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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_OPTIONS_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ | 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ |
| 7 | 7 |
| 8 #include <string> | 8 #include <string> |
| 9 | 9 |
| 10 #include "base/platform_file.h" |
| 10 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
| 11 | 12 |
| 12 namespace blink { | 13 namespace blink { |
| 13 class WebMediaConstraints; | 14 class WebMediaConstraints; |
| 14 } | 15 } |
| 15 | 16 |
| 16 namespace webrtc { | 17 namespace webrtc { |
| 17 | 18 |
| 18 class AudioFrame; | 19 class AudioFrame; |
| 19 class AudioProcessing; | 20 class AudioProcessing; |
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| 63 void EnableHighPassFilter(AudioProcessing* audio_processing); | 64 void EnableHighPassFilter(AudioProcessing* audio_processing); |
| 64 | 65 |
| 65 // Enables the typing detection in |audio_processing|. | 66 // Enables the typing detection in |audio_processing|. |
| 66 void EnableTypingDetection(AudioProcessing* audio_processing, | 67 void EnableTypingDetection(AudioProcessing* audio_processing, |
| 67 webrtc::TypingDetection* typing_detector); | 68 webrtc::TypingDetection* typing_detector); |
| 68 | 69 |
| 69 // Enables the experimental echo cancellation in |audio_processing|. | 70 // Enables the experimental echo cancellation in |audio_processing|. |
| 70 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing); | 71 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing); |
| 71 | 72 |
| 72 // Starts the echo cancellation dump in |audio_processing|. | 73 // Starts the echo cancellation dump in |audio_processing|. |
| 73 void StartAecDump(AudioProcessing* audio_processing); | 74 void StartEchoCancellationDump(AudioProcessing* audio_processing, |
| 75 const base::PlatformFile& aec_dump_file); |
| 74 | 76 |
| 75 // Stops the echo cancellation dump in |audio_processing|. | 77 // Stops the echo cancellation dump in |audio_processing|. |
| 76 void StopAecDump(AudioProcessing* audio_processing); | 78 // This method has no impact if echo cancellation dump has not been started on |
| 79 // |audio_processing|. |
| 80 void StopEchoCancellationDump(AudioProcessing* audio_processing); |
| 77 | 81 |
| 78 void EnableAutomaticGainControl(AudioProcessing* audio_processing); | 82 void EnableAutomaticGainControl(AudioProcessing* audio_processing); |
| 79 | 83 |
| 80 void GetAecStats(AudioProcessing* audio_processing, | 84 void GetAecStats(AudioProcessing* audio_processing, |
| 81 webrtc::AudioProcessorInterface::AudioProcessorStats* stats); | 85 webrtc::AudioProcessorInterface::AudioProcessorStats* stats); |
| 82 | 86 |
| 83 } // namespace content | 87 } // namespace content |
| 84 | 88 |
| 85 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ | 89 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ |
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