webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
Tomas Gunnarsson 117d847901 Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
This reverts commit 23501a2aa6.

Reason for revert: Breaks downstream features

Original change's description:
> Reland: Remove unsupported configuration value, `allow_codec_switching`
>
> This reverts commit 6b0c5babe0.
>
> Reason for revert: Relanding once downstream issues have been addressed
>
> Original change's description:
> > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> >
> > This reverts commit 8f7a17f80f.
> >
> > Reason for revert: breaks downstream
> >
> > Original change's description:
> > > Remove unsupported configuration value, `allow_codec_switching`
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40995}
> >
> > Bug: webrtc:11341
> > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40998}
>
> Bug: webrtc:11341
> Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41032}

Bug: webrtc:11341
Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41161}
2023-11-15 08:10:28 +00:00

268 lines
9.8 KiB
Objective-C

/*
* 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) {
// 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;
/** 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;
/**
* 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