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