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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/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
10 #include "base/threading/thread_checker.h" | 11 #include "base/threading/thread_checker.h" |
11 #include "base/time/time.h" | 12 #include "base/time/time.h" |
12 #include "content/common/content_export.h" | 13 #include "content/common/content_export.h" |
13 #include "content/renderer/media/webrtc_audio_device_impl.h" | 14 #include "content/renderer/media/webrtc_audio_device_impl.h" |
14 #include "media/base/audio_converter.h" | 15 #include "media/base/audio_converter.h" |
15 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " | 16 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " |
16 #include "third_party/webrtc/modules/interface/module_common_types.h" | 17 #include "third_party/webrtc/modules/interface/module_common_types.h" |
17 | 18 |
18 namespace blink { | 19 namespace blink { |
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66 // otherwise false. | 67 // otherwise false. |
67 // |capture_delay|, |volume| and |key_pressed| will be passed to | 68 // |capture_delay|, |volume| and |key_pressed| will be passed to |
68 // webrtc::AudioProcessing to help processing the data. | 69 // webrtc::AudioProcessing to help processing the data. |
69 // Called on the capture audio thread. | 70 // Called on the capture audio thread. |
70 bool ProcessAndConsumeData(base::TimeDelta capture_delay, | 71 bool ProcessAndConsumeData(base::TimeDelta capture_delay, |
71 int volume, | 72 int volume, |
72 bool key_pressed, | 73 bool key_pressed, |
73 int* new_volume, | 74 int* new_volume, |
74 int16** out); | 75 int16** out); |
75 | 76 |
76 | |
77 // The audio format of the input to the processor. | 77 // The audio format of the input to the processor. |
78 const media::AudioParameters& InputFormat() const; | 78 const media::AudioParameters& InputFormat() const; |
79 | 79 |
80 // The audio format of the output from the processor. | 80 // The audio format of the output from the processor. |
81 const media::AudioParameters& OutputFormat() const; | 81 const media::AudioParameters& OutputFormat() const; |
82 | 82 |
83 // Accessor to check if the audio processing is enabled or not. | 83 // Accessor to check if the audio processing is enabled or not. |
84 bool has_audio_processing() const { return audio_processing_ != NULL; } | 84 bool has_audio_processing() const { return audio_processing_ != NULL; } |
85 | 85 |
86 // Starts/Stops the Aec dump on the |audio_processing_|. | |
87 // Called on the main render thread. | |
Henrik Grunell
2014/03/06 10:12:20
Comment that it takes ownership of |aec_dump_file|
no longer working on chromium
2014/03/06 18:57:21
Done.
| |
88 void StartAecDump(const base::PlatformFile& aec_dump_file); | |
89 void StopAecDump(); | |
90 | |
86 protected: | 91 protected: |
87 friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>; | 92 friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>; |
88 virtual ~MediaStreamAudioProcessor(); | 93 virtual ~MediaStreamAudioProcessor(); |
89 | 94 |
90 private: | 95 private: |
91 friend class MediaStreamAudioProcessorTest; | 96 friend class MediaStreamAudioProcessorTest; |
92 | 97 |
93 class MediaStreamAudioConverter; | 98 class MediaStreamAudioConverter; |
94 | 99 |
95 // WebRtcPlayoutDataSource::Sink implementation. | 100 // WebRtcPlayoutDataSource::Sink implementation. |
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163 // Used by the typing detection. | 168 // Used by the typing detection. |
164 scoped_ptr<webrtc::TypingDetection> typing_detector_; | 169 scoped_ptr<webrtc::TypingDetection> typing_detector_; |
165 | 170 |
166 // Result from the typing detection. | 171 // Result from the typing detection. |
167 bool typing_detected_; | 172 bool typing_detected_; |
168 }; | 173 }; |
169 | 174 |
170 } // namespace content | 175 } // namespace content |
171 | 176 |
172 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 177 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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