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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 WebRtcAudioCapturer;
20 class WebRtcLocalAudioTrack;
21 
22 // WebAudioCapturerSource is the missing link between
23 // WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
24 //
25 // 1. WebKit calls the setFormat() method setting up the basic stream format
26 //    (channels, and sample-rate).
27 // 2. consumeAudio() is called periodically by WebKit which dispatches the
28 //    audio stream to the WebRtcLocalAudioTrack::Capture() method.
29 class WebAudioCapturerSource
30     :  public base::RefCountedThreadSafe<WebAudioCapturerSource>,
31        public blink::WebAudioDestinationConsumer {
32  public:
33   WebAudioCapturerSource();
34 
35   // WebAudioDestinationConsumer implementation.
36   // setFormat() is called early on, so that we can configure the audio track.
37   virtual void setFormat(size_t number_of_channels, float sample_rate) OVERRIDE;
38   // MediaStreamAudioDestinationNode periodically calls consumeAudio().
39   // Called on the WebAudio audio thread.
40   virtual void consumeAudio(const blink::WebVector<const float*>& audio_data,
41       size_t number_of_frames) OVERRIDE;
42 
43   // Called when the WebAudioCapturerSource is hooking to a media audio track.
44   // |track| is the sink of the data flow. |source_provider| is the source of
45   // the data flow where stream information like delay, volume, key_pressed,
46   // is stored.
47   void Start(WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer);
48 
49   // Called when the media audio track is stopping.
50   void Stop();
51 
52  protected:
53   friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
54   virtual ~WebAudioCapturerSource();
55 
56  private:
57   // Used to DCHECK that some methods are called on the correct thread.
58   base::ThreadChecker thread_checker_;
59 
60   // The audio track this WebAudioCapturerSource is feeding data to.
61   // WebRtcLocalAudioTrack is reference counted, and owning this object.
62   // To avoid circular reference, a raw pointer is kept here.
63   WebRtcLocalAudioTrack* track_;
64 
65   // A raw pointer to the capturer to get audio processing params like
66   // delay, volume, key_pressed information.
67   // This |capturer_| is guaranteed to outlive this object.
68   WebRtcAudioCapturer* capturer_;
69 
70   media::AudioParameters params_;
71 
72   // Flag to help notify the |track_| when the audio format has changed.
73   bool audio_format_changed_;
74 
75   // Wraps data coming from HandleCapture().
76   scoped_ptr<media::AudioBus> wrapper_bus_;
77 
78   // Bus for reading from FIFO and calling the CaptureCallback.
79   scoped_ptr<media::AudioBus> capture_bus_;
80 
81   // Handles mismatch between WebAudio buffer size and WebRTC.
82   scoped_ptr<media::AudioFifo> fifo_;
83 
84   // Synchronizes HandleCapture() with AudioCapturerSource calls.
85   base::Lock lock_;
86   bool started_;
87 
88   DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
89 };
90 
91 }  // namespace content
92 
93 #endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
94