chrome.bluetoothSocket: clean-up Listen functions
[chromium-blink-merge.git] / content / renderer / media / webaudio_capturer_source.h
blobfdd3f9c68ff3e438d541de3282a350a7b43e6446
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.
5 #ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
6 #define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
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"
17 namespace content {
19 class WebRtcAudioCapturer;
20 class WebRtcLocalAudioTrack;
22 // WebAudioCapturerSource is the missing link between
23 // WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
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();
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;
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);
49 // Called when the media audio track is stopping.
50 void Stop();
52 protected:
53 friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
54 virtual ~WebAudioCapturerSource();
56 private:
57 // Used to DCHECK that some methods are called on the correct thread.
58 base::ThreadChecker thread_checker_;
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_;
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_;
70 media::AudioParameters params_;
72 // Flag to help notify the |track_| when the audio format has changed.
73 bool audio_format_changed_;
75 // Wraps data coming from HandleCapture().
76 scoped_ptr<media::AudioBus> wrapper_bus_;
78 // Bus for reading from FIFO and calling the CaptureCallback.
79 scoped_ptr<media::AudioBus> capture_bus_;
81 // Handles mismatch between WebAudio buffer size and WebRTC.
82 scoped_ptr<media::AudioFifo> fifo_;
84 // Buffer to pass audio data to WebRtc.
85 scoped_ptr<int16[]> audio_data_;
87 // Synchronizes HandleCapture() with AudioCapturerSource calls.
88 base::Lock lock_;
89 bool started_;
91 DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
94 } // namespace content
96 #endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_