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Side by Side Diff: content/renderer/media/webaudio_media_stream_source.h

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
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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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