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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_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| 7 | 7 |
| 8 #include "base/atomicops.h" | 8 #include "base/atomicops.h" |
| 9 #include "base/platform_file.h" | 9 #include "base/platform_file.h" |
| 10 #include "base/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
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| 108 | 108 |
| 109 private: | 109 private: |
| 110 friend class MediaStreamAudioProcessorTest; | 110 friend class MediaStreamAudioProcessorTest; |
| 111 | 111 |
| 112 class MediaStreamAudioConverter; | 112 class MediaStreamAudioConverter; |
| 113 | 113 |
| 114 // WebRtcPlayoutDataSource::Sink implementation. | 114 // WebRtcPlayoutDataSource::Sink implementation. |
| 115 virtual void OnPlayoutData(media::AudioBus* audio_bus, | 115 virtual void OnPlayoutData(media::AudioBus* audio_bus, |
| 116 int sample_rate, | 116 int sample_rate, |
| 117 int audio_delay_milliseconds) OVERRIDE; | 117 int audio_delay_milliseconds) OVERRIDE; |
| 118 virtual void OnPlayoutDataSourceChanged() OVERRIDE; |
| 118 | 119 |
| 119 // webrtc::AudioProcessorInterface implementation. | 120 // webrtc::AudioProcessorInterface implementation. |
| 120 // This method is called on the libjingle thread. | 121 // This method is called on the libjingle thread. |
| 121 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; | 122 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; |
| 122 | 123 |
| 123 // Helper to initialize the WebRtc AudioProcessing. | 124 // Helper to initialize the WebRtc AudioProcessing. |
| 124 void InitializeAudioProcessingModule( | 125 void InitializeAudioProcessingModule( |
| 125 const blink::WebMediaConstraints& constraints, int effects, | 126 const blink::WebMediaConstraints& constraints, int effects, |
| 126 MediaStreamType type); | 127 MediaStreamType type); |
| 127 | 128 |
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| 189 | 190 |
| 190 // This flag is used to show the result of typing detection. | 191 // This flag is used to show the result of typing detection. |
| 191 // It can be accessed by the capture audio thread and by the libjingle thread | 192 // It can be accessed by the capture audio thread and by the libjingle thread |
| 192 // which calls GetStats(). | 193 // which calls GetStats(). |
| 193 base::subtle::Atomic32 typing_detected_; | 194 base::subtle::Atomic32 typing_detected_; |
| 194 }; | 195 }; |
| 195 | 196 |
| 196 } // namespace content | 197 } // namespace content |
| 197 | 198 |
| 198 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 199 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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