Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework.
BUG=NONE Review-Url: https://codereview.webrtc.org/2855023003 Cr-Commit-Position: refs/heads/master@{#18443}
This commit is contained in:
denicija
@ -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"
|
||||
|
||||
@ -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"
|
||||
|
||||
@ -176,6 +176,7 @@ rtc_static_library("audio_device") {
|
||||
public_deps = [
|
||||
"../../base:gtest_prod",
|
||||
"../../base:rtc_base",
|
||||
"../../sdk:objc_audio",
|
||||
"../../sdk:objc_common",
|
||||
]
|
||||
sources += [
|
||||
@ -183,12 +184,6 @@ rtc_static_library("audio_device") {
|
||||
"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 (rtc_include_tests) {
|
||||
]
|
||||
}
|
||||
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) {
|
||||
|
||||
@ -11,4 +11,19 @@ specific_include_rules = {
|
||||
"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",
|
||||
],
|
||||
}
|
||||
|
||||
@ -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 {
|
||||
|
||||
|
||||
@ -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;
|
||||
|
||||
@ -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"
|
||||
|
||||
@ -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"
|
||||
|
||||
@ -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;
|
||||
|
||||
@ -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) {
|
||||
|
||||
@ -80,6 +80,27 @@ if (is_ios || is_mac) {
|
||||
}
|
||||
|
||||
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 @@ if (is_ios || is_mac) {
|
||||
"//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 @@ if (is_ios || is_mac) {
|
||||
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 @@ if (is_ios || is_mac) {
|
||||
]
|
||||
|
||||
deps = [
|
||||
":objc_audio",
|
||||
":objc_peerconnection",
|
||||
":objc_ui",
|
||||
"../base:rtc_base_approved",
|
||||
|
||||
@ -15,4 +15,4 @@ include_rules = [
|
||||
"+webrtc/modules/video_coding",
|
||||
"+webrtc/pc",
|
||||
"+webrtc/system_wrappers",
|
||||
]
|
||||
]
|
||||
@ -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)
|
||||
|
||||
@ -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>
|
||||
|
||||
@ -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;
|
||||
@ -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
|
||||
242
webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
Normal file
242
webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
Normal file
@ -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
|
||||
@ -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
|
||||
@ -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>
|
||||
|
||||
Reference in New Issue
Block a user