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

content/renderer/media/webaudio_media_stream_source.h

 // Copyright (c) 2012 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_WEBAUDIO_CAPTURER_SOURCE_H_
-#define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
+#ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_
 
-#include <stddef.h>
+#include <memory>
 
-#include "base/macros.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
 #include "base/time/time.h"
+#include "content/renderer/media/media_stream_audio_source.h"
 #include "media/base/audio_bus.h"
-#include "media/base/audio_capturer_source.h"
-#include "media/base/audio_parameters.h"
 #include "media/base/audio_push_fifo.h"
 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 
 namespace content {
 
-class WebRtcLocalAudioTrack;
+// Implements the WebAudioDestinationConsumer interface to provide a source of
+// audio data (i.e., the output from a graph of WebAudio nodes) to one or more
+// MediaStreamAudioTracks. Audio data is transported directly to the tracks in
+// 10 ms chunks.
+class WebAudioMediaStreamSource final
+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      public blink::WebAudioDestinationConsumer {
+ public:
+  explicit WebAudioMediaStreamSource(blink::WebMediaStreamSource* blink_source);
 
-// WebAudioCapturerSource is the missing link between
-// WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
-//
-// 1. WebKit calls the setFormat() method setting up the basic stream format
-//    (channels, and sample-rate).
-// 2. consumeAudio() is called periodically by WebKit which dispatches the
-//    audio stream to the WebRtcLocalAudioTrack::Capture() method.
-class WebAudioCapturerSource : public blink::WebAudioDestinationConsumer {
- public:
-  explicit WebAudioCapturerSource(blink::WebMediaStreamSource* blink_source);
+  ~WebAudioMediaStreamSource() override;
 
-  ~WebAudioCapturerSource() override;
-
+ private:
   // WebAudioDestinationConsumer implementation.
-  // setFormat() is called early on, so that we can configure the audio track.
+  //
+  // Note: Blink ensures setFormat() and consumeAudio() are not called
+  // concurrently across threads, but these methods could be called on any
+  // thread.
   void setFormat(size_t number_of_channels, float sample_rate) override;
-  // MediaStreamAudioDestinationNode periodically calls consumeAudio().
-  // Called on the WebAudio audio thread.
   void consumeAudio(const blink::WebVector<const float*>& audio_data,
                     size_t number_of_frames) override;
 
-  // Called when the WebAudioCapturerSource is hooking to a media audio track.
-  // |track| is the sink of the data flow and must remain alive until Stop() is
-  // called.
-  void Start(WebRtcLocalAudioTrack* track);
-
-  // Called when the media audio track is stopping.
-  void Stop();
-
- private:
   // Called by AudioPushFifo zero or more times during the call to
   // consumeAudio().  Delivers audio data with the required buffer size to the
-  // track.
+  // tracks.
   void DeliverRebufferedAudio(const media::AudioBus& audio_bus,
                               int frame_delay);
 
-  // Deregisters this object from its blink::WebMediaStreamSource.
-  void DeregisterFromBlinkSource();
+  // MediaStreamAudioSource implementation.
+  bool EnsureSourceIsStarted() final;
+  void EnsureSourceIsStopped() final;
 
-  // Used to DCHECK that some methods are called on the correct thread.
+  // 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_;
 
-  // The audio track this WebAudioCapturerSource is feeding data to.
-  WebRtcLocalAudioTrack* track_;
-
-  media::AudioParameters params_;
-
-  // Flag to help notify the |track_| when the audio format has changed.
-  bool audio_format_changed_;
+  // True while this WebAudioMediaStreamSource is registered with
+  // |blink_source_| and is consuming audio.
+  bool is_registered_consumer_;
 
   // A wrapper used for providing audio to |fifo_|.
   std::unique_ptr<media::AudioBus> wrapper_bus_;
 
   // Takes in the audio data passed to consumeAudio() and re-buffers it into 10
-  // ms chunks for the track.  This ensures each chunk of audio delivered to the
-  // track has the required buffer size, regardless of the amount of audio
+  // ms chunks for the tracks. This ensures each chunk of audio delivered to
+  // the tracks has the required buffer size, regardless of the amount of audio
   // provided via each consumeAudio() call.
   media::AudioPushFifo fifo_;
 
   // Used to pass the reference timestamp between DeliverDecodedAudio() and
   // DeliverRebufferedAudio().
   base::TimeTicks current_reference_time_;
 
-  // Synchronizes HandleCapture() with AudioCapturerSource calls.
-  base::Lock lock_;
-
   // This object registers with a blink::WebMediaStreamSource. We keep track of
-  // that in order to be able to deregister before stopping the audio track.
+  // that in order to be able to deregister before stopping this source.
   blink::WebMediaStreamSource blink_source_;
 
-  DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
+  DISALLOW_COPY_AND_ASSIGN(WebAudioMediaStreamSource);
 };
 
 }  // namespace content
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
+#endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_