@szdziedzic/sim-on 0.1.0

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

@@ -0,0 +1,58 @@
+/*
+ *  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
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCodecCapability) : NSObject
+
+/** The preferred RTP payload type. */
+@property(nonatomic, readonly, nullable) NSNumber *preferredPayloadType;
+
+/**
+ * The codec MIME subtype. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, readonly) NSString *name;
+
+/**
+ * The media type of this codec. Equivalent to MIME top-level type.
+ *
+ * Valid values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, readonly, nullable) NSNumber *clockRate;
+
+/**
+ * The number of audio channels (mono=1, stereo=2).
+ * Set to null for video codecs.
+ **/
+@property(nonatomic, readonly, nullable) NSNumber *numChannels;
+
+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
+
+/** The MIME type of the codec. */
+@property(nonatomic, readonly) NSString *mimeType;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,74 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCRtxCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCRedCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCUlpfecCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCFlexfecCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCOpusCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCIsacCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCL16CodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCG722CodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCIlbcCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCPcmuCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCPcmaCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCDtmfCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCComfortNoiseCodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCVp8CodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCVp9CodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCH264CodecName);
+RTC_EXTERN const NSString *const RTC_CONSTANT_TYPE(RTCAv1CodecName);
+
+/** Defined in https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodecparameters */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
+
+/** The RTP payload type. */
+@property(nonatomic, assign) int payloadType;
+
+/**
+ * The codec MIME subtype. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, readonly, nonnull) NSString *name;
+
+/**
+ * The media type of this codec. Equivalent to MIME top-level type.
+ *
+ * Valid values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+@property(nonatomic, readonly, nonnull) NSString *kind;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, readonly, nullable) NSNumber *clockRate;
+
+/**
+ * The number of channels (mono=1, stereo=2).
+ * Set to null for video codecs.
+ **/
+@property(nonatomic, readonly, nullable) NSNumber *numChannels;
+
+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
+@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,81 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::Priority. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCPriority)) {
+  RTC_OBJC_TYPE(RTCPriorityVeryLow),
+  RTC_OBJC_TYPE(RTCPriorityLow),
+  RTC_OBJC_TYPE(RTCPriorityMedium),
+  RTC_OBJC_TYPE(RTCPriorityHigh)
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters) : NSObject
+
+/** The idenfifier for the encoding layer. This is used in simulcast. */
+@property(nonatomic, copy, nullable) NSString *rid;
+
+/** Controls whether the encoding is currently transmitted. */
+@property(nonatomic, assign) BOOL isActive;
+
+/** The maximum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
+
+/** The minimum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
+
+/** The maximum framerate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxFramerate;
+
+/** The requested number of temporal layers to use for the encoding, or nil
+ * if the default should be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *numTemporalLayers;
+
+/** Scale the width and height down by this factor for video. If nil,
+ * implementation default scaling factor will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *scaleResolutionDownBy;
+
+/** The SSRC being used by this encoding. */
+@property(nonatomic, readonly, nullable) NSNumber *ssrc;
+
+/** The relative bitrate priority. */
+@property(nonatomic, assign) double bitratePriority;
+
+/** The relative DiffServ Code Point priority. */
+@property(nonatomic, assign) RTC_OBJC_TYPE(RTCPriority) networkPriority;
+
+/** Allow dynamic frame length changes for audio:
+ https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
+ */
+@property(nonatomic, assign) BOOL adaptiveAudioPacketTime;
+
+/** A case-sensitive identifier of the scalability mode to be used for this stream.
+  https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters */
+@property(nonatomic, copy, nullable) NSString *scalabilityMode;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,33 @@
+/*
+ *  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 (RTCRtpHeaderExtension) : NSObject
+
+/** The URI of the RTP header extension, as defined in RFC5285. */
+@property(nonatomic, readonly, copy) NSString *uri;
+
+/** The value put in the RTP packet to identify the header extension. */
+@property(nonatomic, readonly) int id;
+
+/** Whether the header extension is encrypted or not. */
+@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,39 @@
+/*
+ *  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>
+
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCRtpTransceiverDirection));
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtensionCapability) : NSObject
+
+/** The URI of the RTP header extension, as defined in RFC5285. */
+@property(nonatomic, readonly, copy) NSString *uri;
+
+/** The value put in the RTP packet to identify the header extension. */
+@property(nonatomic, readonly, nullable) NSNumber* preferredId;
+
+/** Whether the header extension is encrypted or not. */
+@property(nonatomic, readonly, getter=isPreferredEncrypted)
+    BOOL preferredEncrypted;
+
+/** Direction of the header extension. */
+@property(nonatomic) RTC_OBJC_TYPE(RTCRtpTransceiverDirection) direction;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,63 @@
+/*
+ *  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 <LiveKitWebRTC/RTCRtcpParameters.h>
+#import <LiveKitWebRTC/RTCRtpCodecParameters.h>
+#import <LiveKitWebRTC/RTCRtpEncodingParameters.h>
+#import <LiveKitWebRTC/RTCRtpHeaderExtension.h>
+#import <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::DegradationPreference. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCDegradationPreference)) {
+  RTC_OBJC_TYPE(RTCDegradationPreferenceMaintainFramerateAndResolution),
+  // TODO(webrtc:450044904): Switch downstream projects to
+  // RTCDegradationPreferenceMaintainFramerateAndResolution and remove
+  // RTCDegradationPreferenceDisabled.
+  RTC_OBJC_TYPE(RTCDegradationPreferenceMaintainFramerate),
+  RTC_OBJC_TYPE(RTCDegradationPreferenceMaintainResolution),
+  RTC_OBJC_TYPE(RTCDegradationPreferenceBalanced)
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpParameters) : NSObject
+
+/** A unique identifier for the last set of parameters applied. */
+@property(nonatomic, copy) NSString *transactionId;
+
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCRtcpParameters) * rtcp;
+
+/** An array containing parameters for RTP header extensions. */
+@property(nonatomic, readonly, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
+
+/** The currently active encodings in the order of preference. */
+@property(nonatomic, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
+
+/** The negotiated set of send codecs in order of preference. */
+@property(nonatomic, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
+
+/**
+ * Degradation preference in case of CPU adaptation or constrained bandwidth.
+ * If nil, implementation default degradation preference will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *degradationPreference;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,105 @@
+/*
+ *  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 <LiveKitWebRTC/RTCMediaStreamTrack.h>
+#import <LiveKitWebRTC/RTCRtpParameters.h>
+#import <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents the media type of the RtpReceiver. */
+typedef NS_ENUM(NSInteger, RTC_OBJC_TYPE(RTCRtpMediaType)) {
+  RTC_OBJC_TYPE(RTCRtpMediaTypeAudio),
+  RTC_OBJC_TYPE(RTCRtpMediaTypeVideo),
+  RTC_OBJC_TYPE(RTCRtpMediaTypeData),
+  RTC_OBJC_TYPE(RTCRtpMediaTypeUnsupported),
+  RTC_OBJC_TYPE(RTCRtpMediaTypeAny),
+};
+
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+@class RTC_OBJC_TYPE(RTCRtpSource);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiverDelegate)<NSObject>
+
+    /** Called when the first RTP packet is received.
+     *
+     *  Note: Currently if there are multiple RtpReceivers of the same media
+     * type, they will all call OnFirstPacketReceived at once.
+     *
+     *  For example, if we create three audio receivers, A/B/C, they will listen
+     * to the same signal from the underneath network layer. Whenever the first
+     * audio packet is received, the underneath signal will be fired. All the
+     * receivers A/B/C will be notified and the callback of the receiver's
+     * delegate will be called.
+     *
+     *  The process is the same for video receivers.
+     */
+    - (void)rtpReceiver : (RTC_OBJC_TYPE(RTCRtpReceiver) *)
+                              rtpReceiver didReceiveFirstPacketForMediaType
+    : (RTC_OBJC_TYPE(RTCRtpMediaType))mediaType;
+/** Called when the first RTP packet is received after a change in
+ * receptiveness.
+ */
+// TODO: crbug.com/40821064 - remove @optional.
+@optional
+- (void)rtpReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
+    didReceiveFirstPacketForMediaTypeAfterReceptiveChange:
+        (RTC_OBJC_TYPE(RTCRtpMediaType))mediaType;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiver)<NSObject>
+
+    /** A unique identifier for this receiver. */
+    @property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) *
+    track;
+
+/**
+Returns an array that contains an object for each unique SSRC (synchronization
+source) identifier and for each unique CSRC (contributing source) received by
+the current RTCRtpReceiver in the last ten seconds.
+*/
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSource) *> *sources;
+
+/** The delegate for this RtpReceiver. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,55 @@
+/*
+ *  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 <LiveKitWebRTC/RTCDtmfSender.h>
+#import <LiveKitWebRTC/RTCMediaStreamTrack.h>
+#import <LiveKitWebRTC/RTCRtpParameters.h>
+#import <LiveKitWebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpSender)<NSObject>
+
+    /** A unique identifier for this sender. */
+    @property(nonatomic, readonly) NSString *senderId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ */
+@property(nonatomic, copy) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the sender.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, copy, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** IDs of streams associated with the RTP sender */
+@property(nonatomic, copy) NSArray<NSString *> *streamIds;
+
+/** The RTCDtmfSender accociated with the RTP sender. */
+@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)>
+    dtmfSender;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

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

@@ -0,0 +1,66 @@
+/*
+ *  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
+
+/** Represents the source type of received media. */
+typedef NS_ENUM(NSInteger, RTCRtpSourceType) {
+  RTCRtpSourceTypeSSRC,
+  RTCRtpSourceTypeCSRC,
+};
+
+@class RTC_OBJC_TYPE(RTCRtpSource);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpSource)<NSObject>
+
+    /**
+    A positive integer value specifying the CSRC identifier of the contributing
+    source or SSRC identifier of the synchronization source. This uniquely
+    identifies the source of the particular stream RTP packets. */
+    @property(nonatomic, readonly) uint32_t sourceId;
+
+@property(nonatomic, readonly) RTCRtpSourceType sourceType;
+
+/**
+A floating-point value between 0.0 and 1.0 specifying the audio level contained
+in the last RTP packet played from the contributing source.
+*/
+@property(nonatomic, readonly, nullable) NSNumber *audioLevel;
+
+/**
+A timestamp indicating the most recent time at which a frame originating from
+this source was delivered to the receiver's track
+*/
+@property(nonatomic, readonly) CFTimeInterval timestampUs;
+
+/**
+The RTP timestamp of the media. This source-generated timestamp indicates the
+time at which the media in this packet, scheduled for play out at the time
+indicated by timestamp, was initially sampled or generated. It may be useful for
+sequencing and synchronization purposes.
+*/
+@property(nonatomic, readonly) uint32_t rtpTimestamp;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpSource) : NSObject <RTC_OBJC_TYPE(RTCRtpSource)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END