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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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| 63 | 63 |
| 64 private: | 64 private: |
| 65 typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList; | 65 typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList; |
| 66 | 66 |
| 67 // content::WebRtcAudioCapturerSink implementation. | 67 // content::WebRtcAudioCapturerSink implementation. |
| 68 // Called on the AudioInputDevice worker thread. | 68 // Called on the AudioInputDevice worker thread. |
| 69 virtual void CaptureData(const int16* audio_data, | 69 virtual void CaptureData(const int16* audio_data, |
| 70 int number_of_channels, | 70 int number_of_channels, |
| 71 int number_of_frames, | 71 int number_of_frames, |
| 72 int audio_delay_milliseconds, | 72 int audio_delay_milliseconds, |
| 73 double volume) OVERRIDE; | 73 double volume, |
| 74 bool key_pressed) OVERRIDE; |
| 74 | 75 |
| 75 // Can be called on different user threads. | 76 // Can be called on different user threads. |
| 76 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; | 77 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; |
| 77 | 78 |
| 78 // webrtc::AudioTrackInterface implementation. | 79 // webrtc::AudioTrackInterface implementation. |
| 79 virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE; | 80 virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE; |
| 80 | 81 |
| 81 // webrtc::MediaStreamTrack implementation. | 82 // webrtc::MediaStreamTrack implementation. |
| 82 virtual std::string kind() const OVERRIDE; | 83 virtual std::string kind() const OVERRIDE; |
| 83 | 84 |
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| 100 | 101 |
| 101 // Protects |params_| and |sinks_|. | 102 // Protects |params_| and |sinks_|. |
| 102 mutable base::Lock lock_; | 103 mutable base::Lock lock_; |
| 103 | 104 |
| 104 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 105 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
| 105 }; | 106 }; |
| 106 | 107 |
| 107 } // namespace content | 108 } // namespace content |
| 108 | 109 |
| 109 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 110 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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