Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2855023003
Cr-Commit-Position: refs/heads/master@{#18443}
diff --git a/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m b/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
index dca48b6..0e93aae 100644
--- a/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
+++ b/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
@@ -12,10 +12,11 @@
#import <AVFoundation/AVFoundation.h>
+#import "WebRTC/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCDispatcher.h"
#import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+
#import "ARDAppClient.h"
#import "ARDMainView.h"
diff --git a/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m b/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
index 9a9b9c5..fd33e01 100644
--- a/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
+++ b/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
@@ -10,7 +10,7 @@
#import "ARDVideoCallViewController.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSession.h"
#import "ARDAppClient.h"
#import "ARDCaptureController.h"
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index b8d884a..b3ba852 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -176,6 +176,7 @@
public_deps = [
"../../base:gtest_prod",
"../../base:rtc_base",
+ "../../sdk:objc_audio",
"../../sdk:objc_common",
]
sources += [
@@ -183,12 +184,6 @@
"ios/audio_device_ios.mm",
"ios/audio_device_not_implemented_ios.mm",
"ios/audio_session_observer.h",
- "ios/objc/RTCAudioSession+Configuration.mm",
- "ios/objc/RTCAudioSession+Private.h",
- "ios/objc/RTCAudioSession.h",
- "ios/objc/RTCAudioSession.mm",
- "ios/objc/RTCAudioSessionConfiguration.h",
- "ios/objc/RTCAudioSessionConfiguration.m",
"ios/objc/RTCAudioSessionDelegateAdapter.h",
"ios/objc/RTCAudioSessionDelegateAdapter.mm",
"ios/voice_processing_audio_unit.h",
@@ -310,10 +305,7 @@
]
}
if (is_ios) {
- sources += [
- "ios/audio_device_unittest_ios.mm",
- "ios/objc/RTCAudioSessionTest.mm",
- ]
+ sources += [ "ios/audio_device_unittest_ios.mm" ]
deps += [ "//third_party/ocmock" ]
}
if (!build_with_chromium && is_clang) {
diff --git a/webrtc/modules/audio_device/DEPS b/webrtc/modules/audio_device/DEPS
index d7f600c..8ad9db7 100644
--- a/webrtc/modules/audio_device/DEPS
+++ b/webrtc/modules/audio_device/DEPS
@@ -11,4 +11,19 @@
"audio_device_ios\.mm": [
"+webrtc/sdk/objc",
],
+ "audio_device_unittest_ios\.mm": [
+ "+webrtc/sdk/objc",
+ ],
+ "RTCAudioSession\.h": [
+ "+webrtc/sdk/objc",
+ ],
+ "RTCAudioSessionConfiguration\.h": [
+ "+webrtc/sdk/objc",
+ ],
+ "RTCAudioSessionDelegateAdapter\.h": [
+ "+webrtc/sdk/objc",
+ ],
+ "voice_processing_audio_unit\.mm": [
+ "+webrtc/sdk/objc",
+ ],
}
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index 798b543..128ea53 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -27,10 +27,11 @@
#include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
#import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
+#import "webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
+
namespace webrtc {
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
index cccbfbc..2c22c0f 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
@@ -31,8 +31,8 @@
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
+#import "webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
using std::cout;
using std::endl;
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
index c0ea216..2f52ca3 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
+++ b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
@@ -1,5 +1,5 @@
/*
- * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ * Copyright 2017 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -8,235 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import <AVFoundation/AVFoundation.h>
-#import <Foundation/Foundation.h>
-
-#import "WebRTC/RTCMacros.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-extern NSString * const kRTCAudioSessionErrorDomain;
-/** Method that requires lock was called without lock. */
-extern NSInteger const kRTCAudioSessionErrorLockRequired;
-/** Unknown configuration error occurred. */
-extern NSInteger const kRTCAudioSessionErrorConfiguration;
-
-@class RTCAudioSession;
-@class RTCAudioSessionConfiguration;
-
-// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
-// from AVAudioSession and handle them before calling these delegate methods,
-// at which point applications can perform additional processing if required.
-RTC_EXPORT
-@protocol RTCAudioSessionDelegate <NSObject>
-
-@optional
-/** Called on a system notification thread when AVAudioSession starts an
- * interruption event.
- */
-- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
-
-/** Called on a system notification thread when AVAudioSession ends an
- * interruption event.
- */
-- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
- shouldResumeSession:(BOOL)shouldResumeSession;
-
-/** Called on a system notification thread when AVAudioSession changes the
- * route.
- */
-- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
- reason:(AVAudioSessionRouteChangeReason)reason
- previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
-
-/** Called on a system notification thread when AVAudioSession media server
- * terminates.
- */
-- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
-
-/** Called on a system notification thread when AVAudioSession media server
- * restarts.
- */
-- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
-
-// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
-
-- (void)audioSession:(RTCAudioSession *)session
- didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
-
-/** Called on a WebRTC thread when the audio device is notified to begin
- * playback or recording.
- */
-- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
-
-/** Called on a WebRTC thread when the audio device is notified to stop
- * playback or recording.
- */
-- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
-
-/** Called when the AVAudioSession output volume value changes. */
-- (void)audioSession:(RTCAudioSession *)audioSession
- didChangeOutputVolume:(float)outputVolume;
-
-@end
-
-/** This is a protocol used to inform RTCAudioSession when the audio session
- * activation state has changed outside of RTCAudioSession. The current known use
- * case of this is when CallKit activates the audio session for the application
- */
-RTC_EXPORT
-@protocol RTCAudioSessionActivationDelegate <NSObject>
-
-/** Called when the audio session is activated outside of the app by iOS. */
-- (void)audioSessionDidActivate:(AVAudioSession *)session;
-
-/** Called when the audio session is deactivated outside of the app by iOS. */
-- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
-
-@end
-
-/** Proxy class for AVAudioSession that adds a locking mechanism similar to
- * AVCaptureDevice. This is used to that interleaving configurations between
- * WebRTC and the application layer are avoided.
- *
- * RTCAudioSession also coordinates activation so that the audio session is
- * activated only once. See |setActive:error:|.
- */
-RTC_EXPORT
-@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
-
-/** Convenience property to access the AVAudioSession singleton. Callers should
- * not call setters on AVAudioSession directly, but other method invocations
- * are fine.
- */
-@property(nonatomic, readonly) AVAudioSession *session;
-
-/** Our best guess at whether the session is active based on results of calls to
- * AVAudioSession.
- */
-@property(nonatomic, readonly) BOOL isActive;
-/** Whether RTCAudioSession is currently locked for configuration. */
-@property(nonatomic, readonly) BOOL isLocked;
-
-/** If YES, WebRTC will not initialize the audio unit automatically when an
- * audio track is ready for playout or recording. Instead, applications should
- * call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
- * as soon as an audio track is ready for playout or recording.
- */
-@property(nonatomic, assign) BOOL useManualAudio;
-
-/** This property is only effective if useManualAudio is YES.
- * Represents permission for WebRTC to initialize the VoIP audio unit.
- * When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
- * stopped and uninitialized. This will stop incoming and outgoing audio.
- * When set to YES, WebRTC will initialize and start the audio unit when it is
- * needed (e.g. due to establishing an audio connection).
- * This property was introduced to work around an issue where if an AVPlayer is
- * playing audio while the VoIP audio unit is initialized, its audio would be
- * either cut off completely or played at a reduced volume. By preventing
- * the audio unit from being initialized until after the audio has completed,
- * we are able to prevent the abrupt cutoff.
- */
-@property(nonatomic, assign) BOOL isAudioEnabled;
-
-// Proxy properties.
-@property(readonly) NSString *category;
-@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
-@property(readonly) NSString *mode;
-@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
-@property(readonly) AVAudioSessionRouteDescription *currentRoute;
-@property(readonly) NSInteger maximumInputNumberOfChannels;
-@property(readonly) NSInteger maximumOutputNumberOfChannels;
-@property(readonly) float inputGain;
-@property(readonly) BOOL inputGainSettable;
-@property(readonly) BOOL inputAvailable;
-@property(readonly, nullable)
- NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
-@property(readonly, nullable)
- AVAudioSessionDataSourceDescription *inputDataSource;
-@property(readonly, nullable)
- NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
-@property(readonly, nullable)
- AVAudioSessionDataSourceDescription *outputDataSource;
-@property(readonly) double sampleRate;
-@property(readonly) double preferredSampleRate;
-@property(readonly) NSInteger inputNumberOfChannels;
-@property(readonly) NSInteger outputNumberOfChannels;
-@property(readonly) float outputVolume;
-@property(readonly) NSTimeInterval inputLatency;
-@property(readonly) NSTimeInterval outputLatency;
-@property(readonly) NSTimeInterval IOBufferDuration;
-@property(readonly) NSTimeInterval preferredIOBufferDuration;
-
-/** Default constructor. */
-+ (instancetype)sharedInstance;
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Adds a delegate, which is held weakly. */
-- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
-/** Removes an added delegate. */
-- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
-
-/** Request exclusive access to the audio session for configuration. This call
- * will block if the lock is held by another object.
- */
-- (void)lockForConfiguration;
-/** Relinquishes exclusive access to the audio session. */
-- (void)unlockForConfiguration;
-
-/** If |active|, activates the audio session if it isn't already active.
- * Successful calls must be balanced with a setActive:NO when activation is no
- * longer required. If not |active|, deactivates the audio session if one is
- * active and this is the last balanced call. When deactivating, the
- * AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
- * AVAudioSession.
- */
-- (BOOL)setActive:(BOOL)active
- error:(NSError **)outError;
-
-// The following methods are proxies for the associated methods on
-// AVAudioSession. |lockForConfiguration| must be called before using them
-// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
-
-- (BOOL)setCategory:(NSString *)category
- withOptions:(AVAudioSessionCategoryOptions)options
- error:(NSError **)outError;
-- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
-- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
-- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
-- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
- error:(NSError **)outError;
-- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
- error:(NSError **)outError;
-- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
- error:(NSError **)outError;
-- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
- error:(NSError **)outError;
-- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
- error:(NSError **)outError;
-- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
- error:(NSError **)outError;
-- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
- error:(NSError **)outError;
-@end
-
-@interface RTCAudioSession (Configuration)
-
-/** Applies the configuration to the current session. Attempts to set all
- * properties even if previous ones fail. Only the last error will be
- * returned.
- * |lockForConfiguration| must be called first.
- */
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
- error:(NSError **)outError;
-
-/** Convenience method that calls both setConfiguration and setActive.
- * |lockForConfiguration| must be called first.
- */
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
- active:(BOOL)active
- error:(NSError **)outError;
-
-@end
-
-NS_ASSUME_NONNULL_END
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h b/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
index 6a02751..e409108 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
+++ b/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
@@ -1,5 +1,5 @@
/*
- * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ * Copyright 2017 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -8,41 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import <AVFoundation/AVFoundation.h>
-#import <Foundation/Foundation.h>
-
-#import "WebRTC/RTCMacros.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-extern const int kRTCAudioSessionPreferredNumberOfChannels;
-extern const double kRTCAudioSessionHighPerformanceSampleRate;
-extern const double kRTCAudioSessionLowComplexitySampleRate;
-extern const double kRTCAudioSessionHighPerformanceIOBufferDuration;
-extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
-
-// Struct to hold configuration values.
-RTC_EXPORT
-@interface RTCAudioSessionConfiguration : NSObject
-
-@property(nonatomic, strong) NSString *category;
-@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
-@property(nonatomic, strong) NSString *mode;
-@property(nonatomic, assign) double sampleRate;
-@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
-@property(nonatomic, assign) NSInteger inputNumberOfChannels;
-@property(nonatomic, assign) NSInteger outputNumberOfChannels;
-
-/** Initializes configuration to defaults. */
-- (instancetype)init NS_DESIGNATED_INITIALIZER;
-
-/** Returns the current configuration of the audio session. */
-+ (instancetype)currentConfiguration;
-/** Returns the configuration that WebRTC needs. */
-+ (instancetype)webRTCConfiguration;
-/** Provide a way to override the default configuration. */
-+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
-
-@end
-
-NS_ASSUME_NONNULL_END
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h b/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
index 0140aa0..9f71e97 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
+++ b/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
namespace webrtc {
class AudioSessionObserver;
diff --git a/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
index d8805a1..3ad8801 100644
--- a/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
@@ -13,7 +13,7 @@
#include "webrtc/base/checks.h"
#import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
#if !defined(NDEBUG)
static void LogStreamDescription(AudioStreamBasicDescription description) {
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index 34ea001..e1d8e93 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -80,6 +80,27 @@
}
if (!build_with_chromium) {
+ rtc_static_library("objc_audio") {
+ sources = [
+ "objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm",
+ "objc/Framework/Classes/Audio/RTCAudioSession+Private.h",
+ "objc/Framework/Classes/Audio/RTCAudioSession.mm",
+ "objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m",
+ "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
+ "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
+ ]
+ configs += [ "..:common_objc" ]
+
+ deps = [
+ ":objc_common",
+ "../base:rtc_base_approved",
+ ]
+
+ if (is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+ }
rtc_static_library("objc_video") {
sources = [
"objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h",
@@ -370,17 +391,20 @@
"//third_party/ocmock",
]
- # RTCMTLVideoView not supported on 32-bit arm
- if (is_ios && current_cpu != "arm") {
- sources += [ "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm" ]
- if (current_cpu != "arm64") {
- # Only include this file on simulator, as it's already
- # included in device builds.
- sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ]
- libs = [ "CoreVideo.framework" ]
+ if (is_ios) {
+ sources += [ "objc/Framework/UnitTests/RTCAudioSessionTest.mm" ]
+
+ # RTCMTLVideoView not supported on 32-bit arm
+ if (current_cpu != "arm") {
+ sources += [ "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm" ]
+ if (current_cpu != "arm64") {
+ # Only include this file on simulator, as it's already
+ # included in device builds.
+ sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ]
+ libs = [ "CoreVideo.framework" ]
+ }
}
}
-
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -394,6 +418,8 @@
output_name = "WebRTC"
common_objc_headers = [
+ "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
+ "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
"objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
@@ -454,6 +480,7 @@
]
deps = [
+ ":objc_audio",
":objc_peerconnection",
":objc_ui",
"../base:rtc_base_approved",
diff --git a/webrtc/sdk/objc/DEPS b/webrtc/sdk/objc/DEPS
index ac54cc0..eea7e4f 100644
--- a/webrtc/sdk/objc/DEPS
+++ b/webrtc/sdk/objc/DEPS
@@ -15,4 +15,4 @@
"+webrtc/modules/video_coding",
"+webrtc/pc",
"+webrtc/system_wrappers",
-]
+]
\ No newline at end of file
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm
similarity index 96%
rename from webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm
rename to webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm
index 5a7600a..c4d0d0c 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm
@@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "RTCAudioSession+Private.h"
+
@implementation RTCAudioSession (Configuration)
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
similarity index 97%
rename from webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h
rename to webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
index 36be014..5a063ed 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h
+++ b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSession.h"
#include <vector>
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
similarity index 98%
rename from webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
rename to webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
index 763f1fa..ce0e263 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
+++ b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
@@ -8,18 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSession.h"
#import <UIKit/UIKit.h>
#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+
+#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCLogging.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+
+#import "RTCAudioSession+Private.h"
+
NSString * const kRTCAudioSessionErrorDomain = @"org.webrtc.RTCAudioSession";
NSInteger const kRTCAudioSessionErrorLockRequired = -1;
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m
similarity index 97%
rename from webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m
rename to webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m
index 9bbd4b7..fe7b544 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m
+++ b/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "WebRTC/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCDispatcher.h"
#import "WebRTC/UIDevice+RTCDevice.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
// Try to use mono to save resources. Also avoids channel format conversion
// in the I/O audio unit. Initial tests have shown that it is possible to use
diff --git a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
new file mode 100644
index 0000000..c0ea216
--- /dev/null
+++ b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
@@ -0,0 +1,242 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "WebRTC/RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString * const kRTCAudioSessionErrorDomain;
+/** Method that requires lock was called without lock. */
+extern NSInteger const kRTCAudioSessionErrorLockRequired;
+/** Unknown configuration error occurred. */
+extern NSInteger const kRTCAudioSessionErrorConfiguration;
+
+@class RTCAudioSession;
+@class RTCAudioSessionConfiguration;
+
+// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
+// from AVAudioSession and handle them before calling these delegate methods,
+// at which point applications can perform additional processing if required.
+RTC_EXPORT
+@protocol RTCAudioSessionDelegate <NSObject>
+
+@optional
+/** Called on a system notification thread when AVAudioSession starts an
+ * interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ * interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
+ shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ * route.
+ */
+- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
+ reason:(AVAudioSessionRouteChangeReason)reason
+ previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+
+/** Called on a system notification thread when AVAudioSession media server
+ * terminates.
+ */
+- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ * restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (void)audioSession:(RTCAudioSession *)session
+ didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+
+/** Called on a WebRTC thread when the audio device is notified to begin
+ * playback or recording.
+ */
+- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ * playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (void)audioSession:(RTCAudioSession *)audioSession
+ didChangeOutputVolume:(float)outputVolume;
+
+@end
+
+/** This is a protocol used to inform RTCAudioSession when the audio session
+ * activation state has changed outside of RTCAudioSession. The current known use
+ * case of this is when CallKit activates the audio session for the application
+ */
+RTC_EXPORT
+@protocol RTCAudioSessionActivationDelegate <NSObject>
+
+/** Called when the audio session is activated outside of the app by iOS. */
+- (void)audioSessionDidActivate:(AVAudioSession *)session;
+
+/** Called when the audio session is deactivated outside of the app by iOS. */
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
+
+@end
+
+/** Proxy class for AVAudioSession that adds a locking mechanism similar to
+ * AVCaptureDevice. This is used to that interleaving configurations between
+ * WebRTC and the application layer are avoided.
+ *
+ * RTCAudioSession also coordinates activation so that the audio session is
+ * activated only once. See |setActive:error:|.
+ */
+RTC_EXPORT
+@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
+
+/** Convenience property to access the AVAudioSession singleton. Callers should
+ * not call setters on AVAudioSession directly, but other method invocations
+ * are fine.
+ */
+@property(nonatomic, readonly) AVAudioSession *session;
+
+/** Our best guess at whether the session is active based on results of calls to
+ * AVAudioSession.
+ */
+@property(nonatomic, readonly) BOOL isActive;
+/** Whether RTCAudioSession is currently locked for configuration. */
+@property(nonatomic, readonly) BOOL isLocked;
+
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ * audio track is ready for playout or recording. Instead, applications should
+ * call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ * as soon as an audio track is ready for playout or recording.
+ */
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ * Represents permission for WebRTC to initialize the VoIP audio unit.
+ * When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ * stopped and uninitialized. This will stop incoming and outgoing audio.
+ * When set to YES, WebRTC will initialize and start the audio unit when it is
+ * needed (e.g. due to establishing an audio connection).
+ * This property was introduced to work around an issue where if an AVPlayer is
+ * playing audio while the VoIP audio unit is initialized, its audio would be
+ * either cut off completely or played at a reduced volume. By preventing
+ * the audio unit from being initialized until after the audio has completed,
+ * we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
+
+// Proxy properties.
+@property(readonly) NSString *category;
+@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
+@property(readonly) NSString *mode;
+@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
+@property(readonly) AVAudioSessionRouteDescription *currentRoute;
+@property(readonly) NSInteger maximumInputNumberOfChannels;
+@property(readonly) NSInteger maximumOutputNumberOfChannels;
+@property(readonly) float inputGain;
+@property(readonly) BOOL inputGainSettable;
+@property(readonly) BOOL inputAvailable;
+@property(readonly, nullable)
+ NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
+@property(readonly, nullable)
+ AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable)
+ NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
+@property(readonly, nullable)
+ AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
+@property(readonly) NSInteger inputNumberOfChannels;
+@property(readonly) NSInteger outputNumberOfChannels;
+@property(readonly) float outputVolume;
+@property(readonly) NSTimeInterval inputLatency;
+@property(readonly) NSTimeInterval outputLatency;
+@property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
+/** Removes an added delegate. */
+- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
+
+/** Request exclusive access to the audio session for configuration. This call
+ * will block if the lock is held by another object.
+ */
+- (void)lockForConfiguration;
+/** Relinquishes exclusive access to the audio session. */
+- (void)unlockForConfiguration;
+
+/** If |active|, activates the audio session if it isn't already active.
+ * Successful calls must be balanced with a setActive:NO when activation is no
+ * longer required. If not |active|, deactivates the audio session if one is
+ * active and this is the last balanced call. When deactivating, the
+ * AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
+ * AVAudioSession.
+ */
+- (BOOL)setActive:(BOOL)active
+ error:(NSError **)outError;
+
+// The following methods are proxies for the associated methods on
+// AVAudioSession. |lockForConfiguration| must be called before using them
+// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
+
+- (BOOL)setCategory:(NSString *)category
+ withOptions:(AVAudioSessionCategoryOptions)options
+ error:(NSError **)outError;
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
+ error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
+ error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
+ error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
+ error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
+ error:(NSError **)outError;
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+ error:(NSError **)outError;
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+ error:(NSError **)outError;
+@end
+
+@interface RTCAudioSession (Configuration)
+
+/** Applies the configuration to the current session. Attempts to set all
+ * properties even if previous ones fail. Only the last error will be
+ * returned.
+ * |lockForConfiguration| must be called first.
+ */
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+ error:(NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ * |lockForConfiguration| must be called first.
+ */
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+ active:(BOOL)active
+ error:(NSError **)outError;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
new file mode 100644
index 0000000..6a02751
--- /dev/null
+++ b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "WebRTC/RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern const int kRTCAudioSessionPreferredNumberOfChannels;
+extern const double kRTCAudioSessionHighPerformanceSampleRate;
+extern const double kRTCAudioSessionLowComplexitySampleRate;
+extern const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_EXPORT
+@interface RTCAudioSessionConfiguration : NSObject
+
+@property(nonatomic, strong) NSString *category;
+@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
+@property(nonatomic, strong) NSString *mode;
+@property(nonatomic, assign) double sampleRate;
+@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
+@property(nonatomic, assign) NSInteger inputNumberOfChannels;
+@property(nonatomic, assign) NSInteger outputNumberOfChannels;
+
+/** Initializes configuration to defaults. */
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+/** Returns the current configuration of the audio session. */
++ (instancetype)currentConfiguration;
+/** Returns the configuration that WebRTC needs. */
++ (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
similarity index 97%
rename from webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm
rename to webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
index c050e31..f932457 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
@@ -11,11 +11,12 @@
#import <Foundation/Foundation.h>
#import <OCMock/OCMock.h>
-#include "webrtc/test/gtest.h"
+#include "webrtc/base/gunit.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
-#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
+#import "RTCAudioSession+Private.h"
+
+#import "WebRTC/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
@interface RTCAudioSessionTestDelegate : NSObject <RTCAudioSessionDelegate>