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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_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/files/file.h" | 10 #include "base/files/file.h" |
11 #include "content/common/content_export.h" | 11 #include "content/common/content_export.h" |
| 12 #include "content/public/common/media_stream_request.h" |
12 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 13 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
13 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 14 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
14 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h
" | 15 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h
" |
15 | 16 |
16 namespace webrtc { | 17 namespace webrtc { |
17 | 18 |
18 class AudioFrame; | |
19 class EchoCancellation; | 19 class EchoCancellation; |
20 class MediaConstraintsInterface; | 20 class MediaConstraintsInterface; |
21 class TypingDetection; | 21 class TypingDetection; |
22 | 22 |
23 } | 23 } |
24 | 24 |
25 namespace content { | 25 namespace content { |
26 | 26 |
27 class RTCMediaConstraints; | 27 class RTCMediaConstraints; |
28 | 28 |
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132 // Stops the echo cancellation dump in |audio_processing|. | 132 // Stops the echo cancellation dump in |audio_processing|. |
133 // This method has no impact if echo cancellation dump has not been started on | 133 // This method has no impact if echo cancellation dump has not been started on |
134 // |audio_processing|. | 134 // |audio_processing|. |
135 void StopEchoCancellationDump(AudioProcessing* audio_processing); | 135 void StopEchoCancellationDump(AudioProcessing* audio_processing); |
136 | 136 |
137 void EnableAutomaticGainControl(AudioProcessing* audio_processing); | 137 void EnableAutomaticGainControl(AudioProcessing* audio_processing); |
138 | 138 |
139 void GetAecStats(webrtc::EchoCancellation* echo_cancellation, | 139 void GetAecStats(webrtc::EchoCancellation* echo_cancellation, |
140 webrtc::AudioProcessorInterface::AudioProcessorStats* stats); | 140 webrtc::AudioProcessorInterface::AudioProcessorStats* stats); |
141 | 141 |
142 // Parses the microphone array geometry from |geometry_string| formatted as | 142 // Returns the array geometry from the media constraints if existing and |
143 // "x1 y1 z1 ... xn yn zn" for an n-microphone array. See | 143 // otherwise that provided by the input device. |
144 // switches::kMicrophonePositions for more detail. | 144 CONTENT_EXPORT std::vector<webrtc::Point> GetArrayGeometryPreferringConstraints( |
145 // | 145 const MediaAudioConstraints& audio_constraints, |
146 // Returns a zero-sized vector if |geometry_string| isn't a parseable geometry. | 146 const MediaStreamDevice::AudioDeviceParameters& input_params); |
147 CONTENT_EXPORT std::vector<webrtc::Point> ParseArrayGeometry( | |
148 const std::string& geometry_string); | |
149 | 147 |
150 } // namespace content | 148 } // namespace content |
151 | 149 |
152 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ | 150 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ |
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