Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(640)

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

Issue 1903663002: Build dynamic iOS SDK. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
11 #if !defined(__has_feature) || !__has_feature(objc_arc) 11 #if !defined(__has_feature) || !__has_feature(objc_arc)
12 #error "This file requires ARC support." 12 #error "This file requires ARC support."
13 #endif 13 #endif
14 14
15 #import <AVFoundation/AVFoundation.h> 15 #import <AVFoundation/AVFoundation.h>
16 #import <Foundation/Foundation.h> 16 #import <Foundation/Foundation.h>
17 17
18 #include "webrtc/modules/audio_device/ios/audio_device_ios.h" 18 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
19 19
20 #include "webrtc/base/atomicops.h" 20 #include "webrtc/base/atomicops.h"
21 #include "webrtc/base/bind.h" 21 #include "webrtc/base/bind.h"
22 #include "webrtc/base/checks.h" 22 #include "webrtc/base/checks.h"
23 #include "webrtc/base/criticalsection.h" 23 #include "webrtc/base/criticalsection.h"
24 #include "webrtc/base/logging.h" 24 #include "webrtc/base/logging.h"
25 #include "webrtc/base/thread.h" 25 #include "webrtc/base/thread.h"
26 #include "webrtc/base/thread_annotations.h" 26 #include "webrtc/base/thread_annotations.h"
27 #include "webrtc/modules/audio_device/fine_audio_buffer.h" 27 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
28 #include "webrtc/modules/utility/include/helpers_ios.h" 28 #include "webrtc/modules/utility/include/helpers_ios.h"
29 29
30 #import "webrtc/base/objc/RTCLogging.h"
31 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h" 30 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
32 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h" 31 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
33 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h" 32 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
34 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h" 33 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
34 #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h"
35 35
36 namespace webrtc { 36 namespace webrtc {
37 37
38 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::" 38 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
39 39
40 #define LOG_AND_RETURN_IF_ERROR(error, message) \ 40 #define LOG_AND_RETURN_IF_ERROR(error, message) \
41 do { \ 41 do { \
42 OSStatus err = error; \ 42 OSStatus err = error; \
43 if (err) { \ 43 if (err) { \
44 LOG(LS_ERROR) << message << ": " << err; \ 44 LOG(LS_ERROR) << message << ": " << err; \
(...skipping 658 matching lines...) Expand 10 before | Expand all | Expand 10 after
703 [session removeDelegate:audio_session_observer_]; 703 [session removeDelegate:audio_session_observer_];
704 704
705 // All I/O should be stopped or paused prior to deactivating the audio 705 // All I/O should be stopped or paused prior to deactivating the audio
706 // session, hence we deactivate as last action. 706 // session, hence we deactivate as last action.
707 [session lockForConfiguration]; 707 [session lockForConfiguration];
708 [session endWebRTCSession:nil]; 708 [session endWebRTCSession:nil];
709 [session unlockForConfiguration]; 709 [session unlockForConfiguration];
710 } 710 }
711 711
712 } // namespace webrtc 712 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698