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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/platform_file.h" |
10 #include "base/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
11 #include "base/threading/thread_checker.h" | 11 #include "base/threading/thread_checker.h" |
12 #include "base/time/time.h" | 12 #include "base/time/time.h" |
13 #include "content/common/content_export.h" | 13 #include "content/common/content_export.h" |
| 14 #include "content/public/common/media_stream_request.h" |
14 #include "content/renderer/media/webrtc_audio_device_impl.h" | 15 #include "content/renderer/media/webrtc_audio_device_impl.h" |
15 #include "media/base/audio_converter.h" | 16 #include "media/base/audio_converter.h" |
16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 17 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
17 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h
" | 18 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h
" |
18 #include "third_party/webrtc/modules/interface/module_common_types.h" | 19 #include "third_party/webrtc/modules/interface/module_common_types.h" |
19 | 20 |
20 namespace blink { | 21 namespace blink { |
21 class WebMediaConstraints; | 22 class WebMediaConstraints; |
22 } | 23 } |
23 | 24 |
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44 // of 10 ms data chunk. | 45 // of 10 ms data chunk. |
45 class CONTENT_EXPORT MediaStreamAudioProcessor : | 46 class CONTENT_EXPORT MediaStreamAudioProcessor : |
46 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), | 47 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), |
47 NON_EXPORTED_BASE(public AudioProcessorInterface) { | 48 NON_EXPORTED_BASE(public AudioProcessorInterface) { |
48 public: | 49 public: |
49 // |playout_data_source| is used to register this class as a sink to the | 50 // |playout_data_source| is used to register this class as a sink to the |
50 // WebRtc playout data for processing AEC. If clients do not enable AEC, | 51 // WebRtc playout data for processing AEC. If clients do not enable AEC, |
51 // |playout_data_source| won't be used. | 52 // |playout_data_source| won't be used. |
52 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, | 53 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, |
53 int effects, | 54 int effects, |
| 55 MediaStreamType type, |
54 WebRtcPlayoutDataSource* playout_data_source); | 56 WebRtcPlayoutDataSource* playout_data_source); |
55 | 57 |
56 // Called when format of the capture data has changed. | 58 // Called when format of the capture data has changed. |
57 // Called on the main render thread. The caller is responsible for stopping | 59 // Called on the main render thread. The caller is responsible for stopping |
58 // the capture thread before calling this method. | 60 // the capture thread before calling this method. |
59 // After this method, the capture thread will be changed to a new capture | 61 // After this method, the capture thread will be changed to a new capture |
60 // thread. | 62 // thread. |
61 void OnCaptureFormatChanged(const media::AudioParameters& source_params); | 63 void OnCaptureFormatChanged(const media::AudioParameters& source_params); |
62 | 64 |
63 // Pushes capture data in |audio_source| to the internal FIFO. | 65 // Pushes capture data in |audio_source| to the internal FIFO. |
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112 int sample_rate, | 114 int sample_rate, |
113 int audio_delay_milliseconds) OVERRIDE; | 115 int audio_delay_milliseconds) OVERRIDE; |
114 virtual void OnPlayoutDataSourceChanged() OVERRIDE; | 116 virtual void OnPlayoutDataSourceChanged() OVERRIDE; |
115 | 117 |
116 // webrtc::AudioProcessorInterface implementation. | 118 // webrtc::AudioProcessorInterface implementation. |
117 // This method is called on the libjingle thread. | 119 // This method is called on the libjingle thread. |
118 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; | 120 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; |
119 | 121 |
120 // Helper to initialize the WebRtc AudioProcessing. | 122 // Helper to initialize the WebRtc AudioProcessing. |
121 void InitializeAudioProcessingModule( | 123 void InitializeAudioProcessingModule( |
122 const blink::WebMediaConstraints& constraints, int effects); | 124 const blink::WebMediaConstraints& constraints, int effects, |
| 125 MediaStreamType type); |
123 | 126 |
124 // Helper to initialize the capture converter. | 127 // Helper to initialize the capture converter. |
125 void InitializeCaptureConverter(const media::AudioParameters& source_params); | 128 void InitializeCaptureConverter(const media::AudioParameters& source_params); |
126 | 129 |
127 // Helper to initialize the render converter. | 130 // Helper to initialize the render converter. |
128 void InitializeRenderConverterIfNeeded(int sample_rate, | 131 void InitializeRenderConverterIfNeeded(int sample_rate, |
129 int number_of_channels, | 132 int number_of_channels, |
130 int frames_per_buffer); | 133 int frames_per_buffer); |
131 | 134 |
132 // Called by ProcessAndConsumeData(). | 135 // Called by ProcessAndConsumeData(). |
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185 | 188 |
186 // This flag is used to show the result of typing detection. | 189 // This flag is used to show the result of typing detection. |
187 // It can be accessed by the capture audio thread and by the libjingle thread | 190 // It can be accessed by the capture audio thread and by the libjingle thread |
188 // which calls GetStats(). | 191 // which calls GetStats(). |
189 base::subtle::Atomic32 typing_detected_; | 192 base::subtle::Atomic32 typing_detected_; |
190 }; | 193 }; |
191 | 194 |
192 } // namespace content | 195 } // namespace content |
193 | 196 |
194 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 197 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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