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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 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 | 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_LOCAL_AUDIO_TRACK_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
7 | 7 |
8 #include <list> | 8 #include <list> |
9 #include <string> | 9 #include <string> |
10 | 10 |
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55 // same sink interface as MediaStreamAudioSink. | 55 // same sink interface as MediaStreamAudioSink. |
56 void AddSink(PeerConnectionAudioSink* sink); | 56 void AddSink(PeerConnectionAudioSink* sink); |
57 void RemoveSink(PeerConnectionAudioSink* sink); | 57 void RemoveSink(PeerConnectionAudioSink* sink); |
58 | 58 |
59 // Starts the local audio track. Called on the main render thread and | 59 // Starts the local audio track. Called on the main render thread and |
60 // should be called only once when audio track is created. | 60 // should be called only once when audio track is created. |
61 void Start(); | 61 void Start(); |
62 | 62 |
63 // Stops the local audio track. Called on the main render thread and | 63 // Stops the local audio track. Called on the main render thread and |
64 // should be called only once when audio track going away. | 64 // should be called only once when audio track going away. |
65 virtual void Stop() override; | 65 void Stop() override; |
66 | 66 |
67 // Method called by the capturer to deliver the capture data. | 67 // Method called by the capturer to deliver the capture data. |
68 // Called on the capture audio thread. | 68 // Called on the capture audio thread. |
69 void Capture(const int16* audio_data, | 69 void Capture(const int16* audio_data, |
70 base::TimeDelta delay, | 70 base::TimeDelta delay, |
71 int volume, | 71 int volume, |
72 bool key_pressed, | 72 bool key_pressed, |
73 bool need_audio_processing, | 73 bool need_audio_processing, |
74 bool force_report_nonzero_energy); | 74 bool force_report_nonzero_energy); |
75 | 75 |
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119 // Used to calculate the signal level that shows in the UI. | 119 // Used to calculate the signal level that shows in the UI. |
120 // Accessed on only the audio thread. | 120 // Accessed on only the audio thread. |
121 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; | 121 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; |
122 | 122 |
123 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 123 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
124 }; | 124 }; |
125 | 125 |
126 } // namespace content | 126 } // namespace content |
127 | 127 |
128 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 128 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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