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Side by Side Diff: content/renderer/media/webrtc_local_audio_track.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 2013 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_WEBRTC_LOCAL_AUDIO_TRACK_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
7
8 #include <list>
9 #include <string>
10
11 #include "base/macros.h"
12 #include "base/memory/ref_counted.h"
13 #include "base/memory/scoped_ptr.h"
14 #include "base/synchronization/lock.h"
15 #include "base/threading/thread_checker.h"
16 #include "content/renderer/media/media_stream_audio_track.h"
17 #include "content/renderer/media/tagged_list.h"
18 #include "media/audio/audio_parameters.h"
19
20 namespace media {
21 class AudioBus;
22 }
23
24 namespace content {
25
26 class MediaStreamAudioLevelCalculator;
27 class MediaStreamAudioProcessor;
28 class MediaStreamAudioSink;
29 class MediaStreamAudioSinkOwner;
30 class MediaStreamAudioTrackSink;
31 class WebAudioCapturerSource;
32 class WebRtcAudioCapturer;
33 class WebRtcLocalAudioTrackAdapter;
34
35 // A WebRtcLocalAudioTrack instance contains the implementations of
36 // MediaStreamTrackExtraData.
37 // When an instance is created, it will register itself as a track to the
38 // WebRtcAudioCapturer to get the captured data, and forward the data to
39 // its |sinks_|. The data flow can be stopped by disabling the audio track.
40 // TODO(tommi): Rename to MediaStreamLocalAudioTrack.
41 class CONTENT_EXPORT WebRtcLocalAudioTrack
42 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
43 public:
44 WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
45 const scoped_refptr<WebRtcAudioCapturer>& capturer,
46 WebAudioCapturerSource* webaudio_source);
47
48 ~WebRtcLocalAudioTrack() override;
49
50 // Add a sink to the track. This function will trigger a OnSetFormat()
51 // call on the |sink|.
52 // Called on the main render thread.
53 void AddSink(MediaStreamAudioSink* sink) override;
54
55 // Remove a sink from the track.
56 // Called on the main render thread.
57 void RemoveSink(MediaStreamAudioSink* sink) override;
58
59 // Starts the local audio track. Called on the main render thread and
60 // should be called only once when audio track is created.
61 void Start();
62
63 // Overrides for MediaStreamTrack.
64
65 void SetEnabled(bool enabled) override;
66
67 // Stops the local audio track. Called on the main render thread and
68 // should be called only once when audio track going away.
69 void Stop() override;
70
71 webrtc::AudioTrackInterface* GetAudioAdapter() override;
72
73 // Returns the output format of the capture source. May return an invalid
74 // AudioParameters if the format is not yet available.
75 // Called on the main render thread.
76 media::AudioParameters GetOutputFormat() const override;
77
78 // Method called by the capturer to deliver the capture data.
79 // Called on the capture audio thread.
80 void Capture(const media::AudioBus& audio_bus,
81 base::TimeTicks estimated_capture_time,
82 bool force_report_nonzero_energy);
83
84 // Method called by the capturer to set the audio parameters used by source
85 // of the capture data..
86 // Called on the capture audio thread.
87 void OnSetFormat(const media::AudioParameters& params);
88
89 // Method called by the capturer to set the processor that applies signal
90 // processing on the data of the track.
91 // Called on the capture audio thread.
92 void SetAudioProcessor(
93 const scoped_refptr<MediaStreamAudioProcessor>& processor);
94
95 private:
96 typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
97
98 // All usage of libjingle is through this adapter. The adapter holds
99 // a pointer to this object, but no reference.
100 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
101
102 // The provider of captured data to render.
103 scoped_refptr<WebRtcAudioCapturer> capturer_;
104
105 // The source of the audio track which is used by WebAudio, which provides
106 // data to the audio track when hooking up with WebAudio.
107 scoped_refptr<WebAudioCapturerSource> webaudio_source_;
108
109 // A tagged list of sinks that the audio data is fed to. Tags
110 // indicate tracks that need to be notified that the audio format
111 // has changed.
112 SinkList sinks_;
113
114 // Tests that methods are called on libjingle's signaling thread.
115 base::ThreadChecker signal_thread_checker_;
116
117 // Used to DCHECK that some methods are called on the capture audio thread.
118 base::ThreadChecker capture_thread_checker_;
119
120 // Protects |params_| and |sinks_|.
121 mutable base::Lock lock_;
122
123 // Audio parameters of the audio capture stream.
124 // Accessed on only the audio capture thread.
125 media::AudioParameters audio_parameters_;
126
127 // Used to calculate the signal level that shows in the UI.
128 // Accessed on only the audio thread.
129 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
130
131 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
132 };
133
134 } // namespace content
135
136 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
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