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

Issue 1721273002: MediaStream audio object graph untangling and clean-ups. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Addressed mcasas's 1st round comments, plus REBASE. Created 4 years, 9 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 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 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_WEBRTC_AUDIO_CAPTURER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
7 7
8 #include <list> 8 #include <list>
9 #include <string> 9 #include <string>
10 10
11 #include "base/callback.h" 11 #include "base/callback.h"
12 #include "base/files/file.h" 12 #include "base/files/file.h"
13 #include "base/macros.h" 13 #include "base/macros.h"
14 #include "base/memory/ref_counted.h" 14 #include "base/memory/ref_counted.h"
15 #include "base/memory/scoped_ptr.h"
16 #include "base/memory/weak_ptr.h"
15 #include "base/synchronization/lock.h" 17 #include "base/synchronization/lock.h"
16 #include "base/threading/thread_checker.h" 18 #include "base/threading/thread_checker.h"
17 #include "base/time/time.h" 19 #include "base/time/time.h"
18 #include "content/common/media/media_stream_options.h" 20 #include "content/common/media/media_stream_options.h"
21 #include "content/renderer/media/media_stream_audio_level_calculator.h"
19 #include "content/renderer/media/tagged_list.h" 22 #include "content/renderer/media/tagged_list.h"
20 #include "media/audio/audio_input_device.h" 23 #include "media/audio/audio_input_device.h"
21 #include "media/base/audio_capturer_source.h" 24 #include "media/base/audio_capturer_source.h"
22 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 25 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
23 26
24 namespace media { 27 namespace media {
25 class AudioBus; 28 class AudioBus;
26 } 29 }
27 30
28 namespace content { 31 namespace content {
29 32
30 class MediaStreamAudioProcessor; 33 class MediaStreamAudioProcessor;
31 class MediaStreamAudioSource; 34 class MediaStreamAudioSource;
32 class WebRtcAudioDeviceImpl; 35 class WebRtcAudioDeviceImpl;
33 class WebRtcLocalAudioRenderer; 36 class WebRtcLocalAudioRenderer;
34 class WebRtcLocalAudioTrack; 37 class WebRtcLocalAudioTrack;
35 38
36 // This class manages the capture data flow by getting data from its 39 // This class manages the capture data flow by getting data from its
37 // |source_|, and passing it to its |tracks_|. 40 // |source_|, and passing it to its |tracks_|.
38 // The threading model for this class is rather complex since it will be 41 // The threading model for this class is rather complex since it will be
39 // created on the main render thread, captured data is provided on a dedicated 42 // created on the main render thread, captured data is provided on a dedicated
40 // AudioInputDevice thread, and methods can be called either on the Libjingle 43 // AudioInputDevice thread, and methods can be called either on the Libjingle
41 // thread or on the main render thread but also other client threads 44 // thread or on the main render thread but also other client threads
42 // if an alternative AudioCapturerSource has been set. 45 // if an alternative AudioCapturerSource has been set.
43 class CONTENT_EXPORT WebRtcAudioCapturer 46 class CONTENT_EXPORT WebRtcAudioCapturer
44 : public base::RefCountedThreadSafe<WebRtcAudioCapturer>, 47 : NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
45 NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
46 public: 48 public:
47 // Used to construct the audio capturer. |render_frame_id| specifies the 49 // Used to construct the audio capturer. |render_frame_id| specifies the
48 // RenderFrame consuming audio for capture; -1 is used for tests. 50 // RenderFrame consuming audio for capture; -1 is used for tests.
49 // |device_info| contains all the device information that the capturer is 51 // |device_info| contains all the device information that the capturer is
50 // created for. |constraints| contains the settings for audio processing. 52 // created for. |constraints| contains the settings for audio processing.
51 // TODO(xians): Implement the interface for the audio source and move the 53 // TODO(xians): Implement the interface for the audio source and move the
52 // |constraints| to ApplyConstraints(). Called on the main render thread. 54 // |constraints| to ApplyConstraints(). Called on the main render thread.
53 static scoped_refptr<WebRtcAudioCapturer> CreateCapturer( 55 static scoped_ptr<WebRtcAudioCapturer> CreateCapturer(
54 int render_frame_id, 56 int render_frame_id,
55 const StreamDeviceInfo& device_info, 57 const StreamDeviceInfo& device_info,
56 const blink::WebMediaConstraints& constraints, 58 const blink::WebMediaConstraints& constraints,
57 WebRtcAudioDeviceImpl* audio_device, 59 WebRtcAudioDeviceImpl* audio_device,
58 MediaStreamAudioSource* audio_source); 60 MediaStreamAudioSource* audio_source);
59 61
62 ~WebRtcAudioCapturer() override;
63
60 // Add a audio track to the sinks of the capturer. 64 // Add a audio track to the sinks of the capturer.
61 // WebRtcAudioDeviceImpl calls this method on the main render thread but 65 // WebRtcAudioDeviceImpl calls this method on the main render thread but
62 // other clients may call it from other threads. The current implementation 66 // other clients may call it from other threads. The current implementation
63 // does not support multi-thread calling. 67 // does not support multi-thread calling.
64 // The first AddTrack will implicitly trigger the Start() of this object. 68 // The first AddTrack will implicitly trigger the Start() of this object.
65 void AddTrack(WebRtcLocalAudioTrack* track); 69 void AddTrack(WebRtcLocalAudioTrack* track);
66 70
67 // Remove a audio track from the sinks of the capturer. 71 // Remove a audio track from the sinks of the capturer.
68 // If the track has been added to the capturer, it must call RemoveTrack() 72 // If the track has been added to the capturer, it must call RemoveTrack()
69 // before it goes away. 73 // before it goes away.
70 // Called on the main render thread or libjingle working thread. 74 // Called on the main render thread or libjingle working thread.
71 void RemoveTrack(WebRtcLocalAudioTrack* track); 75 void RemoveTrack(WebRtcLocalAudioTrack* track);
72 76
73 // Called when a stream is connecting to a peer connection. This will set 77 // Called when a stream is connecting to a peer connection. This will set
74 // up the native buffer size for the stream in order to optimize the 78 // up the native buffer size for the stream in order to optimize the
75 // performance for peer connection. 79 // performance for peer connection.
76 void EnablePeerConnectionMode(); 80 void EnablePeerConnectionMode();
77 81
78 // Volume APIs used by WebRtcAudioDeviceImpl. 82 // Volume APIs used by WebRtcAudioDeviceImpl.
79 // Called on the AudioInputDevice audio thread. 83 // Called on the AudioInputDevice audio thread.
80 void SetVolume(int volume); 84 void SetVolume(int volume);
81 int Volume() const; 85 int Volume() const;
82 int MaxVolume() const; 86 int MaxVolume() const;
83 87
84 // Audio parameters utilized by the source of the audio capturer. 88 // Audio parameters utilized by the source of the audio capturer.
85 // TODO(phoglund): Think over the implications of this accessor and if we can 89 // TODO(phoglund): Think over the implications of this accessor and if we can
86 // remove it. 90 // remove it.
87 media::AudioParameters source_audio_parameters() const; 91 media::AudioParameters GetInputFormat() const;
88 92
89 // Gets information about the paired output device. Returns true if such a 93 const StreamDeviceInfo& device_info() const { return device_info_; }
90 // device exists.
91 bool GetPairedOutputParameters(int* session_id,
92 int* output_sample_rate,
93 int* output_frames_per_buffer) const;
94
95 const std::string& device_id() const { return device_info_.device.id; }
96 int session_id() const { return device_info_.session_id; }
97 94
98 // Stops recording audio. This method will empty its track lists since 95 // Stops recording audio. This method will empty its track lists since
99 // stopping the capturer will implicitly invalidate all its tracks. 96 // stopping the capturer will implicitly invalidate all its tracks.
100 // This method is exposed to the public because the MediaStreamAudioSource can 97 // This method is exposed to the public because the MediaStreamAudioSource can
101 // call Stop() 98 // call Stop()
102 void Stop(); 99 void Stop();
103 100
104 // Returns the output format. 101 // Returns the output format.
105 // Called on the main render thread. 102 // Called on the main render thread.
106 media::AudioParameters GetOutputFormat() const; 103 media::AudioParameters GetOutputFormat() const;
107 104
108 // Used by clients to inject their own source to the capturer. 105 // Used by clients to inject their own source to the capturer.
109 void SetCapturerSource( 106 void SetCapturerSource(
110 const scoped_refptr<media::AudioCapturerSource>& source, 107 const scoped_refptr<media::AudioCapturerSource>& source,
111 media::AudioParameters params); 108 media::AudioParameters params);
112 109
113 protected:
114 friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
115 ~WebRtcAudioCapturer() override;
116
117 private: 110 private:
118 class TrackOwner; 111 class TrackOwner;
119 typedef TaggedList<TrackOwner> TrackList; 112 typedef TaggedList<TrackOwner> TrackList;
120 113
121 WebRtcAudioCapturer(int render_frame_id, 114 WebRtcAudioCapturer(int render_frame_id,
122 const StreamDeviceInfo& device_info, 115 const StreamDeviceInfo& device_info,
123 const blink::WebMediaConstraints& constraints, 116 const blink::WebMediaConstraints& constraints,
124 WebRtcAudioDeviceImpl* audio_device, 117 WebRtcAudioDeviceImpl* audio_device,
125 MediaStreamAudioSource* audio_source); 118 MediaStreamAudioSource* audio_source);
126 119
(...skipping 10 matching lines...) Expand all
137 bool Initialize(); 130 bool Initialize();
138 131
139 // SetCapturerSourceInternal() is called if the client on the source side 132 // SetCapturerSourceInternal() is called if the client on the source side
140 // desires to provide their own captured audio data. Client is responsible 133 // desires to provide their own captured audio data. Client is responsible
141 // for calling Start() on its own source to get the ball rolling. 134 // for calling Start() on its own source to get the ball rolling.
142 // Called on the main render thread. 135 // Called on the main render thread.
143 // buffer_size is optional. Set to 0 to let it be chosen automatically. 136 // buffer_size is optional. Set to 0 to let it be chosen automatically.
144 void SetCapturerSourceInternal( 137 void SetCapturerSourceInternal(
145 const scoped_refptr<media::AudioCapturerSource>& source, 138 const scoped_refptr<media::AudioCapturerSource>& source,
146 media::ChannelLayout channel_layout, 139 media::ChannelLayout channel_layout,
147 int sample_rate, 140 int sample_rate);
148 int buffer_size);
149 141
150 // Starts recording audio. 142 // Starts recording audio.
151 // Triggered by AddSink() on the main render thread or a Libjingle working 143 // Triggered by AddSink() on the main render thread or a Libjingle working
152 // thread. It should NOT be called under |lock_|. 144 // thread. It should NOT be called under |lock_|.
153 void Start(); 145 void Start();
154 146
155 // Helper function to get the buffer size based on |peer_connection_mode_| 147 // Helper function to get the buffer size based on |peer_connection_mode_|
156 // and sample rate; 148 // and sample rate;
157 int GetBufferSize(int sample_rate) const; 149 int GetBufferSize(int sample_rate) const;
158 150
(...skipping 10 matching lines...) Expand all
169 TrackList tracks_; 161 TrackList tracks_;
170 162
171 // The audio data source from the browser process. 163 // The audio data source from the browser process.
172 scoped_refptr<media::AudioCapturerSource> source_; 164 scoped_refptr<media::AudioCapturerSource> source_;
173 165
174 // Cached audio constraints for the capturer. 166 // Cached audio constraints for the capturer.
175 blink::WebMediaConstraints constraints_; 167 blink::WebMediaConstraints constraints_;
176 168
177 // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output 169 // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
178 // data is in a unit of 10 ms data chunk. 170 // data is in a unit of 10 ms data chunk.
179 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; 171 const scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
180 172
181 bool running_; 173 bool running_;
182 174
183 int render_frame_id_; 175 int render_frame_id_;
184 176
185 // Cached information of the device used by the capturer. 177 // Cached information of the device used by the capturer.
186 const StreamDeviceInfo device_info_; 178 const StreamDeviceInfo device_info_;
187 179
188 // Stores latest microphone volume received in a CaptureData() callback. 180 // Stores latest microphone volume received in a CaptureData() callback.
189 // Range is [0, 255]. 181 // Range is [0, 255].
190 int volume_; 182 int volume_;
191 183
192 // Flag which affects the buffer size used by the capturer. 184 // Flag which affects the buffer size used by the capturer.
193 bool peer_connection_mode_; 185 bool peer_connection_mode_;
194 186
195 // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime 187 // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
196 // of RenderThread. 188 // of RenderThread.
197 WebRtcAudioDeviceImpl* audio_device_; 189 WebRtcAudioDeviceImpl* audio_device_;
198 190
199 // Raw pointer to the MediaStreamAudioSource object that holds a reference 191 // Raw pointer to the MediaStreamAudioSource object that holds a reference
200 // to this WebRtcAudioCapturer. 192 // to this WebRtcAudioCapturer.
201 // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and 193 // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and
202 // blink guarantees that the blink::WebMediaStreamSource outlives any 194 // blink guarantees that the blink::WebMediaStreamSource outlives any
203 // blink::WebMediaStreamTrack connected to the source, |audio_source_| is 195 // blink::WebMediaStreamTrack connected to the source, |audio_source_| is
204 // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this 196 // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
205 // WebRtcAudioCapturer. 197 // WebRtcAudioCapturer.
206 MediaStreamAudioSource* const audio_source_; 198 MediaStreamAudioSource* const audio_source_;
207 199
200 // Used to calculate the signal level that shows in the UI.
201 MediaStreamAudioLevelCalculator level_calculator_;
202
203 // Provides weak pointers so that the stop callbacks given to
204 // WebRtcLocalAudioTracks can be canceled once they are no longer necessary.
205 base::WeakPtrFactory<WebRtcAudioCapturer> weak_factory_;
206
208 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer); 207 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
209 }; 208 };
210 209
211 } // namespace content 210 } // namespace content
212 211
213 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ 212 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
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