MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 
 #include <vector>
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_vector.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "third_party/webrtc/api/mediastreamtrack.h"
-#include "third_party/webrtc/media/base/audiorenderer.h"
+#include "third_party/webrtc/base/refcount.h"
 
-namespace cricket {
-class AudioRenderer;
+namespace base {
+class WaitableEvent;
 }
 
 namespace webrtc {
 class AudioSourceInterface;
 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
+// adapter that sits between the media stream object graph and WebRtc's object
+// graph and proxies between the two.
+//
+// TODO(miu): Rename to WebRtcAudioTrackAdapter since this can be used to proxy
+// between any kind (local or remote) of MediaStreamAudioTrack.
 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
     : NON_EXPORTED_BASE(
           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  public:
   static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
       const std::string& label,
-      webrtc::AudioSourceInterface* track_source);
+      webrtc::AudioSourceInterface* track_source,
+      const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner);
 
-  WebRtcLocalAudioTrackAdapter(
-      const std::string& label,
-      webrtc::AudioSourceInterface* track_source,
-      const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread);
+  // Set the |track| that manages the MediaStreamAudioSinks.  It is the client's
+  // responsibility to call this method with null before the track's destruction
+  // time.  This is needed because WebRtcLocalAudioTrackAdapter is ref-counted
+  // and could potentially out-live |track|.
+  //
+  // This method must only be called on the main thread.
+  void SetMediaStreamAudioTrack(MediaStreamAudioTrack* track);
 
-  ~WebRtcLocalAudioTrackAdapter() override;
-
-  void Initialize(WebRtcLocalAudioTrack* owner);
-
-  // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
-  // level of the audio data.
-  void SetSignalLevel(int signal_level);
-
-  // Method called by the WebRtcLocalAudioTrack to set the processor that
-  // applies signal processing on the data of the track.
+  // Set the processor that applies signal processing on the data of the track.
   // This class will keep a reference of the |processor|.
   // Called on the main render thread.
   void SetAudioProcessor(
       const scoped_refptr<MediaStreamAudioProcessor>& processor);
 
+  // Set the object that provides shared access to the current audio signal
+  // level.
+  void SetReportedLevel(
+      const scoped_refptr<MediaStreamAudioLevelCalculator::ReportedLevel>&
+          reported_level);
+
   // webrtc::MediaStreamTrack implementation.
-  std::string kind() const override;
-  bool set_enabled(bool enable) override;
+  std::string kind() const final;
+  bool set_enabled(bool enable) final;
+
+ protected:
+  friend class rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>;
+
+  WebRtcLocalAudioTrackAdapter(
+      const std::string& label,
+      webrtc::AudioSourceInterface* track_source,
+      const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner);
+
+  ~WebRtcLocalAudioTrackAdapter() override;
 
  private:
+  // 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 (on the signaling thread).
+  void RemoveSinkOnMainThread(webrtc::AudioTrackSinkInterface* sink,
+                              base::WaitableEvent* done_event);
+
   // webrtc::AudioTrackInterface implementation.
-  void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
-  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
-  bool GetSignalLevel(int* level) override;
-  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
-      override;
-  webrtc::AudioSourceInterface* GetSource() const override;
-
-  // Weak reference.
-  WebRtcLocalAudioTrack* owner_;
+  void AddSink(webrtc::AudioTrackSinkInterface* sink) final;
+  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) final;
+  bool GetSignalLevel(int* level) final;
+  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() final;
+  webrtc::AudioSourceInterface* GetSource() const final;
 
   // 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.
-  const scoped_refptr<base::SingleThreadTaskRunner> 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.
+  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 that implements the MediaStreamAudioSink interface to call
+  // into the webrtc::AudioTrackSinkInterface.  This must only be accessed on
+  // the main thread.
+  MediaStreamAudioTrack* track_;
+
+  // A vector of the peer connection sink adapters which receive the audio data
+  // from the audio track.  This must only be accessed on the main thread.
+  ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
+
+  // Protects |audio_processor_|, |voe_channels_|, and |signal_level_|.
+  mutable base::Lock lock_;
 
   // The audio processsor that applies audio processing on the data of audio
   // track.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
+  // Thread-safe accessor to current audio signal level.
+  scoped_refptr<MediaStreamAudioLevelCalculator::ReportedLevel> reported_level_;
+
   // A vector of WebRtc VoE channels that the capturer sends data to.
   std::vector<int> voe_channels_;
-
-  // A vector of the peer connection sink adapters which receive the audio data
-  // from the audio track.
-  ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
-
-  // The amplitude of the signal.
-  int signal_level_;
-
-  // Thread checker for libjingle's signaling thread.
-  base::ThreadChecker signaling_thread_checker_;
-  base::ThreadChecker capture_thread_;
-
-  // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
-  mutable base::Lock lock_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_