M69

WebRTC.framework/Headers/RTCAudioSession.h

@@ -0,0 +1,249 @@
+/*
+ *  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;
+
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTCAudioSession *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTCAudioSession *)audioSession
+    failedToSetActive:(BOOL)active
+                error:(NSError *)error;
+
+@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

WebRTC.framework/Headers/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

WebRTC.framework/Headers/RTCAudioSource.h

@@ -0,0 +1,32 @@
+/*
+ *  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 <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaSource.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCAudioSource : RTCMediaSource
+
+- (instancetype)init NS_UNAVAILABLE;
+
+// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
+// [0, 10].
+// Temporary fix to be able to modify volume of remote audio tracks.
+// TODO(kthelgason): Property stays here temporarily until a proper volume-api
+// is available on the surface exposed by webrtc.
+@property(nonatomic, assign) double volume;
+
+@end
+
+NS_ASSUME_NONNULL_END

WebRTC.framework/Headers/RTCAudioTrack.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 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 <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCAudioSource;
+
+RTC_EXPORT
+@interface RTCAudioTrack : RTCMediaStreamTrack
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** The audio source for this audio track. */
+@property(nonatomic, readonly) RTCAudioSource *source;
+
+@end
+
+NS_ASSUME_NONNULL_END

WebRTC.framework/Headers/RTCCallbackLogger.h

@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2018 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 <Foundation/Foundation.h>
+
+#import <WebRTC/RTCLogging.h>
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_EXPORT
+@interface RTCCallbackLogger : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback will be called on the same thread that does the logging, so
+// if the logging callback can be slow it may be a good idea to implement
+// dispatching to some other queue.
+- (void)start:(nullable void (^)(NSString*))callback;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END