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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/synchronization/lock.h" | 9 #include "base/synchronization/lock.h" |
| 10 #include "base/threading/thread_checker.h" | 10 #include "base/threading/thread_checker.h" |
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| 92 | 92 |
| 93 private: | 93 private: |
| 94 friend class MediaStreamAudioProcessorTest; | 94 friend class MediaStreamAudioProcessorTest; |
| 95 | 95 |
| 96 class MediaStreamAudioConverter; | 96 class MediaStreamAudioConverter; |
| 97 | 97 |
| 98 // WebRtcPlayoutDataSource::Sink implementation. | 98 // WebRtcPlayoutDataSource::Sink implementation. |
| 99 virtual void OnPlayoutData(media::AudioBus* audio_bus, | 99 virtual void OnPlayoutData(media::AudioBus* audio_bus, |
| 100 int sample_rate, | 100 int sample_rate, |
| 101 int audio_delay_milliseconds) OVERRIDE; | 101 int audio_delay_milliseconds) OVERRIDE; |
| 102 virtual void OnPlayoutDataSourceChanged() OVERRIDE; |
| 102 | 103 |
| 103 // webrtc::AudioProcessorInterface implementation. | 104 // webrtc::AudioProcessorInterface implementation. |
| 104 // This method is called on the libjingle thread. | 105 // This method is called on the libjingle thread. |
| 105 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; | 106 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; |
| 106 | 107 |
| 107 // Helper to initialize the WebRtc AudioProcessing. | 108 // Helper to initialize the WebRtc AudioProcessing. |
| 108 void InitializeAudioProcessingModule( | 109 void InitializeAudioProcessingModule( |
| 109 const blink::WebMediaConstraints& constraints, int effects); | 110 const blink::WebMediaConstraints& constraints, int effects); |
| 110 | 111 |
| 111 // Helper to initialize the capture converter. | 112 // Helper to initialize the capture converter. |
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| 172 | 173 |
| 173 // This flag is used to show the result of typing detection. | 174 // This flag is used to show the result of typing detection. |
| 174 // It can be accessed by the capture audio thread and by the libjingle thread | 175 // It can be accessed by the capture audio thread and by the libjingle thread |
| 175 // which calls GetStats(). | 176 // which calls GetStats(). |
| 176 base::subtle::Atomic32 typing_detected_; | 177 base::subtle::Atomic32 typing_detected_; |
| 177 }; | 178 }; |
| 178 | 179 |
| 179 } // namespace content | 180 } // namespace content |
| 180 | 181 |
| 181 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 182 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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