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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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210 | 210 |
211 // Ensures that methods are called from the same thread as this object is | 211 // Ensures that methods are called from the same thread as this object is |
212 // created on. | 212 // created on. |
213 rtc::ThreadChecker thread_checker_; | 213 rtc::ThreadChecker thread_checker_; |
214 // Thread that this object is created on. | 214 // Thread that this object is created on. |
215 rtc::Thread* thread_; | 215 rtc::Thread* thread_; |
216 // Invoker used to execute methods on thread_. | 216 // Invoker used to execute methods on thread_. |
217 std::unique_ptr<rtc::AsyncInvoker> async_invoker_; | 217 std::unique_ptr<rtc::AsyncInvoker> async_invoker_; |
218 | 218 |
219 // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the | 219 // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the |
220 // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create(). | 220 // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create(). |
221 // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance | 221 // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance |
222 // and therefore outlives this object. | 222 // and therefore outlives this object. |
223 AudioDeviceBuffer* audio_device_buffer_; | 223 AudioDeviceBuffer* audio_device_buffer_; |
224 | 224 |
225 // Contains audio parameters (sample rate, #channels, buffer size etc.) for | 225 // Contains audio parameters (sample rate, #channels, buffer size etc.) for |
226 // the playout and recording sides. These structure is set in two steps: | 226 // the playout and recording sides. These structure is set in two steps: |
227 // first, native sample rate and #channels are defined in Init(). Next, the | 227 // first, native sample rate and #channels are defined in Init(). Next, the |
228 // audio session is activated and we verify that the preferred parameters | 228 // audio session is activated and we verify that the preferred parameters |
229 // were granted by the OS. At this stage it is also possible to add a third | 229 // were granted by the OS. At this stage it is also possible to add a third |
230 // component to the parameters; the native I/O buffer duration. | 230 // component to the parameters; the native I/O buffer duration. |
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282 // Set to true if audio session is interrupted, false otherwise. | 282 // Set to true if audio session is interrupted, false otherwise. |
283 bool is_interrupted_; | 283 bool is_interrupted_; |
284 | 284 |
285 // Audio interruption observer instance. | 285 // Audio interruption observer instance. |
286 RTCAudioSessionDelegateAdapter* audio_session_observer_; | 286 RTCAudioSessionDelegateAdapter* audio_session_observer_; |
287 }; | 287 }; |
288 | 288 |
289 } // namespace webrtc | 289 } // namespace webrtc |
290 | 290 |
291 #endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ | 291 #endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |
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