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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 virtual 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 | 74 |
75 // Method called by the capturer to set the audio parameters used by source | 75 // Method called by the capturer to set the audio parameters used by source |
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118 // Used to calculate the signal level that shows in the UI. | 118 // Used to calculate the signal level that shows in the UI. |
119 // Accessed on only the audio thread. | 119 // Accessed on only the audio thread. |
120 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; | 120 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; |
121 | 121 |
122 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 122 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
123 }; | 123 }; |
124 | 124 |
125 } // namespace content | 125 } // namespace content |
126 | 126 |
127 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 127 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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