF16ToUTF8(description.sdp());
   std::string type = base::UTF16ToUTF8(description.type());
   webrtc::SessionDescriptionInterface* native_desc =
       dependency_factory_->CreateSessionDescription(type, sdp, error);
 
   LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
                               << " Type: " << type << " SDP: " << sdp;
 
   return native_desc;
 }
 
 }  // namespace content