Update webrtc&libjingle 6774:6825.

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>
 
@@ skipping 274 matching lines @@
     }
   }
 
   void AddStatistic(int idx, const std::string& name,
                     const std::string& value) {
     response_->addStatistic(idx,
                             blink::WebString::fromUTF8(name),
                             blink::WebString::fromUTF8(value));
   }
 
-  talk_base::scoped_refptr<LocalRTCStatsRequest> request_;
-  talk_base::scoped_refptr<LocalRTCStatsResponse> response_;
+  rtc::scoped_refptr<LocalRTCStatsRequest> request_;
+  rtc::scoped_refptr<LocalRTCStatsResponse> response_;
 };
 
 // Implementation of LocalRTCStatsRequest.
 LocalRTCStatsRequest::LocalRTCStatsRequest(blink::WebRTCStatsRequest impl)
     : impl_(impl),
       response_(NULL) {
 }
 
 LocalRTCStatsRequest::LocalRTCStatsRequest() {}
 LocalRTCStatsRequest::~LocalRTCStatsRequest() {}
 
 bool LocalRTCStatsRequest::hasSelector() const {
   return impl_.hasSelector();
 }
 
 blink::WebMediaStreamTrack LocalRTCStatsRequest::component() const {
   return impl_.component();
 }
 
 scoped_refptr<LocalRTCStatsResponse> LocalRTCStatsRequest::createResponse() {
   DCHECK(!response_);
-  response_ = new talk_base::RefCountedObject<LocalRTCStatsResponse>(
+  response_ = new rtc::RefCountedObject<LocalRTCStatsResponse>(
       impl_.createResponse());
   return response_.get();
 }
 
 void LocalRTCStatsRequest::requestSucceeded(
     const LocalRTCStatsResponse* response) {
   impl_.requestSucceeded(response->webKitStatsResponse());
 }
 
 // Implementation of LocalRTCStatsResponse.
@@ skipping 136 matching lines @@
           config, &constraints, frame_, this);
 
   if (!native_peer_connection_.get()) {
     LOG(ERROR) << "Failed to initialize native PeerConnection.";
     return false;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->RegisterPeerConnection(
         this, config, constraints, frame_);
 
-  uma_observer_ = new talk_base::RefCountedObject<PeerConnectionUMAObserver>();
+  uma_observer_ = new rtc::RefCountedObject<PeerConnectionUMAObserver>();
   native_peer_connection_->RegisterUMAObserver(uma_observer_.get());
   return true;
 }
 
 bool RTCPeerConnectionHandler::InitializeForTest(
     const blink::WebRTCConfiguration& server_configuration,
     const blink::WebMediaConstraints& options,
     PeerConnectionTracker* peer_connection_tracker) {
   webrtc::PeerConnectionInterface::RTCConfiguration config;
   GetNativeRtcConfiguration(server_configuration, &config);
 
   RTCMediaConstraints constraints(options);
   native_peer_connection_ =
       dependency_factory_->CreatePeerConnection(
           config, &constraints, NULL, this);
   if (!native_peer_connection_.get()) {
     LOG(ERROR) << "Failed to initialize native PeerConnection.";
     return false;
   }
   peer_connection_tracker_ = peer_connection_tracker;
   return true;
 }
 
 void RTCPeerConnectionHandler::createOffer(
     const blink::WebRTCSessionDescriptionRequest& request,
     const blink::WebMediaConstraints& options) {
   scoped_refptr<CreateSessionDescriptionRequest> description_request(
-      new talk_base::RefCountedObject<CreateSessionDescriptionRequest>(
+      new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
           request, this, PeerConnectionTracker::ACTION_CREATE_OFFER));
   RTCMediaConstraints constraints(options);
   native_peer_connection_->CreateOffer(description_request.get(), &constraints);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateOffer(this, constraints);
 }
 
 void RTCPeerConnectionHandler::createOffer(
     const blink::WebRTCSessionDescriptionRequest& request,
     const blink::WebRTCOfferOptions& options) {
   scoped_refptr<CreateSessionDescriptionRequest> description_request(
-      new talk_base::RefCountedObject<CreateSessionDescriptionRequest>(
+      new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
           request, this, PeerConnectionTracker::ACTION_CREATE_OFFER));
 
   RTCMediaConstraints constraints;
   ConvertOfferOptionsToConstraints(options, &constraints);
   native_peer_connection_->CreateOffer(description_request.get(), &constraints);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateOffer(this, constraints);
 }
 
 void RTCPeerConnectionHandler::createAnswer(
     const blink::WebRTCSessionDescriptionRequest& request,
     const blink::WebMediaConstraints& options) {
   scoped_refptr<CreateSessionDescriptionRequest> description_request(
-      new talk_base::RefCountedObject<CreateSessionDescriptionRequest>(
+      new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
           request, this, PeerConnectionTracker::ACTION_CREATE_ANSWER));
   RTCMediaConstraints constraints(options);
   native_peer_connection_->CreateAnswer(description_request.get(),
                                         &constraints);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateAnswer(this, constraints);
 }
 
 void RTCPeerConnectionHandler::setLocalDescription(
     const blink::WebRTCVoidRequest& request,
     const blink::WebRTCSessionDescription& description) {
   webrtc::SdpParseError error;
   webrtc::SessionDescriptionInterface* native_desc =
       CreateNativeSessionDescription(description, &error);
   if (!native_desc) {
     std::string reason_str = "Failed to parse SessionDescription. ";
     reason_str.append(error.line);
     reason_str.append(" ");
     reason_str.append(error.description);
     LOG(ERROR) << reason_str;
     request.requestFailed(blink::WebString::fromUTF8(reason_str));
     return;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackSetSessionDescription(
         this, description, PeerConnectionTracker::SOURCE_LOCAL);
 
   scoped_refptr<SetSessionDescriptionRequest> set_request(
-      new talk_base::RefCountedObject<SetSessionDescriptionRequest>(
+      new rtc::RefCountedObject<SetSessionDescriptionRequest>(
           request, this, PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION));
   native_peer_connection_->SetLocalDescription(set_request.get(), native_desc);
 }
 
 void RTCPeerConnectionHandler::setRemoteDescription(
     const blink::WebRTCVoidRequest& request,
     const blink::WebRTCSessionDescription& description) {
   webrtc::SdpParseError error;
   webrtc::SessionDescriptionInterface* native_desc =
       CreateNativeSessionDescription(description, &error);
   if (!native_desc) {
     std::string reason_str = "Failed to parse SessionDescription. ";
     reason_str.append(error.line);
     reason_str.append(" ");
     reason_str.append(error.description);
     LOG(ERROR) << reason_str;
     request.requestFailed(blink::WebString::fromUTF8(reason_str));
     return;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackSetSessionDescription(
         this, description, PeerConnectionTracker::SOURCE_REMOTE);
 
   scoped_refptr<SetSessionDescriptionRequest> set_request(
-      new talk_base::RefCountedObject<SetSessionDescriptionRequest>(
+      new rtc::RefCountedObject<SetSessionDescriptionRequest>(
           request, this, PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION));
   native_peer_connection_->SetRemoteDescription(set_request.get(), native_desc);
 }
 
 blink::WebRTCSessionDescription
 RTCPeerConnectionHandler::localDescription() {
   const webrtc::SessionDescriptionInterface* native_desc =
       native_peer_connection_->local_description();
   blink::WebRTCSessionDescription description =
       CreateWebKitSessionDescription(native_desc);
@@ skipping 124 matching lines @@
     peer_connection_tracker_->TrackRemoveStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
   PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
   track_metrics_.RemoveStream(MediaStreamTrackMetrics::SENT_STREAM,
                               webrtc_stream);
 }
 
 void RTCPeerConnectionHandler::getStats(
     const blink::WebRTCStatsRequest& request) {
   scoped_refptr<LocalRTCStatsRequest> inner_request(
-      new talk_base::RefCountedObject<LocalRTCStatsRequest>(request));
+      new rtc::RefCountedObject<LocalRTCStatsRequest>(request));
   getStats(inner_request.get());
 }
 
 void RTCPeerConnectionHandler::getStats(LocalRTCStatsRequest* request) {
-  talk_base::scoped_refptr<webrtc::StatsObserver> observer(
-      new talk_base::RefCountedObject<StatsResponse>(request));
+  rtc::scoped_refptr<webrtc::StatsObserver> observer(
+      new rtc::RefCountedObject<StatsResponse>(request));
   webrtc::MediaStreamTrackInterface* track = NULL;
   if (request->hasSelector()) {
     blink::WebMediaStreamSource::Type type =
         request->component().source().type();
     std::string track_id = request->component().id().utf8();
     if (type == blink::WebMediaStreamSource::TypeAudio) {
       track =
           native_peer_connection_->local_streams()->FindAudioTrack(track_id);
       if (!track) {
         track =
@@ skipping 43 matching lines @@
   // TODO(jiayl): remove the deprecated reliable field once Libjingle is updated
   // to handle that.
   config.reliable = false;
   config.id = init.id;
   config.ordered = init.ordered;
   config.negotiated = init.negotiated;
   config.maxRetransmits = init.maxRetransmits;
   config.maxRetransmitTime = init.maxRetransmitTime;
   config.protocol = base::UTF16ToUTF8(init.protocol);
 
-  talk_base::scoped_refptr<webrtc::DataChannelInterface> webrtc_channel(
+  rtc::scoped_refptr<webrtc::DataChannelInterface> webrtc_channel(
       native_peer_connection_->CreateDataChannel(base::UTF16ToUTF8(label),
                                                  &config));
   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);
 }
 
 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     const blink::WebMediaStreamTrack& track) {
   DVLOG(1) << "createDTMFSender.";
 
   MediaStreamTrack* native_track = MediaStreamTrack::GetTrack(track);
   if (!native_track ||
       track.source().type() != blink::WebMediaStreamSource::TypeAudio) {
     DLOG(ERROR) << "Could not create DTMF sender from a non-audio track.";
     return NULL;
   }
 
   webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter();
-  talk_base::scoped_refptr<webrtc::DtmfSenderInterface> sender(
+  rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
       native_peer_connection_->CreateDtmfSender(audio_track));
   if (!sender) {
     DLOG(ERROR) << "Could not create native DTMF sender.";
     return NULL;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateDTMFSender(this, track);
 
   return new RtcDtmfSenderHandler(sender);
 }
@@ skipping 158 matching lines @@
   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