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

content/renderer/media/webaudio_media_stream_source.h

Index: content/renderer/media/webaudio_media_stream_source.h
diff --git a/content/renderer/media/webaudio_capturer_source.h b/content/renderer/media/webaudio_media_stream_source.h
similarity index 25%
rename from content/renderer/media/webaudio_capturer_source.h
rename to content/renderer/media/webaudio_media_stream_source.h
index b0ee262ccd7c3c06135b9252b3b394db8c0dc7dd..ae854542830144415c79671b6e2fa5fff31b021b 100644
--- a/content/renderer/media/webaudio_capturer_source.h
+++ b/content/renderer/media/webaudio_media_stream_source.h
@@ -1,100 +1,77 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright (c) 2016 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 "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/audio_capturer_source.h"
-#include "media/base/audio_fifo.h"
+#include "base/memory/scoped_ptr.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_rechunker.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;
-
-// 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 base::RefCountedThreadSafe<WebAudioCapturerSource>,
-       public blink::WebAudioDestinationConsumer {
+// 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
+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      public blink::WebAudioDestinationConsumer {
  public:
-  explicit WebAudioCapturerSource(
+  explicit WebAudioMediaStreamSource(
       const blink::WebMediaStreamSource& blink_source);
 
+  ~WebAudioMediaStreamSource() final;
+
+ 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. |source_provider| is the source of
-  // the data flow where stream information like delay, volume, key_pressed,
-  // is stored.
-  void Start(WebRtcLocalAudioTrack* track);
+  // Called by AudioRechunker zero or more times during the call to
+  // consumeAudio().  Delivers re-chunked audio data to the tracks.
+  void DeliverRechunkedAudio(const media::AudioBus& audio_bus,
+                             base::TimeDelta reference_timestamp);
 
-  // Called when the media audio track is stopping.
-  void Stop();
+  // MediaStreamAudioSource implementation.
+  void DoStopSource() final;
+  bool EnsureSourceIsStarted() final;
 
- protected:
-  friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
-  ~WebAudioCapturerSource() override;
-
- private:
   // Removes this object from a blink::WebMediaStreamSource with which it
   // might be registered. The goal is to avoid dangling pointers.
   void removeFromBlinkSource();
 
-  // Used to DCHECK that some methods are called on the correct thread.
-  base::ThreadChecker thread_checker_;
-
-  // The audio track this WebAudioCapturerSource is feeding data to.
-  // WebRtcLocalAudioTrack is reference counted, and owning this object.
-  // To avoid circular reference, a raw pointer is kept here.
-  WebRtcLocalAudioTrack* track_;
-
-  media::AudioParameters params_;
+  // This object registers and de-registers as an audio consumer of a
+  // blink::WebMediaStreamSource.
+  blink::WebMediaStreamSource blink_source_;
 
-  // 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_started_;
 
-  // Wraps data coming from HandleCapture().
+  // An adapter used for providing audio to |rechunker_|.
   scoped_ptr<media::AudioBus> wrapper_bus_;
 
-  // Bus for reading from FIFO and calling the CaptureCallback.
-  scoped_ptr<media::AudioBus> capture_bus_;
-
-  // Handles mismatch between WebAudio buffer size and WebRTC.
-  scoped_ptr<media::AudioFifo> fifo_;
-
-  // Synchronizes HandleCapture() with AudioCapturerSource calls.
-  base::Lock lock_;
-  bool started_;
-
-  // This object registers with a blink::WebMediaStreamSource. We keep track of
-  // that in order to be able to deregister before stopping the audio track.
-  blink::WebMediaStreamSource blink_source_;
+  // Takes in the audio data passed to consumeAudio() and re-chunks it into 10
+  // ms chunks for the tracks.  This ensures each chunk of audio delivered to
+  // the tracks has the same buffer size, even if audio is provided in
+  // varying-sized chunks.
+  media::AudioRechunker rechunker_;
 
-  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_