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8cb344acfd
This CL adds wrappers for the following PeerConnection native APIs to the Objective C API: - SdpSemantics enum added to the RTCConfiguration - RTCRtpTransceiver - RTCPeerConnection.addTrack - RTCPeerConnection.removeTrack - RTCPeerConnection.addTransceiver - RTCPeerConnection.transceivers Bug: webrtc:8870 Change-Id: I9449df9742a59e90894712dc7749ca30b569d94b Reviewed-on: https://webrtc-review.googlesource.com/54780 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22214}
173 lines
6.1 KiB
Objective-C
173 lines
6.1 KiB
Objective-C
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import <Foundation/Foundation.h>
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#import <WebRTC/RTCMacros.h>
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@class RTCIceServer;
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@class RTCIntervalRange;
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/**
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* Represents the ice transport policy. This exposes the same states in C++,
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* which include one more state than what exists in the W3C spec.
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*/
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typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
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RTCIceTransportPolicyNone,
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RTCIceTransportPolicyRelay,
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RTCIceTransportPolicyNoHost,
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RTCIceTransportPolicyAll
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};
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/** Represents the bundle policy. */
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typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
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RTCBundlePolicyBalanced,
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RTCBundlePolicyMaxCompat,
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RTCBundlePolicyMaxBundle
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};
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/** Represents the rtcp mux policy. */
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typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
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RTCRtcpMuxPolicyNegotiate,
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RTCRtcpMuxPolicyRequire
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};
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/** Represents the tcp candidate policy. */
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typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
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RTCTcpCandidatePolicyEnabled,
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RTCTcpCandidatePolicyDisabled
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};
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/** Represents the candidate network policy. */
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typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
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RTCCandidateNetworkPolicyAll,
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RTCCandidateNetworkPolicyLowCost
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};
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/** Represents the continual gathering policy. */
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typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
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RTCContinualGatheringPolicyGatherOnce,
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RTCContinualGatheringPolicyGatherContinually
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};
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/** Represents the encryption key type. */
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typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
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RTCEncryptionKeyTypeRSA,
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RTCEncryptionKeyTypeECDSA,
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};
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/** Represents the chosen SDP semantics for the RTCPeerConnection. */
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typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
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RTCSdpSemanticsDefault,
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RTCSdpSemanticsPlanB,
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RTCSdpSemanticsUnifiedPlan,
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};
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NS_ASSUME_NONNULL_BEGIN
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RTC_EXPORT
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@interface RTCConfiguration : NSObject
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/** An array of Ice Servers available to be used by ICE. */
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@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
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/** Which candidates the ICE agent is allowed to use. The W3C calls it
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* |iceTransportPolicy|, while in C++ it is called |type|. */
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@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
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/** The media-bundling policy to use when gathering ICE candidates. */
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@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
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/** The rtcp-mux policy to use when gathering ICE candidates. */
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@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
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@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
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@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
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@property(nonatomic, assign)
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RTCContinualGatheringPolicy continualGatheringPolicy;
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/** By default, the PeerConnection will use a limited number of IPv6 network
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* interfaces, in order to avoid too many ICE candidate pairs being created
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* and delaying ICE completion.
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*
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* Can be set to INT_MAX to effectively disable the limit.
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*/
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@property(nonatomic, assign) int maxIPv6Networks;
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/** Exclude link-local network interfaces
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* from considertaion for gathering ICE candidates.
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* Defaults to NO.
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*/
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@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
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@property(nonatomic, assign) int audioJitterBufferMaxPackets;
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@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
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@property(nonatomic, assign) int iceConnectionReceivingTimeout;
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@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
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/** Key type used to generate SSL identity. Default is ECDSA. */
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@property(nonatomic, assign) RTCEncryptionKeyType keyType;
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/** ICE candidate pool size as defined in JSEP. Default is 0. */
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@property(nonatomic, assign) int iceCandidatePoolSize;
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/** Prune turn ports on the same network to the same turn server.
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* Default is NO.
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*/
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@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
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/** If set to YES, this means the ICE transport should presume TURN-to-TURN
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* candidate pairs will succeed, even before a binding response is received.
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*/
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@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
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/** If set to non-nil, controls the minimal interval between consecutive ICE
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* check packets.
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*/
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@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
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/** ICE Periodic Regathering
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* If set, WebRTC will periodically create and propose candidates without
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* starting a new ICE generation. The regathering happens continuously with
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* interval specified in milliseconds by the uniform distribution [a, b].
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*/
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@property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
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/** Configure the SDP semantics used by this PeerConnection. Note that the
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* WebRTC 1.0 specification requires UnifiedPlan semantics. The
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* RTCRtpTransceiver API is only available with UnifiedPlan semantics.
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*
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* PlanB will cause RTCPeerConnection to create offers and answers with at
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* most one audio and one video m= section with multiple RTCRtpSenders and
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* RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
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* will also cause RTCPeerConnection to ignore all but the first m= section of
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* the same media type.
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*
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* UnifiedPlan will cause RTCPeerConnection to create offers and answers with
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* multiple m= sections where each m= section maps to one RTCRtpSender and one
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* RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
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* will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
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* that form a Plan B stream.
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*
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* For users who only send at most one audio and one video track, this
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* choice does not matter and should be left as Default.
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*
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* For users who wish to send multiple audio/video streams and need to stay
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* interoperable with legacy WebRTC implementations, specify PlanB.
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*
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* For users who wish to send multiple audio/video streams and/or wish to
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* use the new RTCRtpTransceiver API, specify UnifiedPlan.
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*/
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@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
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- (instancetype)init;
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@end
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NS_ASSUME_NONNULL_END
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