OLD | NEW |
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; |
(...skipping 43 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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_ |
OLD | NEW |