Split libjingle's signaling thread from the UI thread

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
index ead38c2da12fc785f410036b45c64bc79e869458..1339f32fffe82335558681b66e9a2f25da4567d6 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -6,8 +6,10 @@
 
 #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 {
@@ -18,20 +20,32 @@ scoped_refptr<WebRtcLocalAudioTrackAdapter>
 WebRtcLocalAudioTrackAdapter::Create(
     const std::string& label,
     webrtc::AudioSourceInterface* track_source) {
+  scoped_refptr<base::MessageLoopProxy> signaling_thread;
+  RenderThreadImpl* current = RenderThreadImpl::current();
+  if (current) {
+    PeerConnectionDependencyFactory* pc_factory =
+        current->GetPeerConnectionDependencyFactory();
+    signaling_thread = pc_factory->GetWebRtcSignalingThread();

[no longer working on chromium, 2014/10/30 11:55:14]

WebRtcLocalAudioTrackAdapter is created by PeerConnectionDependencyFactory
today, can you move the code to PeerConnectionDependencyFactory and inject the
signal thread to the adapter?

[tommi (sloooow) - chröme, 2014/10/30 20:37:36]

Yes, I plan to do that and added a TODO to do that.  There currently are so many
unrelated call sites to this method though that I wanted to do that in a
separate cl.

See:
https://code.google.com/p/chromium/codesearch#search/&q=WebRtcLocalAudioTrack...

+  }
+
+  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();
 }
 
@@ -59,9 +73,25 @@ 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.
+  bool ret = true;
+  if (signaling_thread_.get() && !signaling_thread_->BelongsToCurrentThread()) {
+    signaling_thread_->PostTask(FROM_HERE,
+        base::Bind(
+            base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
+            this, enable));
+  } else {
+    ret = webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
+        set_enabled(enable);
+  }
+  return ret;

[no longer working on chromium, 2014/10/30 11:55:14]

how about:
if (signaling_thread_.get() && !signaling_thread_->BelongsToCurrentThread()) {
    signaling_thread_->PostTask(FROM_HERE, ..
    return true;
}

return
webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::set_enabled(enable);

[tommi (sloooow) - chröme, 2014/10/30 20:37:36]

Done.

+}
+
 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.
@@ -80,7 +110,7 @@ void WebRtcLocalAudioTrackAdapter::AddSink(
 
 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();
@@ -94,7 +124,7 @@ void WebRtcLocalAudioTrackAdapter::RemoveSink(
 }
 
 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
@@ -111,7 +141,7 @@ bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
 
 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();
 }
@@ -129,7 +159,7 @@ void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int 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_);
@@ -144,7 +174,7 @@ void WebRtcLocalAudioTrackAdapter::AddChannel(int 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_);
@@ -155,7 +185,7 @@ void WebRtcLocalAudioTrackAdapter::RemoveChannel(int channel_id) {
 }
 
 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
-  DCHECK(signaling_thread_.CalledOnValidThread());
+  DCHECK(signaling_thread_checker_.CalledOnValidThread());
   return track_source_;
 }