Initial commit

Frameworks/Info.plist

+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-x86_64-simulator</string>
+			<key>LibraryPath</key>
+			<string>JitsiMeetSDK.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>x86_64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+			<key>SupportedPlatformVariant</key>
+			<string>simulator</string>
+		</dict>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64</string>
+			<key>LibraryPath</key>
+			<string>JitsiMeetSDK.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>

Frameworks/WebRTC.xcframework/Info.plist

+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64</string>
+			<key>LibraryPath</key>
+			<string>WebRTC.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>macos-x86_64</string>
+			<key>LibraryPath</key>
+			<string>WebRTC.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>x86_64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>macos</string>
+		</dict>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-x86_64-simulator</string>
+			<key>LibraryPath</key>
+			<string>WebRTC.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>x86_64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+			<key>SupportedPlatformVariant</key>
+			<string>simulator</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioSession.h

+/*
+ *  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 "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 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 RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
+
+    @optional
+/** Called on a system notification thread when AVAudioSession starts an
+ *  interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ *  interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ *  route.
+ */
+- (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:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (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:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (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:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(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_OBJC_EXPORT
+@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 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_OBJC_EXPORT
+@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
+ *  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;
+
+/**
+ When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
+ configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
+ Typically, configurations to preferred attributes are optimizations, and ignoring this type of
+ configuration error allows code flow to continue along the happy path when these optimization are
+ not available. The default value of this property is NO.
+ */
+@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+/** Removes an added 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.
+ */
+- (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 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 : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ *  |lockForConfiguration| must be called first.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h

+/*
+ *  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 "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
+RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (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:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioSource.h

+/*
+ *  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 "RTCMacros.h"
+#import "RTCMediaSource.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(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

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioTrack.h

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

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCVPixelBuffer.h

+/*
+ *  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 <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
+@property(nonatomic, readonly) int cropX;
+@property(nonatomic, readonly) int cropY;
+@property(nonatomic, readonly) int cropWidth;
+@property(nonatomic, readonly) int cropHeight;
+
++ (NSSet<NSNumber *> *)supportedPixelFormats;
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                       adaptedWidth:(int)adaptedWidth
+                      adaptedHeight:(int)adaptedHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY;
+
+- (BOOL)requiresCropping;
+- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
+- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
+
+/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
+ * bufferSizeForCroppingAndScalingToWidth:height: method.
+ * If that size is 0, the |tmpBuffer| may be nil.
+ */
+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
+        withTempBuffer:(nullable uint8_t *)tmpBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCallbackLogger.h

+/*
+ *  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 "RTCLogging.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
+                                                           RTCLoggingSeverity severity);
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback handler 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 RTCCallbackLoggerMessageHandler)handler;
+- (void)startWithMessageAndSeverityHandler:
+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCameraPreviewView.h

+/*
+ *  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 <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+
+@class AVCaptureSession;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ *  AVCaptureSession.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
+
+/** The capture session being rendered in the view. Capture session
+ *  is assigned to AVCaptureVideoPreviewLayer async in the same
+ *  queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession* captureSession;
+
+@end

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCameraVideoCapturer.h

+/*
+ *  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
+ *  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 "RTCMacros.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+// Camera capture that implements RTCVideoCapturer. Delivers frames to a
+// RTCVideoCapturerDelegate (usually RTCVideoSource).
+NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
+@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;
+
+// Returns list of available capture devices that support video capture.
++ (NSArray<AVCaptureDevice *> *)captureDevices;
+// Returns list of formats that are supported by this class for this device.
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
+
+// Returns the most efficient supported output pixel format for this capturer.
+- (FourCharCode)preferredOutputPixelFormat;
+
+// Starts the capture session asynchronously and notifies callback on completion.
+// The device will capture video in the format given in the `format` parameter. If the pixel format
+// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
+// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *))completionHandler;
+// Stops the capture session asynchronously and notifies callback on completion.
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
+
+// Starts the capture session asynchronously.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps;
+// Stops the capture session asynchronously.
+- (void)stopCapture;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCertificate.h

+/*
+ *  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 "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
+
+/** Private key in PEM. */
+@property(nonatomic, readonly, copy) NSString *private_key;
+
+/** Public key in an x509 cert encoded in PEM. */
+@property(nonatomic, readonly, copy) NSString *certificate;
+
+/**
+ * Initialize an RTCCertificate with PEM strings for private_key and certificate.
+ */
+- (instancetype)initWithPrivateKey:(NSString *)private_key
+                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Generate a new certificate for 're' use.
+ *
+ *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
+ *  provided.
+ *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
+ */
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCodecSpecificInfo.h

+/*
+ *  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
+ *  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 "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Implement this protocol to pass codec specific info from the encoder.
+ *  Corresponds to webrtc::CodecSpecificInfo.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCCodecSpecificInfo)<NSObject> @end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h

+/*
+ *  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
+ *  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 "RTCCodecSpecificInfo.h"
+#import "RTCMacros.h"
+
+/** Class for H264 specific config. */
+typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
+  RTCH264PacketizationModeNonInterleaved = 0,  // Mode 1 - STAP-A, FU-A is allowed
+  RTCH264PacketizationModeSingleNalUnit        // Mode 0 - only single NALU allowed
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
+
+@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
+
+@end

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCConfiguration.h

+/*
+ *  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 <Foundation/Foundation.h>
+
+#import "RTCCertificate.h"
+#import "RTCCryptoOptions.h"
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCIceServer);
+
+/**
+ * Represents the ice transport policy. This exposes the same states in C++,
+ * which include one more state than what exists in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
+  RTCIceTransportPolicyNone,
+  RTCIceTransportPolicyRelay,
+  RTCIceTransportPolicyNoHost,
+  RTCIceTransportPolicyAll
+};
+
+/** Represents the bundle policy. */
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  RTCBundlePolicyBalanced,
+  RTCBundlePolicyMaxCompat,
+  RTCBundlePolicyMaxBundle
+};
+
+/** Represents the rtcp mux policy. */
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
+
+/** Represents the tcp candidate policy. */
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  RTCTcpCandidatePolicyEnabled,
+  RTCTcpCandidatePolicyDisabled
+};
+
+/** Represents the candidate network policy. */
+typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
+  RTCCandidateNetworkPolicyAll,
+  RTCCandidateNetworkPolicyLowCost
+};
+
+/** Represents the continual gathering policy. */
+typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
+  RTCContinualGatheringPolicyGatherOnce,
+  RTCContinualGatheringPolicyGatherContinually
+};
+
+/** Represents the encryption key type. */
+typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
+  RTCEncryptionKeyTypeRSA,
+  RTCEncryptionKeyTypeECDSA,
+};
+
+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
+  RTCSdpSemanticsPlanB,
+  RTCSdpSemanticsUnifiedPlan,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@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<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
+
+/** An RTCCertificate for 're' use. */
+@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|. */
+@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
+
+/** The media-bundling policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+
+/** The rtcp-mux policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
+@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
+
+/** If set to YES, don't gather IPv6 ICE candidates.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6;
+
+/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
+ *  Only intended to be used on specific devices. Certain phones disable IPv6
+ *  when the screen is turned off and it would be better to just disable the
+ *  IPv6 ICE candidates on Wi-Fi in those cases.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
+
+/** By default, the PeerConnection will use a limited number of IPv6 network
+ *  interfaces, in order to avoid too many ICE candidate pairs being created
+ *  and delaying ICE completion.
+ *
+ *  Can be set to INT_MAX to effectively disable the limit.
+ */
+@property(nonatomic, assign) int maxIPv6Networks;
+
+/** Exclude link-local network interfaces
+ *  from considertaion for gathering ICE candidates.
+ *  Defaults to NO.
+ */
+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
+
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
+
+/** Key type used to generate SSL identity. Default is ECDSA. */
+@property(nonatomic, assign) RTCEncryptionKeyType keyType;
+
+/** ICE candidate pool size as defined in JSEP. Default is 0. */
+@property(nonatomic, assign) int iceCandidatePoolSize;
+
+/** Prune turn ports on the same network to the same turn server.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
+
+/** If set to YES, this means the ICE transport should presume TURN-to-TURN
+ *  candidate pairs will succeed, even before a binding response is received.
+ */
+@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
+
+/* This flag is only effective when |continualGatheringPolicy| is
+ * RTCContinualGatheringPolicyGatherContinually.
+ *
+ * If YES, after the ICE transport type is changed such that new types of
+ * ICE candidates are allowed by the new transport type, e.g. from
+ * RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
+ * have been gathered by the ICE transport but not matching the previous
+ * transport type and as a result not observed by PeerConnectionDelegateAdapter,
+ * will be surfaced to the delegate.
+ */
+@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+
+/** If set to non-nil, controls the minimal interval between consecutive ICE
+ *  check packets.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
+
+/** Configure the SDP semantics used by this PeerConnection. Note that the
+ *  WebRTC 1.0 specification requires UnifiedPlan semantics. The
+ *  RTCRtpTransceiver API is only available with UnifiedPlan semantics.
+ *
+ *  PlanB will cause RTCPeerConnection to create offers and answers with at
+ *  most one audio and one video m= section with multiple RTCRtpSenders and
+ *  RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
+ *  will also cause RTCPeerConnection to ignore all but the first m= section of
+ *  the same media type.
+ *
+ *  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.
+ *
+ *  For users who wish to send multiple audio/video streams and need to stay
+ *  interoperable with legacy WebRTC implementations or use legacy APIs,
+ *  specify PlanB.
+ *
+ *  For all other users, specify UnifiedPlan.
+ */
+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
+
+/** Actively reset the SRTP parameters when the DTLS transports underneath are
+ *  changed after offer/answer negotiation. This is only intended to be a
+ *  workaround for crbug.com/835958
+ */
+@property(nonatomic, assign) BOOL activeResetSrtpParams;
+
+/** If the remote side support mid-stream codec switches then allow encoder
+ *  switching to be performed.
+ */
+
+@property(nonatomic, assign) BOOL allowCodecSwitching;
+
+/**
+ * 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) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
+
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCryptoOptions.h

+/*
+ *  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 "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
+ * as Objective-C doesn't support nested structures.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
+
+/**
+ * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ * if both sides enable it
+ */
+@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
+/**
+ * If set to true, the (potentially insecure) crypto cipher
+ * SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
+ * during negotiation. It will only be used if both peers support it and no
+ * other ciphers get preferred.
+ */
+@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
+/**
+ * If set to true, encrypted RTP header extensions as defined in RFC 6904
+ * will be negotiated. They will only be used if both peers support them.
+ */
+@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
+
+/**
+ * If set all RtpSenders must have an FrameEncryptor attached to them before
+ * they are allowed to send packets. All RtpReceivers must have a
+ * FrameDecryptor attached to them before they are able to receive packets.
+ */
+@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
+
+/**
+ * Initializes CryptoOptions with all possible options set explicitly. This
+ * is done when converting from a native RTCConfiguration.crypto_options.
+ */
+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDataChannel.h

+/*
+ *  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 <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataBuffer) : NSObject
+
+/** NSData representation of the underlying buffer. */
+@property(nonatomic, readonly) NSData *data;
+
+/** Indicates whether |data| contains UTF-8 or binary data. */
+@property(nonatomic, readonly) BOOL isBinary;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
+ * contains UTF-8 or binary data.
+ */
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCDataChannel);
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDataChannelDelegate)<NSObject>
+
+    /** The data channel state changed. */
+    - (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** The data channel successfully received a data buffer. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
+
+@optional
+/** The data channel's |bufferedAmount| changed. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didChangeBufferedAmount:(uint64_t)amount;
+
+@end
+
+/** Represents the state of the data channel. */
+typedef NS_ENUM(NSInteger, RTCDataChannelState) {
+  RTCDataChannelStateConnecting,
+  RTCDataChannelStateOpen,
+  RTCDataChannelStateClosing,
+  RTCDataChannelStateClosed,
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannel) : NSObject
+
+/**
+ * A label that can be used to distinguish this data channel from other data
+ * channel objects.
+ */
+@property(nonatomic, readonly) NSString *label;
+
+/** Whether the data channel can send messages in unreliable mode. */
+@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
+
+/** Returns whether this data channel is ordered or not. */
+@property(nonatomic, readonly) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
+
+/**
+ * The length of the time window (in milliseconds) during which transmissions
+ * and retransmissions may occur in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
+
+/**
+ * The maximum number of retransmissions that are attempted in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxRetransmits;
+
+/**
+ * The name of the sub-protocol used with this data channel, if any. Otherwise
+ * this returns an empty string.
+ */
+@property(nonatomic, readonly) NSString *protocol;
+
+/**
+ * Returns whether this data channel was negotiated by the application or not.
+ */
+@property(nonatomic, readonly) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
+
+/** The identifier for this data channel. */
+@property(nonatomic, readonly) int channelId;
+
+/** The state of the data channel. */
+@property(nonatomic, readonly) RTCDataChannelState readyState;
+
+/**
+ * The number of bytes of application data that have been queued using
+ * |sendData:| but that have not yet been transmitted to the network.
+ */
+@property(nonatomic, readonly) uint64_t bufferedAmount;
+
+/** The delegate for this data channel. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Closes the data channel. */
+- (void)close;
+
+/** Attempt to send |data| on this data channel's underlying data transport. */
+- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDataChannelConfiguration.h

+/*
+ *  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 <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration) : NSObject
+
+/** Set to YES if ordered delivery is required. */
+@property(nonatomic, assign) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
+
+/**
+ * Max period in milliseconds in which retransmissions will be sent. After this
+ * time, no more retransmissions will be sent. -1 if unset.
+ */
+@property(nonatomic, assign) int maxPacketLifeTime;
+
+/** The max number of retransmissions. -1 if unset. */
+@property(nonatomic, assign) int maxRetransmits;
+
+/** Set to YES if the channel has been externally negotiated and we do not send
+ * an in-band signalling in the form of an "open" message.
+ */
+@property(nonatomic, assign) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
+
+/** The id of the data channel. */
+@property(nonatomic, assign) int channelId;
+
+/** Set by the application and opaque to the WebRTC implementation. */
+@property(nonatomic) NSString* protocol;
+
+@end
+
+NS_ASSUME_NONNULL_END

Frameworks/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h

+/*
+ *  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
+ *  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 "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This decoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
+
+NS_ASSUME_NONNULL_END