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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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