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Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework.

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BUG=NONE

Review-Url: https://codereview.webrtc.org/2855023003
Cr-Commit-Position: refs/heads/master@{#18443}
This commit is contained in:
denicija
2017-06-05 05:48:47 -07:00
committed by Commit Bot
parent 90d9e10330
commit 59ee91b68a
19 changed files with 378 additions and 316 deletions
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5
webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
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@ -12,10 +12,11 @@
#import <AVFoundation/AVFoundation.h> #import <AVFoundation/AVFoundation.h>
#import "WebRTC/RTCAudioSession.h"
#import "WebRTC/RTCAudioSessionConfiguration.h"
#import "WebRTC/RTCDispatcher.h" #import "WebRTC/RTCDispatcher.h"
#import "WebRTC/RTCLogging.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 "ARDAppClient.h"
#import "ARDMainView.h" #import "ARDMainView.h"

2
webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
View File

@ -10,7 +10,7 @@
#import "ARDVideoCallViewController.h" #import "ARDVideoCallViewController.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h" #import "WebRTC/RTCAudioSession.h"
#import "ARDAppClient.h" #import "ARDAppClient.h"
#import "ARDCaptureController.h" #import "ARDCaptureController.h"

12
webrtc/modules/audio_device/BUILD.gn
View File

@ -176,6 +176,7 @@ rtc_static_library("audio_device") {
public_deps = [ public_deps = [
"../../base:gtest_prod", "../../base:gtest_prod",
"../../base:rtc_base", "../../base:rtc_base",
"../../sdk:objc_audio",
"../../sdk:objc_common", "../../sdk:objc_common",
] ]
sources += [ sources += [
@ -183,12 +184,6 @@ rtc_static_library("audio_device") {
"ios/audio_device_ios.mm", "ios/audio_device_ios.mm",
"ios/audio_device_not_implemented_ios.mm", "ios/audio_device_not_implemented_ios.mm",
"ios/audio_session_observer.h", "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.h",
"ios/objc/RTCAudioSessionDelegateAdapter.mm", "ios/objc/RTCAudioSessionDelegateAdapter.mm",
"ios/voice_processing_audio_unit.h", "ios/voice_processing_audio_unit.h",
@ -310,10 +305,7 @@ if (rtc_include_tests) {
] ]
} }
if (is_ios) { if (is_ios) {
sources += [ sources += [ "ios/audio_device_unittest_ios.mm" ]
"ios/audio_device_unittest_ios.mm",
"ios/objc/RTCAudioSessionTest.mm",
]
deps += [ "//third_party/ocmock" ] deps += [ "//third_party/ocmock" ]
} }
if (!build_with_chromium && is_clang) { if (!build_with_chromium && is_clang) {

15
webrtc/modules/audio_device/DEPS
View File

@ -11,4 +11,19 @@ specific_include_rules = {
"audio_device_ios\.mm": [ "audio_device_ios\.mm": [
"+webrtc/sdk/objc", "+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",
],
} }

7
webrtc/modules/audio_device/ios/audio_device_ios.mm
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@ -27,10 +27,11 @@
#include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h" #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
#import "WebRTC/RTCLogging.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/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 { namespace webrtc {

4
webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
View File

@ -31,8 +31,8 @@
#include "webrtc/test/gtest.h" #include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/fileutils.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h" #import "webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h" #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
using std::cout; using std::cout;
using std::endl; using std::endl;

235
webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
View File

@ -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 * 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 * 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#import <AVFoundation/AVFoundation.h> #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.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

41
webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
View File

@ -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 * 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 * 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#import <AVFoundation/AVFoundation.h> #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.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

2
webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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 { namespace webrtc {
class AudioSessionObserver; class AudioSessionObserver;

2
webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
View File

@ -13,7 +13,7 @@
#include "webrtc/base/checks.h" #include "webrtc/base/checks.h"
#import "WebRTC/RTCLogging.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) #if !defined(NDEBUG)
static void LogStreamDescription(AudioStreamBasicDescription description) { static void LogStreamDescription(AudioStreamBasicDescription description) {

31
webrtc/sdk/BUILD.gn
View File

@ -80,6 +80,27 @@ if (is_ios || is_mac) {
} }
if (!build_with_chromium) { 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") { rtc_static_library("objc_video") {
sources = [ sources = [
"objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h", "objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h",
@ -370,8 +391,11 @@ if (is_ios || is_mac) {
"//third_party/ocmock", "//third_party/ocmock",
] ]
if (is_ios) {
sources += [ "objc/Framework/UnitTests/RTCAudioSessionTest.mm" ]
# RTCMTLVideoView not supported on 32-bit arm # RTCMTLVideoView not supported on 32-bit arm
if (is_ios && current_cpu != "arm") { if (current_cpu != "arm") {
sources += [ "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm" ] sources += [ "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm" ]
if (current_cpu != "arm64") { if (current_cpu != "arm64") {
# Only include this file on simulator, as it's already # Only include this file on simulator, as it's already
@ -380,7 +404,7 @@ if (is_ios || is_mac) {
libs = [ "CoreVideo.framework" ] libs = [ "CoreVideo.framework" ]
} }
} }
}
if (!build_with_chromium && is_clang) { if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@ -394,6 +418,8 @@ if (is_ios || is_mac) {
output_name = "WebRTC" output_name = "WebRTC"
common_objc_headers = [ 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/RTCAVFoundationVideoSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioSource.h", "objc/Framework/Headers/WebRTC/RTCAudioSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioTrack.h", "objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
@ -454,6 +480,7 @@ if (is_ios || is_mac) {
] ]
deps = [ deps = [
":objc_audio",
":objc_peerconnection", ":objc_peerconnection",
":objc_ui", ":objc_ui",
"../base:rtc_base_approved", "../base:rtc_base_approved",

7
webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm → webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm
View File

@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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/RTCLogging.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h" #import "RTCAudioSession+Private.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
@implementation RTCAudioSession (Configuration) @implementation RTCAudioSession (Configuration)

2
webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h → webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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> #include <vector>

10
webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm → webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
View File

@ -8,18 +8,20 @@
* be found in the AUTHORS file in the root of the source tree. * 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> #import <UIKit/UIKit.h>
#include "webrtc/base/atomicops.h" #include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h" #include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.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/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"; NSString * const kRTCAudioSessionErrorDomain = @"org.webrtc.RTCAudioSession";
NSInteger const kRTCAudioSessionErrorLockRequired = -1; NSInteger const kRTCAudioSessionErrorLockRequired = -1;

4
webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m → webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m
View File

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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/RTCDispatcher.h"
#import "WebRTC/UIDevice+RTCDevice.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 // 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 // in the I/O audio unit. Initial tests have shown that it is possible to use

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webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h Normal file
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/*
* 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

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webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h Normal file
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/*
* 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

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webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm → webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
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@ -11,11 +11,12 @@
#import <Foundation/Foundation.h> #import <Foundation/Foundation.h>
#import <OCMock/OCMock.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 "RTCAudioSession+Private.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h" #import "WebRTC/RTCAudioSession.h"
#import "WebRTC/RTCAudioSessionConfiguration.h"
@interface RTCAudioSessionTestDelegate : NSObject <RTCAudioSessionDelegate> @interface RTCAudioSessionTestDelegate : NSObject <RTCAudioSessionDelegate>