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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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101 virtual int32_t SetRecordedBuffer(const void* audio_buffer, | 101 virtual int32_t SetRecordedBuffer(const void* audio_buffer, |
102 size_t samples_per_channel); | 102 size_t samples_per_channel); |
103 int32_t SetCurrentMicLevel(uint32_t level); | 103 int32_t SetCurrentMicLevel(uint32_t level); |
104 virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift); | 104 virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift); |
105 virtual int32_t DeliverRecordedData(); | 105 virtual int32_t DeliverRecordedData(); |
106 uint32_t NewMicLevel() const; | 106 uint32_t NewMicLevel() const; |
107 | 107 |
108 virtual int32_t RequestPlayoutData(size_t samples_per_channel); | 108 virtual int32_t RequestPlayoutData(size_t samples_per_channel); |
109 virtual int32_t GetPlayoutData(void* audio_buffer); | 109 virtual int32_t GetPlayoutData(void* audio_buffer); |
110 | 110 |
111 // TODO(henrika): these methods should not be used and does not contain any | |
112 // valid implementation. Investigate the possibility to either remove them | |
113 // or add a proper implementation if needed. | |
114 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); | |
115 int32_t StopInputFileRecording(); | |
116 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); | |
117 int32_t StopOutputFileRecording(); | |
118 | |
119 int32_t SetTypingStatus(bool typing_status); | 111 int32_t SetTypingStatus(bool typing_status); |
120 | 112 |
121 // Called on iOS where the native audio layer can be interrupted by other | 113 // Called on iOS where the native audio layer can be interrupted by other |
122 // audio applications. This method can then be used to reset internal | 114 // audio applications. This method can then be used to reset internal |
123 // states and detach thread checkers to allow for a new audio session where | 115 // states and detach thread checkers to allow for a new audio session where |
124 // native callbacks may come from a new set of I/O threads. | 116 // native callbacks may come from a new set of I/O threads. |
125 void NativeAudioInterrupted(); | 117 void NativeAudioInterrupted(); |
126 | 118 |
127 private: | 119 private: |
128 // Starts/stops periodic logging of audio stats. | 120 // Starts/stops periodic logging of audio stats. |
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262 // Should *never* be defined in production builds. Only used for testing. | 254 // Should *never* be defined in production builds. Only used for testing. |
263 // When defined, the output signal will be replaced by a sinus tone at 440Hz. | 255 // When defined, the output signal will be replaced by a sinus tone at 440Hz. |
264 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE | 256 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE |
265 double phase_ ACCESS_ON(playout_thread_checker_); | 257 double phase_ ACCESS_ON(playout_thread_checker_); |
266 #endif | 258 #endif |
267 }; | 259 }; |
268 | 260 |
269 } // namespace webrtc | 261 } // namespace webrtc |
270 | 262 |
271 #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ | 263 #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
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