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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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114 virtual int32_t StereoRecording(bool* enabled) const { return 0; } | 114 virtual int32_t StereoRecording(bool* enabled) const { return 0; } |
115 virtual int32_t SetRecordingChannel(const ChannelType channel) { return 0; } | 115 virtual int32_t SetRecordingChannel(const ChannelType channel) { return 0; } |
116 virtual int32_t RecordingChannel(ChannelType* channel) const { return 0; } | 116 virtual int32_t RecordingChannel(ChannelType* channel) const { return 0; } |
117 virtual int32_t SetPlayoutBuffer(const BufferType type, | 117 virtual int32_t SetPlayoutBuffer(const BufferType type, |
118 uint16_t sizeMS = 0) { | 118 uint16_t sizeMS = 0) { |
119 return 0; | 119 return 0; |
120 } | 120 } |
121 virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const { | 121 virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const { |
122 return 0; | 122 return 0; |
123 } | 123 } |
124 virtual int32_t PlayoutDelay(uint16_t* delayMS) const { return 0; } | 124 virtual int32_t PlayoutDelay(uint16_t* delayMS) const { |
| 125 *delayMS = 0; |
| 126 return 0; |
| 127 } |
125 virtual int32_t RecordingDelay(uint16_t* delayMS) const { return 0; } | 128 virtual int32_t RecordingDelay(uint16_t* delayMS) const { return 0; } |
126 virtual int32_t CPULoad(uint16_t* load) const { return 0; } | 129 virtual int32_t CPULoad(uint16_t* load) const { return 0; } |
127 virtual int32_t StartRawOutputFileRecording( | 130 virtual int32_t StartRawOutputFileRecording( |
128 const char pcmFileNameUTF8[kAdmMaxFileNameSize]) { | 131 const char pcmFileNameUTF8[kAdmMaxFileNameSize]) { |
129 return 0; | 132 return 0; |
130 } | 133 } |
131 virtual int32_t StopRawOutputFileRecording() { return 0; } | 134 virtual int32_t StopRawOutputFileRecording() { return 0; } |
132 virtual int32_t StartRawInputFileRecording( | 135 virtual int32_t StartRawInputFileRecording( |
133 const char pcmFileNameUTF8[kAdmMaxFileNameSize]) { | 136 const char pcmFileNameUTF8[kAdmMaxFileNameSize]) { |
134 return 0; | 137 return 0; |
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160 } | 163 } |
161 virtual int GetRecordAudioParameters(AudioParameters* params) const { | 164 virtual int GetRecordAudioParameters(AudioParameters* params) const { |
162 return -1; | 165 return -1; |
163 } | 166 } |
164 #endif // WEBRTC_IOS | 167 #endif // WEBRTC_IOS |
165 }; | 168 }; |
166 | 169 |
167 } // namespace webrtc | 170 } // namespace webrtc |
168 | 171 |
169 #endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_ | 172 #endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_ |
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