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

Issue 671793004: Clean up the media stream audio track code (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 2 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/files/file.h" 9 #include "base/files/file.h"
10 #include "base/synchronization/lock.h" 10 #include "base/synchronization/lock.h"
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43 43
44 // This class owns an object of webrtc::AudioProcessing which contains signal 44 // This class owns an object of webrtc::AudioProcessing which contains signal
45 // processing components like AGC, AEC and NS. It enables the components based 45 // processing components like AGC, AEC and NS. It enables the components based
46 // on the getUserMedia constraints, processes the data and outputs it in a unit 46 // on the getUserMedia constraints, processes the data and outputs it in a unit
47 // of 10 ms data chunk. 47 // of 10 ms data chunk.
48 class CONTENT_EXPORT MediaStreamAudioProcessor : 48 class CONTENT_EXPORT MediaStreamAudioProcessor :
49 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), 49 NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
50 NON_EXPORTED_BASE(public AudioProcessorInterface), 50 NON_EXPORTED_BASE(public AudioProcessorInterface),
51 NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) { 51 NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
52 public: 52 public:
53 // Returns false if |kDisableAudioTrackProcessing| is set to true, otherwise
54 // returns true.
55 static bool IsAudioTrackProcessingEnabled();
56
57 // |playout_data_source| is used to register this class as a sink to the 53 // |playout_data_source| is used to register this class as a sink to the
58 // WebRtc playout data for processing AEC. If clients do not enable AEC, 54 // WebRtc playout data for processing AEC. If clients do not enable AEC,
59 // |playout_data_source| won't be used. 55 // |playout_data_source| won't be used.
60 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, 56 MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
61 int effects, 57 int effects,
62 WebRtcPlayoutDataSource* playout_data_source); 58 WebRtcPlayoutDataSource* playout_data_source);
63 59
64 // Called when the format of the capture data has changed. 60 // Called when the format of the capture data has changed.
65 // Called on the main render thread. The caller is responsible for stopping 61 // Called on the main render thread. The caller is responsible for stopping
66 // the capture thread before calling this method. 62 // the capture thread before calling this method.
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200 // Communication with browser for AEC dump. 196 // Communication with browser for AEC dump.
201 scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_; 197 scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
202 198
203 // Flag to avoid executing Stop() more than once. 199 // Flag to avoid executing Stop() more than once.
204 bool stopped_; 200 bool stopped_;
205 }; 201 };
206 202
207 } // namespace content 203 } // namespace content
208 204
209 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ 205 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
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