85.0.4183.4

WebRTC.framework/Headers/RTCAudioSession.h

@@ -21,78 +21,81 @@ extern NSInteger const kRTCAudioSessionErrorLockRequired;
 /** Unknown configuration error occurred. */
 extern NSInteger const kRTCAudioSessionErrorConfiguration;
 
-@class RTCAudioSession;
-@class RTCAudioSessionConfiguration;
+@class RTC_OBJC_TYPE(RTCAudioSession);
+@class RTC_OBJC_TYPE(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_OBJC_EXPORT
-@protocol RTCAudioSessionDelegate <NSObject>
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
 
-@optional
+    @optional
 /** Called on a system notification thread when AVAudioSession starts an
  *  interruption event.
  */
-- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
 
 /** Called on a system notification thread when AVAudioSession ends an
  *  interruption event.
  */
-- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
                    shouldResumeSession:(BOOL)shouldResumeSession;
 
 /** Called on a system notification thread when AVAudioSession changes the
  *  route.
  */
-- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
                             reason:(AVAudioSessionRouteChangeReason)reason
                      previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
 
 /** Called on a system notification thread when AVAudioSession media server
  *  terminates.
  */
-- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
 
 /** Called on a system notification thread when AVAudioSession media server
  *  restarts.
  */
-- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
 
 // TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
 
-- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)audioSession:(RTC_OBJC_TYPE(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;
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
 
 /** Called on a WebRTC thread when the audio device is notified to stop
  *  playback or recording.
  */
-- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
 
 /** Called when the AVAudioSession output volume value changes. */
-- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
+- (void)audioSession:(RTC_OBJC_TYPE(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
+- (void)audioSession:(RTC_OBJC_TYPE(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;
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
 
 /** Called after the audio session sucessfully changed the active state.
  */
-- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
 
 /** Called after the audio session failed to change the active state.
  */
-- (void)audioSession:(RTCAudioSession *)audioSession
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
     failedToSetActive:(BOOL)active
                 error:(NSError *)error;
 
@@ -103,10 +106,11 @@ RTC_OBJC_EXPORT
  *  case of this is when CallKit activates the audio session for the application
  */
 RTC_OBJC_EXPORT
-@protocol RTCAudioSessionActivationDelegate <NSObject>
+@protocol RTC_OBJC_TYPE
+(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 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;
@@ -121,7 +125,7 @@ RTC_OBJC_EXPORT
  *  activated only once. See |setActive:error:|.
  */
 RTC_OBJC_EXPORT
-@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
+@interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
 
 /** Convenience property to access the AVAudioSession singleton. Callers should
  *  not call setters on AVAudioSession directly, but other method invocations
@@ -196,9 +200,9 @@ RTC_OBJC_EXPORT
 - (instancetype)init NS_UNAVAILABLE;
 
 /** Adds a delegate, which is held weakly. */
-- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
 /** Removes an added delegate. */
-- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
 
 /** Request exclusive access to the audio session for configuration. This call
  *  will block if the lock is held by another object.
@@ -237,19 +241,21 @@ RTC_OBJC_EXPORT
                       error:(NSError **)outError;
 @end
 
-@interface RTCAudioSession (Configuration)
+@interface RTC_OBJC_TYPE (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;
+    /** 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 : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
 
 /** Convenience method that calls both setConfiguration and setActive.
  *  |lockForConfiguration| must be called first.
  */
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
                   active:(BOOL)active
                    error:(NSError **)outError;
 

WebRTC.framework/Headers/RTCAudioSessionConfiguration.h

@@ -23,7 +23,7 @@ RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
 
 // Struct to hold configuration values.
 RTC_OBJC_EXPORT
-@interface RTCAudioSessionConfiguration : NSObject
+@interface RTC_OBJC_TYPE (RTCAudioSessionConfiguration) : NSObject
 
 @property(nonatomic, strong) NSString *category;
 @property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
@@ -41,7 +41,7 @@ RTC_OBJC_EXPORT
 /** Returns the configuration that WebRTC needs. */
 + (instancetype)webRTCConfiguration;
 /** Provide a way to override the default configuration. */
-+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
 
 @end
 

WebRTC.framework/Headers/RTCAudioSource.h

@@ -16,7 +16,7 @@
 NS_ASSUME_NONNULL_BEGIN
 
 RTC_OBJC_EXPORT
-@interface RTCAudioSource : RTCMediaSource
+@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(RTCMediaSource)
 
 - (instancetype)init NS_UNAVAILABLE;
 

WebRTC.framework/Headers/RTCAudioTrack.h

@@ -13,15 +13,15 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@class RTCAudioSource;
+@class RTC_OBJC_TYPE(RTCAudioSource);
 
 RTC_OBJC_EXPORT
-@interface RTCAudioTrack : RTCMediaStreamTrack
+@interface RTC_OBJC_TYPE (RTCAudioTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
 
 - (instancetype)init NS_UNAVAILABLE;
 
 /** The audio source for this audio track. */
-@property(nonatomic, readonly) RTCAudioSource *source;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioSource) * source;
 
 @end
 

WebRTC.framework/Headers/RTCCVPixelBuffer.h

@@ -17,7 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
 
 /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
 RTC_OBJC_EXPORT
-@interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
+@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
 
 @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
 @property(nonatomic, readonly) int cropX;

WebRTC.framework/Headers/RTCCallbackLogger.h

@@ -22,7 +22,7 @@ typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
 // This class intercepts WebRTC logs and forwards them to a registered block.
 // This class is not threadsafe.
 RTC_OBJC_EXPORT
-@interface RTCCallbackLogger : NSObject
+@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
 
 // The severity level to capture. The default is kRTCLoggingSeverityInfo.
 @property(nonatomic, assign) RTCLoggingSeverity severity;

WebRTC.framework/Headers/RTCCameraPreviewView.h

@@ -19,7 +19,7 @@
  *  AVCaptureSession.
  */
 RTC_OBJC_EXPORT
-@interface RTCCameraPreviewView : UIView
+@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
 
 /** The capture session being rendered in the view. Capture session
  *  is assigned to AVCaptureVideoPreviewLayer async in the same

WebRTC.framework/Headers/RTCCameraVideoCapturer.h

@@ -17,10 +17,10 @@
 NS_ASSUME_NONNULL_BEGIN
 
 RTC_OBJC_EXPORT
-// Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
-// (usually RTCVideoSource).
+// Camera capture that implements RTCVideoCapturer. Delivers frames to a
+// RTCVideoCapturerDelegate (usually RTCVideoSource).
 NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
-@interface RTCCameraVideoCapturer : RTCVideoCapturer
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
 
 // Capture session that is used for capturing. Valid from initialization to dealloc.
 @property(readonly, nonatomic) AVCaptureSession *captureSession;

WebRTC.framework/Headers/RTCCertificate.h

@@ -15,7 +15,7 @@
 NS_ASSUME_NONNULL_BEGIN
 
 RTC_OBJC_EXPORT
-@interface RTCCertificate : NSObject <NSCopying>
+@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
 
 /** Private key in PEM. */
 @property(nonatomic, readonly, copy) NSString *private_key;
@@ -37,7 +37,7 @@ RTC_OBJC_EXPORT
  *  provided.
  *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
  */
-+ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params;
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
 
 @end
 

WebRTC.framework/Headers/RTCCodecSpecificInfo.h

@@ -18,7 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
  *  Corresponds to webrtc::CodecSpecificInfo.
  */
 RTC_OBJC_EXPORT
-@protocol RTCCodecSpecificInfo <NSObject>
-@end
+@protocol RTC_OBJC_TYPE
+(RTCCodecSpecificInfo)<NSObject> @end
 
 NS_ASSUME_NONNULL_END

WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h

@@ -20,7 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
 };
 
 RTC_OBJC_EXPORT
-@interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
 
 @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
 

WebRTC.framework/Headers/RTCConfiguration.h

@@ -14,7 +14,7 @@
 #import "RTCCryptoOptions.h"
 #import "RTCMacros.h"
 
-@class RTCIceServer;
+@class RTC_OBJC_TYPE(RTCIceServer);
 
 /**
  * Represents the ice transport policy. This exposes the same states in C++,
@@ -70,18 +70,18 @@ typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
 NS_ASSUME_NONNULL_BEGIN
 
 RTC_OBJC_EXPORT
-@interface RTCConfiguration : NSObject
+@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
 
 /** If true, allows DSCP codes to be set on outgoing packets, configured using
  *  networkPriority field of RTCRtpEncodingParameters. Defaults to false.
  */
 @property(nonatomic, assign) BOOL enableDscp;
 
 /** An array of Ice Servers available to be used by ICE. */
-@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
 
 /** An RTCCertificate for 're' use. */
-@property(nonatomic, nullable) RTCCertificate *certificate;
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
 
 /** Which candidates the ICE agent is allowed to use. The W3C calls it
  * |iceTransportPolicy|, while in C++ it is called |type|. */
@@ -173,9 +173,9 @@ RTC_OBJC_EXPORT
  *
  *  UnifiedPlan will cause RTCPeerConnection to create offers and answers with
  *  multiple m= sections where each m= section maps to one RTCRtpSender and one
- *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
- *  will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
- *  that form a Plan B stream.
+ *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both
+ *  video. This will also cause RTCPeerConnection) to ignore all but the first a=ssrc
+ *  lines that form a Plan B stream.
  *
  *  For users who wish to send multiple audio/video streams and need to stay
  *  interoperable with legacy WebRTC implementations or use legacy APIs,
@@ -197,24 +197,12 @@ RTC_OBJC_EXPORT
 
 @property(nonatomic, assign) BOOL allowCodecSwitching;
 
-/**
- * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
- * that it should use the MediaTransportInterface.
- */
-@property(nonatomic, assign) BOOL useMediaTransport;
-
-/**
- * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
- * that it should use the MediaTransportInterface for data channels.
- */
-@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
-
 /**
  * Defines advanced optional cryptographic settings related to SRTP and
  * frame encryption for native WebRTC. Setting this will overwrite any
  * options set through the PeerConnectionFactory (which is deprecated).
  */
-@property(nonatomic, nullable) RTCCryptoOptions *cryptoOptions;
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
 
 /**
  * Time interval between audio RTCP reports.

WebRTC.framework/Headers/RTCCryptoOptions.h

@@ -19,7 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
  * as Objective-C doesn't support nested structures.
  */
 RTC_OBJC_EXPORT
-@interface RTCCryptoOptions : NSObject
+@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
 
 /**
  * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used