MediaStream audio: Refactor 3 separate "glue" implementations into one.

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 f169880cd17f393551d016c861bcda96b2aca0af..e11760715c4c17616665b7603c5815da83ddea3b 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -4,12 +4,17 @@
 
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 
+#include <limits>
+
+#include "base/bind.h"
 #include "base/location.h"
 #include "base/logging.h"
+#include "base/message_loop/message_loop.h"
+#include "base/synchronization/waitable_event.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
+#include "content/renderer/media/media_stream_audio_track.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/webrtc/api/mediastreaminterface.h"
 
@@ -45,17 +50,19 @@ WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
     webrtc::AudioSourceInterface* track_source,
     scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
-      owner_(NULL),
       track_source_(track_source),
       signaling_task_runner_(std::move(signaling_task_runner)) {}
 
 WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
 }
 
-void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
-  DCHECK(!owner_);
-  DCHECK(owner);
-  owner_ = owner;
+void WebRtcLocalAudioTrackAdapter::SetMediaStreamAudioTrack(
+    base::WeakPtr<MediaStreamAudioTrack> track) {
+  // Note: Single-threaded unit tests might not provide a task runner (or even a
+  // main-thread MessageLoop).
+  if (base::MessageLoop* main_loop = base::MessageLoop::current())
+    main_task_runner_ = main_loop->task_runner();
+  track_ = track;
 }
 
 void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
@@ -94,38 +101,65 @@ bool WebRtcLocalAudioTrackAdapter::set_enabled(bool enable) {
 
 void WebRtcLocalAudioTrackAdapter::AddSink(
     webrtc::AudioTrackSinkInterface* sink) {
-  DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
   DCHECK(sink);
+
+  // The MediaStream object graph may only be modified on the main thread, so
+  // trampoline if necessary.
+  if (main_task_runner_ && !main_task_runner_->RunsTasksOnCurrentThread()) {
+    main_task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(&WebRtcLocalAudioTrackAdapter::AddSink, this, sink));
+    return;
+  }
+
 #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));
-  }
+  for (const auto& adapter : sink_adapters_)
+    DCHECK(!adapter->IsEqual(sink));
 #endif
 
-  scoped_ptr<WebRtcAudioSinkAdapter> adapter(
-      new WebRtcAudioSinkAdapter(sink));
-  owner_->AddSink(adapter.get());
-  sink_adapters_.push_back(adapter.release());
+  if (track_) {
+    scoped_ptr<WebRtcAudioSinkAdapter> adapter(
+        new WebRtcAudioSinkAdapter(sink));
+    track_->AddSink(adapter.get());
+    sink_adapters_.push_back(std::move(adapter));
+  }
 }
 
 void WebRtcLocalAudioTrackAdapter::RemoveSink(
     webrtc::AudioTrackSinkInterface* sink) {
-  DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
   DCHECK(sink);
-  for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
-           sink_adapters_.begin();
-       it != sink_adapters_.end(); ++it) {
+
+  // The MediaStream object graph may only be modified on the main thread, so
+  // trampoline if necessary.  Furthermore, block the current thread until the
+  // task has completed because the interface contract requires the audio flow
+  // to |sink| be stopped when this method returns.
+  if (main_task_runner_ && !main_task_runner_->RunsTasksOnCurrentThread()) {
+    base::WaitableEvent done_event(false, false);
+    main_task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(&WebRtcLocalAudioTrackAdapter::RemoveSinkOnMainThread, this,
+                   sink, &done_event));
+    done_event.Wait();
+  } else {
+    RemoveSinkOnMainThread(sink, nullptr);
+  }
+}
+
+void WebRtcLocalAudioTrackAdapter::RemoveSinkOnMainThread(
+    webrtc::AudioTrackSinkInterface* sink,
+    base::WaitableEvent* done_event) {
+  DCHECK(!main_task_runner_ || main_task_runner_->RunsTasksOnCurrentThread());
+  for (auto it = sink_adapters_.begin(); it != sink_adapters_.end(); ++it) {
     if ((*it)->IsEqual(sink)) {
-      owner_->RemoveSink(*it);
+      if (track_)
+        track_->RemoveSink(it->get());
       sink_adapters_.erase(it);
-      return;
+      break;
     }
   }
+  if (done_event)
+    done_event->Signal();
 }
 
 bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {