content/renderer/media/webaudio_capturer_source.h

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
   6 #define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
   7
   8 #include "base/memory/ref_counted.h"
   9 #include "base/synchronization/lock.h"
  10 #include "base/threading/thread_checker.h"
  11 #include "media/audio/audio_parameters.h"
  12 #include "media/base/audio_capturer_source.h"
  13 #include "media/base/audio_fifo.h"
  14 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
  15 #include "third_party/WebKit/public/platform/WebVector.h"
  16
  17 namespace content {
  18
  19 class WebRtcLocalAudioTrack;
  20
  21 // WebAudioCapturerSource is the missing link between
  22 // WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
  23 //
  24 // 1. WebKit calls the setFormat() method setting up the basic stream format
  25 //    (channels, and sample-rate).
  26 // 2. consumeAudio() is called periodically by WebKit which dispatches the
  27 //    audio stream to the WebRtcLocalAudioTrack::Capture() method.
  28 class WebAudioCapturerSource
  29     :  public base::RefCountedThreadSafe<WebAudioCapturerSource>,
  30        public blink::WebAudioDestinationConsumer {
  31  public:
  32   WebAudioCapturerSource();
  33
  34   // WebAudioDestinationConsumer implementation.
  35   // setFormat() is called early on, so that we can configure the audio track.
  36   virtual void setFormat(size_t number_of_channels, float sample_rate) override;
  37   // MediaStreamAudioDestinationNode periodically calls consumeAudio().
  38   // Called on the WebAudio audio thread.
  39   virtual void consumeAudio(const blink::WebVector<const float*>& audio_data,
  40       size_t number_of_frames) override;
  41
  42   // Called when the WebAudioCapturerSource is hooking to a media audio track.
  43   // |track| is the sink of the data flow. |source_provider| is the source of
  44   // the data flow where stream information like delay, volume, key_pressed,
  45   // is stored.
  46   void Start(WebRtcLocalAudioTrack* track);
  47
  48   // Called when the media audio track is stopping.
  49   void Stop();
  50
  51  protected:
  52   friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
  53   virtual ~WebAudioCapturerSource();
  54
  55  private:
  56   // Used to DCHECK that some methods are called on the correct thread.
  57   base::ThreadChecker thread_checker_;
  58
  59   // The audio track this WebAudioCapturerSource is feeding data to.
  60   // WebRtcLocalAudioTrack is reference counted, and owning this object.
  61   // To avoid circular reference, a raw pointer is kept here.
  62   WebRtcLocalAudioTrack* track_;
  63
  64   media::AudioParameters params_;
  65
  66   // Flag to help notify the |track_| when the audio format has changed.
  67   bool audio_format_changed_;
  68
  69   // Wraps data coming from HandleCapture().
  70   scoped_ptr<media::AudioBus> wrapper_bus_;
  71
  72   // Bus for reading from FIFO and calling the CaptureCallback.
  73   scoped_ptr<media::AudioBus> capture_bus_;
  74
  75   // Handles mismatch between WebAudio buffer size and WebRTC.
  76   scoped_ptr<media::AudioFifo> fifo_;
  77
  78   // Synchronizes HandleCapture() with AudioCapturerSource calls.
  79   base::Lock lock_;
  80   bool started_;
  81
  82   DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
  83 };
  84
  85 }  // namespace content
  86
  87 #endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_