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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2016 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_ |
| 7 | 7 |
| 8 #include <stddef.h> | 8 #include "base/memory/scoped_ptr.h" |
| 9 | 9 #include "base/time/time.h" |
| 10 #include "base/macros.h" | 10 #include "content/renderer/media/media_stream_audio_source.h" |
| 11 #include "base/memory/ref_counted.h" | 11 #include "media/base/audio_bus.h" |
| 12 #include "base/synchronization/lock.h" | 12 #include "media/base/audio_rechunker.h" |
| 13 #include "base/threading/thread_checker.h" | |
| 14 #include "media/audio/audio_parameters.h" | |
| 15 #include "media/base/audio_capturer_source.h" | |
| 16 #include "media/base/audio_fifo.h" | |
| 17 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h" | 13 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h" |
| 18 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" | 14 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
| 19 #include "third_party/WebKit/public/platform/WebVector.h" | 15 #include "third_party/WebKit/public/platform/WebVector.h" |
| 20 | 16 |
| 21 namespace content { | 17 namespace content { |
| 22 | 18 |
| 23 class WebRtcLocalAudioTrack; | 19 // Implements the WebAudioDestinationConsumer interface to provide a source of |
| 24 | 20 // audio data (i.e., the output from a graph of WebAudio nodes) to one or more |
| 25 // WebAudioCapturerSource is the missing link between | 21 // MediaStreamAudioTracks. Audio data is transported directly to the tracks in |
| 26 // WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack. | 22 // 10 ms chunks. |
| 27 // | 23 class WebAudioMediaStreamSource |
| 28 // 1. WebKit calls the setFormat() method setting up the basic stream format | 24 : NON_EXPORTED_BASE(public MediaStreamAudioSource), |
| 29 // (channels, and sample-rate). | 25 public blink::WebAudioDestinationConsumer { |
| 30 // 2. consumeAudio() is called periodically by WebKit which dispatches the | |
| 31 // audio stream to the WebRtcLocalAudioTrack::Capture() method. | |
| 32 class WebAudioCapturerSource | |
| 33 : public base::RefCountedThreadSafe<WebAudioCapturerSource>, | |
| 34 public blink::WebAudioDestinationConsumer { | |
| 35 public: | 26 public: |
| 36 explicit WebAudioCapturerSource( | 27 explicit WebAudioMediaStreamSource( |
| 37 const blink::WebMediaStreamSource& blink_source); | 28 const blink::WebMediaStreamSource& blink_source); |
| 38 | 29 |
| 30 ~WebAudioMediaStreamSource() final; |
| 31 |
| 32 private: |
| 39 // WebAudioDestinationConsumer implementation. | 33 // WebAudioDestinationConsumer implementation. |
| 40 // setFormat() is called early on, so that we can configure the audio track. | 34 // |
| 35 // Note: Blink ensures setFormat() and consumeAudio() are not called |
| 36 // concurrently across threads, but these methods could be called on any |
| 37 // thread. |
| 41 void setFormat(size_t number_of_channels, float sample_rate) override; | 38 void setFormat(size_t number_of_channels, float sample_rate) override; |
| 42 // MediaStreamAudioDestinationNode periodically calls consumeAudio(). | |
| 43 // Called on the WebAudio audio thread. | |
| 44 void consumeAudio(const blink::WebVector<const float*>& audio_data, | 39 void consumeAudio(const blink::WebVector<const float*>& audio_data, |
| 45 size_t number_of_frames) override; | 40 size_t number_of_frames) override; |
| 46 | 41 |
| 47 // Called when the WebAudioCapturerSource is hooking to a media audio track. | 42 // Called by AudioRechunker zero or more times during the call to |
| 48 // |track| is the sink of the data flow. |source_provider| is the source of | 43 // consumeAudio(). Delivers re-chunked audio data to the tracks. |
| 49 // the data flow where stream information like delay, volume, key_pressed, | 44 void DeliverRechunkedAudio(const media::AudioBus& audio_bus, |
| 50 // is stored. | 45 base::TimeDelta reference_timestamp); |
| 51 void Start(WebRtcLocalAudioTrack* track); | |
| 52 | 46 |
| 53 // Called when the media audio track is stopping. | 47 // MediaStreamAudioSource implementation. |
| 54 void Stop(); | 48 void DoStopSource() final; |
| 49 bool EnsureSourceIsStarted() final; |
| 55 | 50 |
| 56 protected: | |
| 57 friend class base::RefCountedThreadSafe<WebAudioCapturerSource>; | |
| 58 ~WebAudioCapturerSource() override; | |
| 59 | |
| 60 private: | |
| 61 // Removes this object from a blink::WebMediaStreamSource with which it | 51 // Removes this object from a blink::WebMediaStreamSource with which it |
| 62 // might be registered. The goal is to avoid dangling pointers. | 52 // might be registered. The goal is to avoid dangling pointers. |
| 63 void removeFromBlinkSource(); | 53 void removeFromBlinkSource(); |
| 64 | 54 |
| 65 // Used to DCHECK that some methods are called on the correct thread. | 55 // This object registers and de-registers as an audio consumer of a |
| 66 base::ThreadChecker thread_checker_; | 56 // blink::WebMediaStreamSource. |
| 57 blink::WebMediaStreamSource blink_source_; |
| 67 | 58 |
| 68 // The audio track this WebAudioCapturerSource is feeding data to. | 59 // True while this WebAudioMediaStreamSource is registered with |
| 69 // WebRtcLocalAudioTrack is reference counted, and owning this object. | 60 // |blink_source_| and is consuming audio. |
| 70 // To avoid circular reference, a raw pointer is kept here. | 61 bool is_started_; |
| 71 WebRtcLocalAudioTrack* track_; | |
| 72 | 62 |
| 73 media::AudioParameters params_; | 63 // An adapter used for providing audio to |rechunker_|. |
| 74 | |
| 75 // Flag to help notify the |track_| when the audio format has changed. | |
| 76 bool audio_format_changed_; | |
| 77 | |
| 78 // Wraps data coming from HandleCapture(). | |
| 79 scoped_ptr<media::AudioBus> wrapper_bus_; | 64 scoped_ptr<media::AudioBus> wrapper_bus_; |
| 80 | 65 |
| 81 // Bus for reading from FIFO and calling the CaptureCallback. | 66 // Takes in the audio data passed to consumeAudio() and re-chunks it into 10 |
| 82 scoped_ptr<media::AudioBus> capture_bus_; | 67 // ms chunks for the tracks. This ensures each chunk of audio delivered to |
| 68 // the tracks has the same buffer size, even if audio is provided in |
| 69 // varying-sized chunks. |
| 70 media::AudioRechunker rechunker_; |
| 83 | 71 |
| 84 // Handles mismatch between WebAudio buffer size and WebRTC. | 72 DISALLOW_COPY_AND_ASSIGN(WebAudioMediaStreamSource); |
| 85 scoped_ptr<media::AudioFifo> fifo_; | |
| 86 | |
| 87 // Synchronizes HandleCapture() with AudioCapturerSource calls. | |
| 88 base::Lock lock_; | |
| 89 bool started_; | |
| 90 | |
| 91 // This object registers with a blink::WebMediaStreamSource. We keep track of | |
| 92 // that in order to be able to deregister before stopping the audio track. | |
| 93 blink::WebMediaStreamSource blink_source_; | |
| 94 | |
| 95 DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource); | |
| 96 }; | 73 }; |
| 97 | 74 |
| 98 } // namespace content | 75 } // namespace content |
| 99 | 76 |
| 100 #endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_ | 77 #endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_ |
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