Split libjingle's signaling thread from the UI thread

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

 // Copyright 2014 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/webrtc/webrtc_local_audio_track_adapter.h"
 
 #include "base/logging.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
+#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
+#include "content/renderer/render_thread_impl.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 namespace content {
 
 static const char kAudioTrackKind[] = "audio";
 
 scoped_refptr<WebRtcLocalAudioTrackAdapter>
 WebRtcLocalAudioTrackAdapter::Create(
     const std::string& label,
     webrtc::AudioSourceInterface* track_source) {
+  // TODO(tommi): Change this so that the signaling thread is one of the
+  // parameters to this method.
+  scoped_refptr<base::MessageLoopProxy> signaling_thread;
+  RenderThreadImpl* current = RenderThreadImpl::current();
+  if (current) {
+    PeerConnectionDependencyFactory* pc_factory =
+        current->GetPeerConnectionDependencyFactory();
+    signaling_thread = pc_factory->GetWebRtcSignalingThread();
+  }
+
+  LOG_IF(ERROR, !signaling_thread.get()) << "No signaling thread!";
+
   rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
       new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
-          label, track_source);
+          label, track_source, signaling_thread);
   return adapter;
 }
 
 WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
     const std::string& label,
-    webrtc::AudioSourceInterface* track_source)
+    webrtc::AudioSourceInterface* track_source,
+    const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread)
     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
       owner_(NULL),
       track_source_(track_source),
+      signaling_thread_(signaling_thread),
       signal_level_(0) {
-  signaling_thread_.DetachFromThread();
+  signaling_thread_checker_.DetachFromThread();
   capture_thread_.DetachFromThread();
 }
 
 WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
 }
 
 void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
   DCHECK(!owner_);
   DCHECK(owner);
   owner_ = owner;
 }
 
 void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
     const scoped_refptr<MediaStreamAudioProcessor>& processor) {
   // SetAudioProcessor will be called when a new capture thread has been
   // initialized, so we need to detach from any current capture thread we're
   // checking and attach to the current one.
   capture_thread_.DetachFromThread();
   DCHECK(capture_thread_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   audio_processor_ = processor;
 }
 
 std::string WebRtcLocalAudioTrackAdapter::kind() const {
   return kAudioTrackKind;
 }
 
+bool WebRtcLocalAudioTrackAdapter::set_enabled(bool enable) {
+  // If we're not called on the signaling thread, we need to post a task to
+  // change the state on the correct thread.
+  if (signaling_thread_.get() && !signaling_thread_->BelongsToCurrentThread()) {
+    signaling_thread_->PostTask(FROM_HERE,
+        base::Bind(
+            base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
+            this, enable));
+    return true;
+  }
+
+  return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
+      set_enabled(enable);
+}
+
 void WebRtcLocalAudioTrackAdapter::AddSink(
     webrtc::AudioTrackSinkInterface* sink) {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   DCHECK(sink);
 #ifndef NDEBUG
   // Verify that |sink| has not been added.
   for (ScopedVector<WebRtcAudioSinkAdapter>::const_iterator it =
            sink_adapters_.begin();
        it != sink_adapters_.end(); ++it) {
     DCHECK(!(*it)->IsEqual(sink));
   }
 #endif
 
   scoped_ptr<WebRtcAudioSinkAdapter> adapter(
       new WebRtcAudioSinkAdapter(sink));
   owner_->AddSink(adapter.get());
   sink_adapters_.push_back(adapter.release());
 }
 
 void WebRtcLocalAudioTrackAdapter::RemoveSink(
     webrtc::AudioTrackSinkInterface* sink) {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   DCHECK(sink);
   for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
            sink_adapters_.begin();
        it != sink_adapters_.end(); ++it) {
     if ((*it)->IsEqual(sink)) {
       owner_->RemoveSink(*it);
       sink_adapters_.erase(it);
       return;
     }
   }
 }
 
 bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
 
   // It is required to provide the signal level after audio processing. In
   // case the audio processing is not enabled for the track, we return
   // false here in order not to overwrite the value from WebRTC.
   // TODO(xians): Remove this after we turn on the APM in Chrome by default.
   // http://crbug/365672 .
   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
     return false;
 
   base::AutoLock auto_lock(lock_);
   *level = signal_level_;
   return true;
 }
 
 rtc::scoped_refptr<webrtc::AudioProcessorInterface>
 WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   return audio_processor_.get();
 }
 
 std::vector<int> WebRtcLocalAudioTrackAdapter::VoeChannels() const {
   DCHECK(capture_thread_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   return voe_channels_;
 }
 
 void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int signal_level) {
   DCHECK(capture_thread_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   signal_level_ = signal_level;
 }
 
 void WebRtcLocalAudioTrackAdapter::AddChannel(int channel_id) {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::AddChannel(channel_id="
            << channel_id << ")";
   base::AutoLock auto_lock(lock_);
   if (std::find(voe_channels_.begin(), voe_channels_.end(), channel_id) !=
       voe_channels_.end()) {
     // We need to handle the case when the same channel is connected to the
     // track more than once.
     return;
   }
 
   voe_channels_.push_back(channel_id);
 }
 
 void WebRtcLocalAudioTrackAdapter::RemoveChannel(int channel_id) {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::RemoveChannel(channel_id="
            << channel_id << ")";
   base::AutoLock auto_lock(lock_);
   std::vector<int>::iterator iter =
       std::find(voe_channels_.begin(), voe_channels_.end(), channel_id);
   DCHECK(iter != voe_channels_.end());
   voe_channels_.erase(iter);
 }
 
 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   return track_source_;
 }
 
 cricket::AudioRenderer* WebRtcLocalAudioTrackAdapter::GetRenderer() {
   // When the audio track processing is enabled, return a NULL so that capture
   // data goes through Libjingle LocalAudioTrackHandler::LocalAudioSinkAdapter
   // ==> WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer ==> WebRTC.
   // When the audio track processing is disabled, WebRtcLocalAudioTrackAdapter
   // is used to pass the channel ids to WebRtcAudioDeviceImpl, the data flow
   // becomes WebRtcAudioDeviceImpl ==> WebRTC.
   // TODO(xians): Only return NULL after the APM in WebRTC is deprecated.
   // See See http://crbug/365672 for details.
   return MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()?
       NULL : this;
 }
 
 }  // namespace content