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

content/renderer/media/media_stream_audio_source.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_MEDIA_STREAM_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
 
 #include <memory>
+#include <string>
 
 #include "base/compiler_specific.h"
 #include "base/macros.h"
 #include "base/memory/weak_ptr.h"
 #include "content/common/content_export.h"
+#include "content/renderer/media/media_stream_audio_deliverer.h"
 #include "content/renderer/media/media_stream_source.h"
-#include "content/renderer/media/webaudio_capturer_source.h"
-#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace content {
 
 class MediaStreamAudioTrack;
 
-// TODO(miu): In a soon-upcoming set of refactoring changes, this class will
-// become a base class for managing tracks (part of what WebRtcAudioCapturer
-// does today).  Then, the rest of WebRtcAudioCapturer will be rolled into a
-// subclass.   http://crbug.com/577874
+// Represents a source of audio, and manages the delivery of audio data between
+// the source implementation and one or more MediaStreamAudioTracks.  This is

[perkj_chrome, 2016/04/20 13:34:54]

I am starting to think you want to have to spaces after each "." ? 

[miu, 2016/04/20 22:04:53]

Done.  Sorry, I'm old-school (it's a habit).  I've noticed there's a mix of both
1 or 2 spaces after a period.  I did some searching around on the web and it
seems I should just start adopting the 1-space rule.  :)

+// a base class providing all the necessary functionality to connect tracks and
+// have audio data delivered to them.  Subclasses provide the actual audio
+// source implementation (e.g., media::AudioCapturerSource), and should
+// implement the EnsureSourceIsStarted() and EnsureSourceIsStopped() methods,
+// and call SetFormat() and DeliverDataToTracks().
+//
+// This base class can be instantiated, to be used as a place-holder or a "null"
+// source of audio.  This can be useful for unit testing, wherever a mock is
+// needed, and/or calls to DeliverDataToTracks() must be made at very specific
+// times.
+//
+// An instance of this class is owned by blink::WebMediaStreamSource.
+//
+// Usage example:
+//
+//   class MyAudioSource : public MediaStreamSource { ... };
+//
+//   blink::WebMediaStreamSource blink_source = ...;
+//   blink::WebMediaStreamTrack blink_track = ...;
+//   blink_source.setExtraData(new MyAudioSource());  // Takes ownership.
+//   if (MediaStreamAudioSource::From(blink_source)
+//           ->ConnectToTrack(blink_track)) {
+//     LOG(INFO) << "Success!";
+//   } else {
+//     LOG(ERROR) << "Failed!";
+//   }
 class CONTENT_EXPORT MediaStreamAudioSource
     : NON_EXPORTED_BASE(public MediaStreamSource) {
  public:
-  MediaStreamAudioSource(int render_frame_id,
-                         const StreamDeviceInfo& device_info,
-                         const SourceStoppedCallback& stop_callback,
-                         PeerConnectionDependencyFactory* factory);
-  MediaStreamAudioSource();
+  explicit MediaStreamAudioSource(bool is_local_source);
   ~MediaStreamAudioSource() override;
 
   // Returns the MediaStreamAudioSource instance owned by the given blink
   // |source| or null.
-  static MediaStreamAudioSource* From(const blink::WebMediaStreamSource& track);
+  static MediaStreamAudioSource* From(
+      const blink::WebMediaStreamSource& source);
 
-  void AddTrack(const blink::WebMediaStreamTrack& track,
-                const blink::WebMediaConstraints& constraints,
-                const ConstraintsCallback& callback);
-
+  // Provides a weak reference to this MediaStreamAudioSource.  The weak
+  // pointer may only be dereferenced on the main thread.
   base::WeakPtr<MediaStreamAudioSource> GetWeakPtr() {
     return weak_factory_.GetWeakPtr();
   }
 
+  // Returns true if the source of audio is local to the application (e.g.,
+  // microphone input or loopback audio capture) as opposed to audio being
+  // streamed-in from outside the application.
+  bool is_local_source() const { return is_local_source_; }
+
+  // Connects this source to the given |track|, creating the appropriate
+  // implementation of the content::MediaStreamAudioTrack interface, which
+  // becomes associated with and owned by |track|.
+  //
+  // Returns true if the source was successfully started and the
+  // MediaStreamAudioTrack assigned to |track.extraData()|.
+  bool ConnectToTrack(const blink::WebMediaStreamTrack& track);
+
+  // Returns the current format of the audio passing through this source to the
+  // sinks.  This can return invalid parameters if the source has not yet been
+  // started.  This method is thread-safe.
+  media::AudioParameters GetAudioParameters() const;
+
+  // Returns a unique class identifier.  Some subclasses override and use this
+  // method to provide safe down-casting to their type.
+  virtual void* GetClassIdentifier() const;
+
+ protected:
+  // Returns a new MediaStreamAudioTrack.  |id| is the blink track's ID in
+  // UTF-8.  Subclasses may override this to provide an extended implementation.
+  virtual std::unique_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
+      const std::string& id);
+
+  // Returns true if the source has already been started and has not yet been
+  // stopped.  Otherwise, attempts to start the source and returns true if
+  // successful.  While the source is running, it may provide audio on any
+  // thread by calling DeliverDataToTracks().
+  //
+  // A default no-op implementation is provided in this base class.  Subclasses
+  // should override this method.
+  virtual bool EnsureSourceIsStarted();
+
+  // Stops the source and guarantees the the flow of audio data has stopped
+  // (i.e., by the time this method returns, there will be no further calls to
+  // DeliverDataToTracks() on any thread).
+  //
+  // A default no-op implementation is provided in this base class.  Subclasses
+  // should override this method.
+  virtual void EnsureSourceIsStopped();
+
+  // Called by subclasses to update the format of the audio passing through this
+  // source to the sinks.  This may be called at any time, before or after
+  // tracks have been connected; but must be called at least once before
+  // DeliverDataToTracks().  This method is thread-safe.
+  void SetFormat(const media::AudioParameters& params);
+
+  // Called by subclasses to deliver audio data to the currently-connected
+  // tracks.  This method is thread-safe.
+  void DeliverDataToTracks(const media::AudioBus& audio_bus,
+                           base::TimeTicks reference_time);
+
+ private:
+  // MediaStreamSource override.
+  void DoStopSource() final;
+
   // Removes |track| from the list of instances that get a copy of the source
-  // audio data.
+  // audio data.  The "stop callback" that was provided to the track calls
+  // this.
   void StopAudioDeliveryTo(MediaStreamAudioTrack* track);
 
-  WebRtcAudioCapturer* audio_capturer() const { return audio_capturer_.get(); }
+  // True if the source of audio is a local device.  False if the source is
+  // remote (e.g., streamed-in from a server).
+  const bool is_local_source_;
 
-  void SetAudioCapturer(std::unique_ptr<WebRtcAudioCapturer> capturer) {
-    DCHECK(!audio_capturer_.get());
-    audio_capturer_ = std::move(capturer);
-  }
+  // In debug builds, check that all methods that could cause object graph
+  // or data flow changes are being called on the main thread.
+  base::ThreadChecker thread_checker_;
 
-  webrtc::AudioSourceInterface* local_audio_source() {
-    return local_audio_source_.get();
-  }
+  // Set to true once this source has been permanently stopped.
+  bool is_stopped_;
 
-  void SetLocalAudioSource(scoped_refptr<webrtc::AudioSourceInterface> source) {
-    local_audio_source_ = std::move(source);
-  }
-
-  WebAudioCapturerSource* webaudio_capturer() const {
-    return webaudio_capturer_.get();
-  }
-
-  void SetWebAudioCapturer(std::unique_ptr<WebAudioCapturerSource> capturer) {
-    DCHECK(!webaudio_capturer_.get());
-    webaudio_capturer_ = std::move(capturer);
-  }
-
- protected:
-  void DoStopSource() override;
-
- private:
-  const int render_frame_id_;
-  PeerConnectionDependencyFactory* const factory_;
-
-  // MediaStreamAudioSource is the owner of either a WebRtcAudioCapturer or a
-  // WebAudioCapturerSource.
-  //
-  // TODO(miu): In a series of soon-upcoming changes, WebRtcAudioCapturer and
-  // WebAudioCapturerSource will become subclasses of MediaStreamAudioSource
-  // instead.
-  std::unique_ptr<WebRtcAudioCapturer> audio_capturer_;
-  std::unique_ptr<WebAudioCapturerSource> webaudio_capturer_;
-
-  // This member holds an instance of webrtc::LocalAudioSource. This is used
-  // as a container for audio options.
-  scoped_refptr<webrtc::AudioSourceInterface> local_audio_source_;
+  // Manages tracks connected to this source and the audio format and data flow.
+  MediaStreamAudioDeliverer<MediaStreamAudioTrack> deliverer_;
 
   // Provides weak pointers so that MediaStreamAudioTracks won't call
   // StopAudioDeliveryTo() if this instance dies first.
   base::WeakPtrFactory<MediaStreamAudioSource> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_