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Side by Side Diff: content/renderer/media/media_stream_audio_processor.h

Issue 227743004: Added a kEchoCancellation constraint to turn off the audio processing. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: addressed the comments. Created 6 years, 8 months ago
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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"
15 #include "content/renderer/media/webrtc_audio_device_impl.h" 14 #include "content/renderer/media/webrtc_audio_device_impl.h"
16 #include "media/base/audio_converter.h" 15 #include "media/base/audio_converter.h"
17 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
18 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " 17 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h "
19 #include "third_party/webrtc/modules/interface/module_common_types.h" 18 #include "third_party/webrtc/modules/interface/module_common_types.h"
20 19
21 namespace blink { 20 namespace blink {
22 class WebMediaConstraints; 21 class WebMediaConstraints;
23 } 22 }
24 23
(...skipping 20 matching lines...) Expand all
45 // of 10 ms data chunk. 44 // of 10 ms data chunk.
46 class CONTENT_EXPORT MediaStreamAudioProcessor : 45 class CONTENT_EXPORT MediaStreamAudioProcessor :
47 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), 46 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
48 NON_EXPORTED_BASE(public AudioProcessorInterface) { 47 NON_EXPORTED_BASE(public AudioProcessorInterface) {
49 public: 48 public:
50 // |playout_data_source| is used to register this class as a sink to the 49 // |playout_data_source| is used to register this class as a sink to the
51 // WebRtc playout data for processing AEC. If clients do not enable AEC, 50 // WebRtc playout data for processing AEC. If clients do not enable AEC,
52 // |playout_data_source| won't be used. 51 // |playout_data_source| won't be used.
53 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, 52 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
54 int effects, 53 int effects,
55 MediaStreamType type,
56 WebRtcPlayoutDataSource* playout_data_source); 54 WebRtcPlayoutDataSource* playout_data_source);
57 55
58 // Called when format of the capture data has changed. 56 // Called when format of the capture data has changed.
59 // Called on the main render thread. The caller is responsible for stopping 57 // Called on the main render thread. The caller is responsible for stopping
60 // the capture thread before calling this method. 58 // the capture thread before calling this method.
61 // After this method, the capture thread will be changed to a new capture 59 // After this method, the capture thread will be changed to a new capture
62 // thread. 60 // thread.
63 void OnCaptureFormatChanged(const media::AudioParameters& source_params); 61 void OnCaptureFormatChanged(const media::AudioParameters& source_params);
64 62
65 // Pushes capture data in |audio_source| to the internal FIFO. 63 // Pushes capture data in |audio_source| to the internal FIFO.
(...skipping 50 matching lines...) Expand 10 before | Expand all | Expand 10 after
116 int sample_rate, 114 int sample_rate,
117 int audio_delay_milliseconds) OVERRIDE; 115 int audio_delay_milliseconds) OVERRIDE;
118 virtual void OnPlayoutDataSourceChanged() OVERRIDE; 116 virtual void OnPlayoutDataSourceChanged() OVERRIDE;
119 117
120 // webrtc::AudioProcessorInterface implementation. 118 // webrtc::AudioProcessorInterface implementation.
121 // This method is called on the libjingle thread. 119 // This method is called on the libjingle thread.
122 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; 120 virtual void GetStats(AudioProcessorStats* stats) OVERRIDE;
123 121
124 // Helper to initialize the WebRtc AudioProcessing. 122 // Helper to initialize the WebRtc AudioProcessing.
125 void InitializeAudioProcessingModule( 123 void InitializeAudioProcessingModule(
126 const blink::WebMediaConstraints& constraints, int effects, 124 const blink::WebMediaConstraints& constraints, int effects);
127 MediaStreamType type);
128 125
129 // Helper to initialize the capture converter. 126 // Helper to initialize the capture converter.
130 void InitializeCaptureConverter(const media::AudioParameters& source_params); 127 void InitializeCaptureConverter(const media::AudioParameters& source_params);
131 128
132 // Helper to initialize the render converter. 129 // Helper to initialize the render converter.
133 void InitializeRenderConverterIfNeeded(int sample_rate, 130 void InitializeRenderConverterIfNeeded(int sample_rate,
134 int number_of_channels, 131 int number_of_channels,
135 int frames_per_buffer); 132 int frames_per_buffer);
136 133
137 // Called by ProcessAndConsumeData(). 134 // Called by ProcessAndConsumeData().
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
176 // Used to DCHECK that the destructor is called on the main render thread. 173 // Used to DCHECK that the destructor is called on the main render thread.
177 base::ThreadChecker main_thread_checker_; 174 base::ThreadChecker main_thread_checker_;
178 175
179 // Used to DCHECK that some methods are called on the capture audio thread. 176 // Used to DCHECK that some methods are called on the capture audio thread.
180 base::ThreadChecker capture_thread_checker_; 177 base::ThreadChecker capture_thread_checker_;
181 178
182 // Used to DCHECK that PushRenderData() is called on the render audio thread. 179 // Used to DCHECK that PushRenderData() is called on the render audio thread.
183 base::ThreadChecker render_thread_checker_; 180 base::ThreadChecker render_thread_checker_;
184 181
185 // Flag to enable the stereo channels mirroring. 182 // Flag to enable the stereo channels mirroring.
186 bool audio_mirroring_; 183 bool goog_audio_mirroring_;
187 184
188 // Used by the typing detection. 185 // Used by the typing detection.
189 scoped_ptr<webrtc::TypingDetection> typing_detector_; 186 scoped_ptr<webrtc::TypingDetection> typing_detector_;
190 187
191 // This flag is used to show the result of typing detection. 188 // This flag is used to show the result of typing detection.
192 // It can be accessed by the capture audio thread and by the libjingle thread 189 // It can be accessed by the capture audio thread and by the libjingle thread
perkj_chrome 2014/04/11 11:45:04 is this still true? is libjingle thread involved?
no longer working on chromium 2014/04/11 16:47:05 Yes, by GetStats() from libjingle.
193 // which calls GetStats(). 190 // which calls GetStats().
194 base::subtle::Atomic32 typing_detected_; 191 base::subtle::Atomic32 goog_typing_detected_;
195 }; 192 };
196 193
197 } // namespace content 194 } // namespace content
198 195
199 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ 196 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
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