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

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
index 72b80194b08ed09a01673c98c4ed9816aa4e6d74..899e1abf5584ae6ad1bd6d4ce7b3cb976671a28f 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
@@ -8,7 +8,7 @@
 #include <vector>
 
 #include "base/memory/ref_counted.h"
-#include "base/memory/scoped_vector.h"
+#include "base/memory/weak_ptr.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "content/common/content_export.h"
@@ -17,8 +17,8 @@
 #include "third_party/webrtc/api/mediastreamtrack.h"
 #include "third_party/webrtc/media/base/audiorenderer.h"
 
-namespace cricket {
-class AudioRenderer;
+namespace base {
+class WaitableEvent;
 }
 
 namespace webrtc {
@@ -29,8 +29,8 @@ class AudioProcessorInterface;
 namespace content {
 
 class MediaStreamAudioProcessor;
+class MediaStreamAudioTrack;
 class WebRtcAudioSinkAdapter;
-class WebRtcLocalAudioTrack;
 
 // Provides an implementation of the webrtc::AudioTrackInterface that can be
 // bound/unbound to/from a MediaStreamAudioTrack.  In other words, this is an
@@ -51,7 +51,10 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
 
   ~WebRtcLocalAudioTrackAdapter() override;
 
-  void Initialize(WebRtcLocalAudioTrack* owner);
+  // Set the |track| that manages the MediaStreamAudioSinks.  The WeakPtr will
+  // only be dereferenced on the main thread.  This method must only be called
+  // on the main thread.
+  void SetMediaStreamAudioTrack(base::WeakPtr<MediaStreamAudioTrack> track);
 
   // Set the object that provides shared access to the current audio signal
   // level.  This method may only be called once, before the audio data flow
@@ -79,23 +82,38 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
       override;
   webrtc::AudioSourceInterface* GetSource() const override;
 
-  // Weak reference.
-  WebRtcLocalAudioTrack* owner_;
+  // Removes the |sink| from |track_| and then signals the |done_event| (if
+  // provided).  This is used by RemoveSink() to ensure the audio flow has
+  // halted before it returns.
+  void RemoveSinkOnMainThread(webrtc::AudioTrackSinkInterface* sink,
+                              base::WaitableEvent* done_event);
 
   // The source of the audio track which handles the audio constraints.
-  // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
-  rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
+  const rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
-  // Libjingle's signaling thread.
+  // Task runner for operations that must be done on libjingle's signaling
+  // thread.  May be null for single-threaded unit tests.
   const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
 
+  // Task runner for operations that must be done on the main thread.  May be
+  // null for single-threaded unit tests.

[o1ka, 2016/04/06 18:39:10]

Are you planing to have some multi-threaded unit tests as well? (for example,for
checking done_event trick is not broken)

[miu, 2016/04/19 00:40:23]

Possibly.  But, with Patch Set 4 I was able to get rid of the "done_event trick"
altogether!  :)

To answer your question more broadly: I haven't modified the tests yet because
we are still working out major code structure.  Once it feels like the code
structure has stabilized, I'll modify existing tests and add whatever extra
tests are needed.

+  scoped_refptr<base::SingleThreadTaskRunner> main_task_runner_;
+
+  // The track to add/remove sinks to/from.  When the
+  // webrtc::AudioTrackInterface::Add/RemoveSink() methods are called, they
+  // create a proxy (WebRtcAudioSinkAdapter) that implements the
+  // MediaStreamAudioSink interface to call into the
+  // webrtc::AudioTrackSinkInterface.  This must only be dereferenced on the
+  // main thread.
+  base::WeakPtr<MediaStreamAudioTrack> track_;
+
   // The audio processsor that applies audio processing on the data of audio
   // track.  This must be set before calls to GetAudioProcessor() are made.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   // A vector of the peer connection sink adapters which receive the audio data
   // from the audio track.
-  ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
+  std::vector<scoped_ptr<WebRtcAudioSinkAdapter>> sink_adapters_;
 
   // Thread-safe accessor to current audio signal level.  This must be set
   // before calls to GetSignalLevel() are made.