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 // Buffer to pass audio data to WebRtc. 85 scoped_ptr<int16[]> audio_data_; 86 87 // Synchronizes HandleCapture() with AudioCapturerSource calls. 88 base::Lock lock_; 89 bool started_; 90 91 DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource); 92 }; 93 94 } // namespace content 95 96 #endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_ 97
