@recallai/desktop-sdk 2024.1.28-ed4c05a2ec5340a21818e8292a2c8f86359502da

package/WebRTC.framework/Versions/Current/Headers/RTCConfiguration.h

@@ -0,0 +1,262 @@
+/*
+ *  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 <WebRTC/RTCCertificate.h>
+#import <WebRTC/RTCCryptoOptions.h>
+#import <WebRTC/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) {
+  // TODO(https://crbug.com/webrtc/13528): Remove support for Plan B.
+  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 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. By default, this
+ * is RTCSdpSemanticsUnifiedPlan which is compliant to the WebRTC 1.0
+ * specification. It is possible to overrwite this to the deprecated
+ * RTCSdpSemanticsPlanB SDP format, but note that RTCSdpSemanticsPlanB will be
+ * deleted at some future date, see https://crbug.com/webrtc/13528.
+ *
+ * RTCSdpSemanticsUnifiedPlan 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.
+ *
+ * RTCSdpSemanticsPlanB 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.
+ */
+@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;
+
+/**
+ * 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;
+
+/**
+ * An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
+ * which can be used to correlate client logs with backend logs.
+ */
+@property(nonatomic, nullable, copy) NSString *turnLoggingId;
+
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
+/**
+ * Allow implicit rollback of local description when remote description
+ * conflicts with local description.
+ * See: https://w3c.github.io/webrtc-pc/#dom-peerconnection-setremotedescription
+ */
+@property(nonatomic, assign) BOOL enableImplicitRollback;
+
+/**
+ * Control if "a=extmap-allow-mixed" is included in the offer.
+ * See: https://www.chromestatus.com/feature/6269234631933952
+ */
+@property(nonatomic, assign) BOOL offerExtmapAllowMixed;
+
+/**
+ * Defines the interval applied to ALL candidate pairs
+ * when ICE is strongly connected, and it overrides the
+ * default value of this interval in the ICE implementation;
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
+
+/**
+ * Defines the counterpart for ALL pairs when ICE is
+ * weakly connected, and it overrides the default value of
+ * this interval in the ICE implementation
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalWeakConnectivity;
+
+/**
+ * The min time period for which a candidate pair must wait for response to
+ * connectivity checks before it becomes unwritable. This parameter
+ * overrides the default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceUnwritableTimeout;
+
+/**
+ * The min number of connectivity checks that a candidate pair must sent
+ * without receiving response before it becomes unwritable. This parameter
+ * overrides the default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceUnwritableMinChecks;
+
+/**
+ * The min time period for which a candidate pair must wait for response to
+ * connectivity checks it becomes inactive. This parameter overrides the
+ * default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceInactiveTimeout;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/WebRTC.framework/Versions/Current/Headers/RTCCryptoOptions.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/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
+ * kSrtpAes128CmSha1_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

package/WebRTC.framework/Versions/Current/Headers/RTCDataChannel.h

@@ -0,0 +1,132 @@
+/*
+ *  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 <WebRTC/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

package/WebRTC.framework/Versions/Current/Headers/RTCDataChannelConfiguration.h

@@ -0,0 +1,52 @@
+/*
+ *  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 <WebRTC/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

package/WebRTC.framework/Versions/Current/Headers/RTCDefaultVideoDecoderFactory.h

@@ -0,0 +1,26 @@
+/*
+ *  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 <WebRTC/RTCMacros.h>
+#import <WebRTC/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

package/WebRTC.framework/Versions/Current/Headers/RTCDefaultVideoEncoderFactory.h

@@ -0,0 +1,31 @@
+/*
+ *  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 <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCVideoEncoderFactory.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This encoder 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 (RTCDefaultVideoEncoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+
+@property(nonatomic, retain) RTC_OBJC_TYPE(RTCVideoCodecInfo) *preferredCodec;
+
++ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs;
+
+@end
+
+NS_ASSUME_NONNULL_END

package/WebRTC.framework/Versions/Current/Headers/RTCDispatcher.h

@@ -0,0 +1,46 @@
+/*
+ *  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 <WebRTC/RTCMacros.h>
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+  // Main dispatcher queue.
+  RTCDispatcherTypeMain,
+  // Used for starting/stopping AVCaptureSession, and assigning
+  // capture session to AVCaptureVideoPreviewLayer.
+  RTCDispatcherTypeCaptureSession,
+  // Used for operations on AVAudioSession.
+  RTCDispatcherTypeAudioSession,
+  // Used for operations on NWPathMonitor.
+  RTCDispatcherTypeNetworkMonitor,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ *  shared dispatch queue.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDispatcher) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ *  @param dispatchType The queue type to dispatch on.
+ *  @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
+
+/** Returns YES if run on queue for the dispatchType otherwise NO.
+ *  Useful for asserting that a method is run on a correct queue.
+ */
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
+
+@end

package/WebRTC.framework/Versions/Current/Headers/RTCDtmfSender.h

@@ -0,0 +1,71 @@
+/*
+ *  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 <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDtmfSender)<NSObject>
+
+    /**
+     * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
+     * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+     * able to send packets, and a "telephone-event" codec must be negotiated.
+     */
+    @property(nonatomic, readonly) BOOL canInsertDtmf;
+
+/**
+ * Queues a task that sends the DTMF tones. The tones parameter is treated
+ * as a series of characters. The characters 0 through 9, A through D, #, and *
+ * generate the associated DTMF tones. The characters a to d are equivalent
+ * to A to D. The character ',' indicates a delay of 2 seconds before
+ * processing the next character in the tones parameter.
+ *
+ * Unrecognized characters are ignored.
+ *
+ * @param duration The parameter indicates the duration to use for each
+ * character passed in the tones parameter. The duration cannot be more
+ * than 6000 or less than 70 ms.
+ *
+ * @param interToneGap The parameter indicates the gap between tones.
+ * This parameter must be at least 50 ms but should be as short as
+ * possible.
+ *
+ * If InsertDtmf is called on the same object while an existing task for this
+ * object to generate DTMF is still running, the previous task is canceled.
+ * Returns true on success and false on failure.
+ */
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap;
+
+/** The tones remaining to be played out */
+- (nonnull NSString *)remainingTones;
+
+/**
+ * The current tone duration value. This value will be the value last set via the
+ * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ */
+- (NSTimeInterval)duration;
+
+/**
+ * The current value of the between-tone gap. This value will be the value last set
+ * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * called.
+ */
+- (NSTimeInterval)interToneGap;
+
+@end
+
+NS_ASSUME_NONNULL_END