@szdziedzic/sim-on 0.1.0

package/vendor/helper/LiveKitWebRTC.framework/Versions/A/Headers/RTCPeerConnection.h

@@ -0,0 +1,416 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCDataChannel);
+@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
+@class RTC_OBJC_TYPE(RTCIceCandidate);
+@class RTC_OBJC_TYPE(RTCIceCandidateErrorEvent);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+@class RTC_OBJC_TYPE(RTCRtpSender);
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
+@class RTC_OBJC_TYPE(RTCSessionDescription);
+@class RTC_OBJC_TYPE(RTCStatisticsReport);
+@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
+
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCRtpMediaType));
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const RTC_CONSTANT_TYPE(RTCPeerConnectionErrorDomain);
+extern int const RTC_CONSTANT_TYPE(RTCSessionDescriptionErrorCode);
+
+/** Represents the signaling state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCSignalingState)) {
+  RTC_OBJC_TYPE(RTCSignalingStateStable),
+  RTC_OBJC_TYPE(RTCSignalingStateHaveLocalOffer),
+  RTC_OBJC_TYPE(RTCSignalingStateHaveLocalPrAnswer),
+  RTC_OBJC_TYPE(RTCSignalingStateHaveRemoteOffer),
+  RTC_OBJC_TYPE(RTCSignalingStateHaveRemotePrAnswer),
+  // Not an actual state, represents the total number of states.
+  RTC_OBJC_TYPE(RTCSignalingStateClosed),
+};
+
+/** Represents the ice connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCIceConnectionState)) {
+  RTC_OBJC_TYPE(RTCIceConnectionStateNew),
+  RTC_OBJC_TYPE(RTCIceConnectionStateChecking),
+  RTC_OBJC_TYPE(RTCIceConnectionStateConnected),
+  RTC_OBJC_TYPE(RTCIceConnectionStateCompleted),
+  RTC_OBJC_TYPE(RTCIceConnectionStateFailed),
+  RTC_OBJC_TYPE(RTCIceConnectionStateDisconnected),
+  RTC_OBJC_TYPE(RTCIceConnectionStateClosed),
+  RTC_OBJC_TYPE(RTCIceConnectionStateCount),
+};
+
+/** Represents the combined ice+dtls connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCPeerConnectionState)) {
+  RTC_OBJC_TYPE(RTCPeerConnectionStateNew),
+  RTC_OBJC_TYPE(RTCPeerConnectionStateConnecting),
+  RTC_OBJC_TYPE(RTCPeerConnectionStateConnected),
+  RTC_OBJC_TYPE(RTCPeerConnectionStateDisconnected),
+  RTC_OBJC_TYPE(RTCPeerConnectionStateFailed),
+  RTC_OBJC_TYPE(RTCPeerConnectionStateClosed),
+};
+
+/** Represents the ice gathering state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCIceGatheringState)) {
+  RTC_OBJC_TYPE(RTCIceGatheringStateNew),
+  RTC_OBJC_TYPE(RTCIceGatheringStateGathering),
+  RTC_OBJC_TYPE(RTCIceGatheringStateComplete),
+};
+
+/** Represents the stats output level. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCStatsOutputLevel)) {
+  RTC_OBJC_TYPE(RTCStatsOutputLevelStandard),
+  RTC_OBJC_TYPE(RTCStatsOutputLevelDebug),
+};
+
+typedef void (^RTCCreateSessionDescriptionCompletionHandler)(
+    RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp,
+    NSError *_Nullable error);
+
+typedef void (^RTCSetSessionDescriptionCompletionHandler)(
+    NSError *_Nullable error);
+
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate)<NSObject>
+
+    /** Called when the SignalingState changed. */
+    - (void)peerConnection
+    : (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection didChangeSignalingState
+    : (RTC_OBJC_TYPE(RTCSignalingState))stateChanged;
+
+/** Called when media is received on a new stream from remote peer. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+          didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when a remote peer closes a stream.
+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
+ */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+       didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when negotiation is needed, for example ICE has restarted. */
+- (void)peerConnectionShouldNegotiate:
+    (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
+
+/** Called any time the IceConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceConnectionState:(RTC_OBJC_TYPE(RTCIceConnectionState))newState;
+
+/** Called any time the IceGatheringState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceGatheringState:(RTC_OBJC_TYPE(RTCIceGatheringState))newState;
+
+/** New ice candidate has been found. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Called when a group of local Ice candidates have been removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didRemoveIceCandidates:
+        (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** New data channel has been opened. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** Called when signaling indicates a transceiver will be receiving media from
+ *  the remote endpoint.
+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@optional
+/** Called any time the IceConnectionState changes following standardized
+ * transition. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeStandardizedIceConnectionState:(RTC_OBJC_TYPE(RTCIceConnectionState))newState;
+
+/** Called any time the PeerConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeConnectionState:(RTC_OBJC_TYPE(RTCPeerConnectionState))newState;
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didStartReceivingOnTransceiver:
+        (RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
+
+/** Called when a receiver and its track are created. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+        didAddReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
+               streams:(NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)mediaStreams;
+
+/** Called when the receiver and its track are removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+     didRemoveReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver;
+
+/** Called when the selected ICE candidate pair is changed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
+            remoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
+             lastReceivedMs:(int)lastDataReceivedMs
+               changeReason:(NSString *)reason;
+
+/** Called when gathering of an ICE candidate failed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didFailToGatherIceCandidate:
+        (RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnection) : NSObject
+
+/** The object that will be notifed about events such as state changes and
+ *  streams being added or removed.
+ */
+@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
+ *  `senders` instead.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCSignalingState) signalingState;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCIceConnectionState) iceConnectionState;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCPeerConnectionState) connectionState;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCIceGatheringState) iceGatheringState;
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
+
+/** Gets all RTCRtpSenders associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpSender.
+ *  Use isEqual: instead of == to compare RTCRtpSender instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
+
+/** Gets all RTCRtpReceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpReceiver.
+ *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
+ */
+@property(nonatomic, readonly)
+    NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
+
+/** Gets all RTCRtpTransceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of
+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare
+ *  RTCRtpTransceiver instances. This is only available with
+ * RTCSdpSemanticsUnifiedPlan specified.
+ */
+@property(nonatomic, readonly)
+    NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Sets the PeerConnection's global configuration to `configuration`.
+ *  Any changes to STUN/TURN servers or ICE candidate policy will affect the
+ *  next gathering phase, and cause the next call to createOffer to generate
+ *  new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
+ *  cannot be changed with this method.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration;
+
+/** Terminate all media and close the transport. */
+- (void)close;
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
+    DEPRECATED_MSG_ATTRIBUTE(
+        "Please use addIceCandidate:completionHandler: instead");
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
+      completionHandler:(void (^)(NSError *_Nullable error))completionHandler;
+
+/** Remove a group of remote candidates from the ICE Agent. */
+- (void)removeIceCandidates:
+    (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** Add a new media stream to be sent on this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTrack instead.
+ */
+- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Remove the given media stream from this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  removeTrack instead.
+ */
+- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Add a new media stream track to be sent on this peer connection, and return
+ *  the newly created RTCRtpSender. The RTCRtpSender will be
+ * associated with the streams specified in the `streamIds` list.
+ *
+ *  Errors: If an error occurs, returns nil. An error can occur if:
+ *  - A sender already exists for the track.
+ *  - The peer connection is closed.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)
+     addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+    streamIds:(NSArray<NSString *> *)streamIds;
+
+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
+ *
+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
+ *  send component from the associated RTCRtpTransceiver's direction.
+ *
+ *  Returns YES on success.
+ */
+- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
+
+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
+ *  to add a media description for the corresponding transceiver.
+ *
+ *  The initial value of `mid` in the returned transceiver is nil. Setting a
+ *  new session description may change it to a non-nil value.
+ *
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
+ *
+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
+ *  the transceiver from construction. If not specified, the transceiver will
+ *  default to having a direction of kSendRecv and not be part of any streams.
+ *
+ *  These methods are only available when Unified Plan is enabled (see
+ *  RTCConfiguration).
+ */
+
+/** Adds a transceiver with a sender set to transmit the given track. The kind
+ *  of the transceiver (and sender/receiver) will be derived from the kind of
+ *  the track.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
+    (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                       init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
+ *  or RTCRtpMediaTypeVideo.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTC_OBJC_TYPE(RTCRtpMediaType))mediaType;
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverOfType:(RTC_OBJC_TYPE(RTCRtpMediaType))mediaType
+                    init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Tells the PeerConnection that ICE should be restarted. This triggers a need
+ * for negotiation and subsequent offerForConstraints:completionHandler call
+ * will act as if RTCOfferAnswerOptions::ice_restart is true.
+ */
+- (void)restartIce;
+
+/** Generate an SDP offer. */
+- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+          completionHandler:
+              (RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+
+/** Generate an SDP answer. */
+- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+           completionHandler:
+               (RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the local description. */
+- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+          completionHandler:
+              (RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Creates an offer or answer (depending on current signaling state) and sets
+ * it as the local session description. */
+- (void)setLocalDescriptionWithCompletionHandler:
+    (RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the remote description. */
+- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+           completionHandler:
+               (RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Limits the bandwidth allocated for all RTP streams sent by this
+ *  PeerConnection. Nil parameters will be unchanged. Setting
+ * `currentBitrateBps` will force the available bitrate estimate to the given
+ *  value. Returns YES if the parameters were successfully updated.
+ */
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
+
+/** Start or stop recording an Rtc EventLog. */
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
+                      maxSizeInBytes:(int64_t)maxSizeInBytes;
+- (void)stopRtcEventLog;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Media)
+
+    /** Create an RTCRtpSender with the specified kind and media stream ID.
+     *  See RTCMediaStreamTrack.h for available kinds.
+     *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+     *  addTransceiver instead.
+     */
+    - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
+    : (NSString *)streamId;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(DataChannel)
+
+    /** Create a new data channel with the given label and configuration. */
+    - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
+    : (NSString *)label configuration
+    : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
+
+@end
+
+typedef void (^RTCStatisticsCompletionHandler)(
+    RTC_OBJC_TYPE(RTCStatisticsReport) *);
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Stats)
+
+    /** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is
+     * nil statistics are gathered for all tracks.
+     */
+    - (void)statsForTrack
+    : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
+    : (RTC_OBJC_TYPE(RTCStatsOutputLevel))statsOutputLevel completionHandler
+    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
+
+/** Gather statistic through the v2 statistics API. */
+- (void)statisticsWithCompletionHandler:
+    (RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  sender.
+ */
+- (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
+          completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  receiver.
+ */
+- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
+            completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/vendor/helper/LiveKitWebRTC.framework/Versions/A/Headers/RTCPeerConnectionFactory.h

@@ -0,0 +1,159 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+#import <LiveKitWebRTC/RTCAudioDeviceModule.h>
+#import <LiveKitWebRTC/RTCAudioProcessingState.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCRtpCapabilities);
+@class RTC_OBJC_TYPE(RTCAudioSource);
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions);
+@class RTC_OBJC_TYPE(RTCAudioDeviceModule);
+@class RTC_OBJC_TYPE(RTCRtpCapabilities);
+
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCRtpMediaType));
+
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate);
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCSSLCertificateVerifier);
+@protocol RTC_OBJC_TYPE
+(RTCAudioDevice);
+@protocol RTC_OBJC_TYPE
+(RTCAudioProcessingModule);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
+
+/* Initialize object with default H264 video encoder/decoder factories and default ADM */
+- (instancetype)init;
+
+/* Initialize object with injectable video encoder/decoder factories and default ADM */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+
+/* Initialize object with injectable video encoder/decoder factories and injectable ADM */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+               audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
+
+/* Initialize object with bypass voice processing */
+- (instancetype)
+    initWithAudioDeviceModuleType:(RTC_OBJC_TYPE(RTCAudioDeviceModuleType))audioDeviceModuleType
+            bypassVoiceProcessing:(BOOL)bypassVoiceProcessing
+                   encoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+                   decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+            audioProcessingModule:
+                (nullable id<RTC_OBJC_TYPE(RTCAudioProcessingModule)>)audioProcessingModule;
+
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioDeviceModule) *audioDeviceModule;
+
+/** Diagnostic snapshot of the shared audio processing module's state: per
+ *  component, what was requested, what the resolver decided per path, and
+ *  what is actually running. The APM is owned by this factory and shared
+ *  across every peer connection it creates, so this reflects the
+ *  factory-scoped processing state. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioProcessingState) *audioProcessingState;
+
+/**
+ * Valid kind values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:(NSString *)kind;
+
+/**
+ * Valid kind values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:(NSString *)kind;
+
+/** Initialize an RTCAudioSource with constraints. */
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+    (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
+
+/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
+ * with no constraints.
+ */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
+
+/** Initialize an RTCAudioTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize a generic RTCVideoSource. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
+
+/** Initialize a generic RTCVideoSource with he posibility of marking
+ * it as usable for screen sharing. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast;
+
+/** Initialize an RTCVideoTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize an RTCMediaStream with an id. */
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                certificateVerifier:
+                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+/** Set field trials to use when creating the PeerConnectionFactory.
+ *  Must be called before initializing the factory.
+ *  Format: "Key/Value/Key/Value/" (e.g. "WebRTC-MyFeature/Enabled/")
+ */
++ (void)configureFieldTrials:(nullable NSString *)fieldTrials;
+
+/** Set the options to be used for subsequently created RTCPeerConnections */
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
+
+/** Start an AecDump recording. This API call will likely change in the future. */
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+
+/* Stop an active AecDump recording */
+- (void)stopAecDump;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/vendor/helper/LiveKitWebRTC.framework/Versions/A/Headers/RTCPeerConnectionFactoryOptions.h

@@ -0,0 +1,38 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions) : NSObject
+
+@property(nonatomic, assign) BOOL disableEncryption;
+
+@property(nonatomic, assign) BOOL disableNetworkMonitor;
+
+@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/vendor/helper/LiveKitWebRTC.framework/Versions/A/Headers/RTCRtcpParameters.h

@@ -0,0 +1,30 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtcpParameters) : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/vendor/helper/LiveKitWebRTC.framework/Versions/A/Headers/RTCRtpCapabilities.h

@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2024 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 <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCRtpCodecCapability);
+@class RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCapabilities) : NSObject
+
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecCapability) *> *codecs;
+@property(nonatomic, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *> *headerExtensions;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END