| // Copyright (c) 2012 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_ |
| |
| #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 "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 { |
| public: |
| explicit WebAudioCapturerSource( |
| const blink::WebMediaStreamSource& blink_source); |
| |
| // WebAudioDestinationConsumer implementation. |
| // setFormat() is called early on, so that we can configure the audio track. |
| 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 when the media audio track is stopping. |
| void Stop(); |
| |
| 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_; |
| |
| // Flag to help notify the |track_| when the audio format has changed. |
| bool audio_format_changed_; |
| |
| // Wraps data coming from HandleCapture(). |
| 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_; |
| |
| DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_ |