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Side by Side Diff: webrtc/modules/audio_device/ios/audio_device_ios.h

Issue 1944883002: Move ADM Create() method to public interface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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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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