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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/files/file.h" | 9 #include "base/files/file.h" |
10 #include "base/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
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39 using webrtc::AudioProcessorInterface; | 39 using webrtc::AudioProcessorInterface; |
40 | 40 |
41 // This class owns an object of webrtc::AudioProcessing which contains signal | 41 // This class owns an object of webrtc::AudioProcessing which contains signal |
42 // processing components like AGC, AEC and NS. It enables the components based | 42 // processing components like AGC, AEC and NS. It enables the components based |
43 // on the getUserMedia constraints, processes the data and outputs it in a unit | 43 // on the getUserMedia constraints, processes the data and outputs it in a unit |
44 // of 10 ms data chunk. | 44 // of 10 ms data chunk. |
45 class CONTENT_EXPORT MediaStreamAudioProcessor : | 45 class CONTENT_EXPORT MediaStreamAudioProcessor : |
46 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), | 46 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), |
47 NON_EXPORTED_BASE(public AudioProcessorInterface) { | 47 NON_EXPORTED_BASE(public AudioProcessorInterface) { |
48 public: | 48 public: |
49 // Returns true if |kEnableAudioTrackProcessing| is on or if the | 49 // Returns false if |kDisableAudioTrackProcessing| is set to true, otherwise |
50 // |MediaStreamAudioTrackProcessing| finch experiment is enabled. | 50 // returns true. |
51 static bool IsAudioTrackProcessingEnabled(); | 51 static bool IsAudioTrackProcessingEnabled(); |
52 | 52 |
53 // |playout_data_source| is used to register this class as a sink to the | 53 // |playout_data_source| is used to register this class as a sink to the |
54 // WebRtc playout data for processing AEC. If clients do not enable AEC, | 54 // WebRtc playout data for processing AEC. If clients do not enable AEC, |
55 // |playout_data_source| won't be used. | 55 // |playout_data_source| won't be used. |
56 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, | 56 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, |
57 int effects, | 57 int effects, |
58 WebRtcPlayoutDataSource* playout_data_source); | 58 WebRtcPlayoutDataSource* playout_data_source); |
59 | 59 |
60 // Called when format of the capture data has changed. | 60 // Called when format of the capture data has changed. |
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189 | 189 |
190 // This flag is used to show the result of typing detection. | 190 // 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 | 191 // It can be accessed by the capture audio thread and by the libjingle thread |
192 // which calls GetStats(). | 192 // which calls GetStats(). |
193 base::subtle::Atomic32 typing_detected_; | 193 base::subtle::Atomic32 typing_detected_; |
194 }; | 194 }; |
195 | 195 |
196 } // namespace content | 196 } // namespace content |
197 | 197 |
198 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 198 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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