Removal dependency on OpenSSL using SwiftPM and replaced OpenSSL and WebRTC frameworks dependency with custom dependency update script #siskinim-256
Andrzej Wójcik
3 years ago
parent
611890393b
commit
39025c3624
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/*
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* Copyright 2016 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 <AVFoundation/AVFoundation.h>
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#import <Foundation/Foundation.h>
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#import "RTCMacros.h"
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NS_ASSUME_NONNULL_BEGIN
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extern NSString *const kRTCAudioSessionErrorDomain;
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/** Method that requires lock was called without lock. */
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extern NSInteger const kRTCAudioSessionErrorLockRequired;
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/** Unknown configuration error occurred. */
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extern NSInteger const kRTCAudioSessionErrorConfiguration;
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@class RTCAudioSession;
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@class RTCAudioSessionConfiguration;
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// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
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// from AVAudioSession and handle them before calling these delegate methods,
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// at which point applications can perform additional processing if required.
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RTC_OBJC_EXPORT
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@protocol RTCAudioSessionDelegate <NSObject>
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@optional
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/** Called on a system notification thread when AVAudioSession starts an
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* interruption event.
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*/
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- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
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/** Called on a system notification thread when AVAudioSession ends an
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* interruption event.
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*/
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- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
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shouldResumeSession:(BOOL)shouldResumeSession;
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/** Called on a system notification thread when AVAudioSession changes the
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* route.
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*/
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- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
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reason:(AVAudioSessionRouteChangeReason)reason
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previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
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/** Called on a system notification thread when AVAudioSession media server
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* terminates.
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*/
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- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
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/** Called on a system notification thread when AVAudioSession media server
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* restarts.
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*/
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- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
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// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
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- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
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/** Called on a WebRTC thread when the audio device is notified to begin
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* playback or recording.
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*/
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- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
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/** Called on a WebRTC thread when the audio device is notified to stop
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* playback or recording.
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*/
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- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
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/** Called when the AVAudioSession output volume value changes. */
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- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
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/** Called when the audio device detects a playout glitch. The argument is the
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* number of glitches detected so far in the current audio playout session.
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*/
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- (void)audioSession:(RTCAudioSession *)audioSession
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didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
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/** Called when the audio session is about to change the active state.
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*/
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- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
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/** Called after the audio session sucessfully changed the active state.
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*/
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- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
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/** Called after the audio session failed to change the active state.
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*/
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- (void)audioSession:(RTCAudioSession *)audioSession
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failedToSetActive:(BOOL)active
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error:(NSError *)error;
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@end
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/** This is a protocol used to inform RTCAudioSession when the audio session
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* activation state has changed outside of RTCAudioSession. The current known use
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* case of this is when CallKit activates the audio session for the application
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*/
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RTC_OBJC_EXPORT
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@protocol RTCAudioSessionActivationDelegate <NSObject>
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/** Called when the audio session is activated outside of the app by iOS. */
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- (void)audioSessionDidActivate:(AVAudioSession *)session;
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/** Called when the audio session is deactivated outside of the app by iOS. */
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- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
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@end
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/** Proxy class for AVAudioSession that adds a locking mechanism similar to
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* AVCaptureDevice. This is used to that interleaving configurations between
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* WebRTC and the application layer are avoided.
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*
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* RTCAudioSession also coordinates activation so that the audio session is
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* activated only once. See |setActive:error:|.
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*/
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RTC_OBJC_EXPORT
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@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
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/** Convenience property to access the AVAudioSession singleton. Callers should
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* not call setters on AVAudioSession directly, but other method invocations
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* are fine.
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*/
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@property(nonatomic, readonly) AVAudioSession *session;
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/** Our best guess at whether the session is active based on results of calls to
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* AVAudioSession.
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*/
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@property(nonatomic, readonly) BOOL isActive;
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/** Whether RTCAudioSession is currently locked for configuration. */
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@property(nonatomic, readonly) BOOL isLocked;
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/** If YES, WebRTC will not initialize the audio unit automatically when an
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* audio track is ready for playout or recording. Instead, applications should
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* call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
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* as soon as an audio track is ready for playout or recording.
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*/
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@property(nonatomic, assign) BOOL useManualAudio;
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/** This property is only effective if useManualAudio is YES.
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* Represents permission for WebRTC to initialize the VoIP audio unit.
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* When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
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* stopped and uninitialized. This will stop incoming and outgoing audio.
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* When set to YES, WebRTC will initialize and start the audio unit when it is
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* needed (e.g. due to establishing an audio connection).
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* This property was introduced to work around an issue where if an AVPlayer is
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* playing audio while the VoIP audio unit is initialized, its audio would be
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* either cut off completely or played at a reduced volume. By preventing
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* the audio unit from being initialized until after the audio has completed,
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* we are able to prevent the abrupt cutoff.
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*/
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@property(nonatomic, assign) BOOL isAudioEnabled;
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// Proxy properties.
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@property(readonly) NSString *category;
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@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
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@property(readonly) NSString *mode;
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@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
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@property(readonly) AVAudioSessionRouteDescription *currentRoute;
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@property(readonly) NSInteger maximumInputNumberOfChannels;
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@property(readonly) NSInteger maximumOutputNumberOfChannels;
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@property(readonly) float inputGain;
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@property(readonly) BOOL inputGainSettable;
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@property(readonly) BOOL inputAvailable;
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@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
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@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
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@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
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@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
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@property(readonly) double sampleRate;
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@property(readonly) double preferredSampleRate;
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@property(readonly) NSInteger inputNumberOfChannels;
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@property(readonly) NSInteger outputNumberOfChannels;
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@property(readonly) float outputVolume;
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@property(readonly) NSTimeInterval inputLatency;
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@property(readonly) NSTimeInterval outputLatency;
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@property(readonly) NSTimeInterval IOBufferDuration;
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@property(readonly) NSTimeInterval preferredIOBufferDuration;
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/**
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When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
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configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
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Typically, configurations to preferred attributes are optimizations, and ignoring this type of
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configuration error allows code flow to continue along the happy path when these optimization are
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not available. The default value of this property is NO.
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*/
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@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
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/** Default constructor. */
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+ (instancetype)sharedInstance;
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- (instancetype)init NS_UNAVAILABLE;
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/** Adds a delegate, which is held weakly. */
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- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
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/** Removes an added delegate. */
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- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
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/** Request exclusive access to the audio session for configuration. This call
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* will block if the lock is held by another object.
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*/
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- (void)lockForConfiguration;
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/** Relinquishes exclusive access to the audio session. */
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- (void)unlockForConfiguration;
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/** If |active|, activates the audio session if it isn't already active.
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* Successful calls must be balanced with a setActive:NO when activation is no
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* longer required. If not |active|, deactivates the audio session if one is
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* active and this is the last balanced call. When deactivating, the
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* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
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* AVAudioSession.
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*/
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- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
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// The following methods are proxies for the associated methods on
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// AVAudioSession. |lockForConfiguration| must be called before using them
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// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
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- (BOOL)setCategory:(NSString *)category
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withOptions:(AVAudioSessionCategoryOptions)options
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error:(NSError **)outError;
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- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
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- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
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- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
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- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
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- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
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- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
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- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
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- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
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- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
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error:(NSError **)outError;
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- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
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error:(NSError **)outError;
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@end
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@interface RTCAudioSession (Configuration)
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/** Applies the configuration to the current session. Attempts to set all
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* properties even if previous ones fail. Only the last error will be
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* returned.
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* |lockForConfiguration| must be called first.
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*/
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- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
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/** Convenience method that calls both setConfiguration and setActive.
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* |lockForConfiguration| must be called first.
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*/
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- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
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active:(BOOL)active
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error:(NSError **)outError;
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@end
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NS_ASSUME_NONNULL_END
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@ -1,48 +0,0 @@
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/*
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* Copyright 2016 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 <AVFoundation/AVFoundation.h>
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#import <Foundation/Foundation.h>
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#import "RTCMacros.h"
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NS_ASSUME_NONNULL_BEGIN
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RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
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RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
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RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
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RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
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RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
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// Struct to hold configuration values.
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RTC_OBJC_EXPORT
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@interface RTCAudioSessionConfiguration : NSObject
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@property(nonatomic, strong) NSString *category;
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@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
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@property(nonatomic, strong) NSString *mode;
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@property(nonatomic, assign) double sampleRate;
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@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
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@property(nonatomic, assign) NSInteger inputNumberOfChannels;
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@property(nonatomic, assign) NSInteger outputNumberOfChannels;
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/** Initializes configuration to defaults. */
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- (instancetype)init NS_DESIGNATED_INITIALIZER;
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/** Returns the current configuration of the audio session. */
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+ (instancetype)currentConfiguration;
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/** Returns the configuration that WebRTC needs. */
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+ (instancetype)webRTCConfiguration;
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/** Provide a way to override the default configuration. */
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+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
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@end
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NS_ASSUME_NONNULL_END
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@ -1,32 +0,0 @@
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/*
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* Copyright 2016 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 "RTCMacros.h"
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#import "RTCMediaSource.h"
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NS_ASSUME_NONNULL_BEGIN
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RTC_OBJC_EXPORT
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@interface RTCAudioSource : RTCMediaSource
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- (instancetype)init NS_UNAVAILABLE;
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// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
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// [0, 10].
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// Temporary fix to be able to modify volume of remote audio tracks.
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// TODO(kthelgason): Property stays here temporarily until a proper volume-api
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// is available on the surface exposed by webrtc.
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@property(nonatomic, assign) double volume;
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@end
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NS_ASSUME_NONNULL_END
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@ -1,28 +0,0 @@
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/*
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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 "RTCMacros.h"
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#import "RTCMediaStreamTrack.h"
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NS_ASSUME_NONNULL_BEGIN
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@class RTCAudioSource;
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RTC_OBJC_EXPORT
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@interface RTCAudioTrack : RTCMediaStreamTrack
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- (instancetype)init NS_UNAVAILABLE;
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/** The audio source for this audio track. */
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@property(nonatomic, readonly) RTCAudioSource *source;
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@end
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NS_ASSUME_NONNULL_END
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@ -1,52 +0,0 @@
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/*
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* Copyright 2018 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
|
||||
* 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 <AVFoundation/AVFoundation.h>
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#import "RTCMacros.h"
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#import "RTCVideoFrameBuffer.h"
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NS_ASSUME_NONNULL_BEGIN
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/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
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RTC_OBJC_EXPORT
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@interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
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@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
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@property(nonatomic, readonly) int cropX;
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@property(nonatomic, readonly) int cropY;
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@property(nonatomic, readonly) int cropWidth;
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@property(nonatomic, readonly) int cropHeight;
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+ (NSSet<NSNumber *> *)supportedPixelFormats;
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- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
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- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
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adaptedWidth:(int)adaptedWidth
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adaptedHeight:(int)adaptedHeight
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cropWidth:(int)cropWidth
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cropHeight:(int)cropHeight
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cropX:(int)cropX
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cropY:(int)cropY;
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- (BOOL)requiresCropping;
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- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
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- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
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/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
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* bufferSizeForCroppingAndScalingToWidth:height: method.
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* If that size is 0, the |tmpBuffer| may be nil.
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*/
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- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
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withTempBuffer:(nullable uint8_t *)tmpBuffer;
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@end
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NS_ASSUME_NONNULL_END
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@ -1,41 +0,0 @@
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/*
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* Copyright 2018 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
|
||||
* 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.
|
||||
*/
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#import <Foundation/Foundation.h>
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#import "RTCLogging.h"
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#import "RTCMacros.h"
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NS_ASSUME_NONNULL_BEGIN
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typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
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typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
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RTCLoggingSeverity severity);
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// This class intercepts WebRTC logs and forwards them to a registered block.
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// This class is not threadsafe.
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RTC_OBJC_EXPORT
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@interface RTCCallbackLogger : NSObject
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|
||||
// The severity level to capture. The default is kRTCLoggingSeverityInfo.
|
||||
@property(nonatomic, assign) RTCLoggingSeverity severity;
|
||||
|
||||
// The callback handler will be called on the same thread that does the
|
||||
// logging, so if the logging callback can be slow it may be a good idea
|
||||
// to implement dispatching to some other queue.
|
||||
- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
|
||||
- (void)startWithMessageAndSeverityHandler:
|
||||
(nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
|
||||
|
||||
- (void)stop;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,30 +0,0 @@
|
||||
/*
|
||||
* 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 <UIKit/UIKit.h>
|
||||
|
||||
#import "RTCMacros.h"
|
||||
|
||||
@class AVCaptureSession;
|
||||
|
||||
/** RTCCameraPreviewView is a view that renders local video from an
|
||||
* AVCaptureSession.
|
||||
*/
|
||||
RTC_OBJC_EXPORT
|
||||
@interface RTCCameraPreviewView : UIView
|
||||
|
||||
/** The capture session being rendered in the view. Capture session
|
||||
* is assigned to AVCaptureVideoPreviewLayer async in the same
|
||||
* queue that the AVCaptureSession is started/stopped.
|
||||
*/
|
||||
@property(nonatomic, strong) AVCaptureSession* captureSession;
|
||||
|
||||
@end
|
||||
@ -1,56 +0,0 @@
|
||||
/*
|
||||
* 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 <AVFoundation/AVFoundation.h>
|
||||
#import <Foundation/Foundation.h>
|
||||
|
||||
#import "RTCMacros.h"
|
||||
#import "RTCVideoCapturer.h"
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
// Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
|
||||
// (usually RTCVideoSource).
|
||||
NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
|
||||
@interface RTCCameraVideoCapturer : RTCVideoCapturer
|
||||
|
||||
// Capture session that is used for capturing. Valid from initialization to dealloc.
|
||||
@property(readonly, nonatomic) AVCaptureSession *captureSession;
|
||||
|
||||
// Returns list of available capture devices that support video capture.
|
||||
+ (NSArray<AVCaptureDevice *> *)captureDevices;
|
||||
// Returns list of formats that are supported by this class for this device.
|
||||
+ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
|
||||
|
||||
// Returns the most efficient supported output pixel format for this capturer.
|
||||
- (FourCharCode)preferredOutputPixelFormat;
|
||||
|
||||
// Starts the capture session asynchronously and notifies callback on completion.
|
||||
// The device will capture video in the format given in the `format` parameter. If the pixel format
|
||||
// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
|
||||
// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
|
||||
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
|
||||
format:(AVCaptureDeviceFormat *)format
|
||||
fps:(NSInteger)fps
|
||||
completionHandler:(nullable void (^)(NSError *))completionHandler;
|
||||
// Stops the capture session asynchronously and notifies callback on completion.
|
||||
- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
|
||||
|
||||
// Starts the capture session asynchronously.
|
||||
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
|
||||
format:(AVCaptureDeviceFormat *)format
|
||||
fps:(NSInteger)fps;
|
||||
// Stops the capture session asynchronously.
|
||||
- (void)stopCapture;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,44 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
@interface RTCCertificate : NSObject <NSCopying>
|
||||
|
||||
/** Private key in PEM. */
|
||||
@property(nonatomic, readonly, copy) NSString *private_key;
|
||||
|
||||
/** Public key in an x509 cert encoded in PEM. */
|
||||
@property(nonatomic, readonly, copy) NSString *certificate;
|
||||
|
||||
/**
|
||||
* Initialize an RTCCertificate with PEM strings for private_key and certificate.
|
||||
*/
|
||||
- (instancetype)initWithPrivateKey:(NSString *)private_key
|
||||
certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
|
||||
|
||||
- (instancetype)init NS_UNAVAILABLE;
|
||||
|
||||
/** Generate a new certificate for 're' use.
|
||||
*
|
||||
* Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
|
||||
* provided.
|
||||
* - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
|
||||
*/
|
||||
+ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,24 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
/** Implement this protocol to pass codec specific info from the encoder.
|
||||
* Corresponds to webrtc::CodecSpecificInfo.
|
||||
*/
|
||||
RTC_OBJC_EXPORT
|
||||
@protocol RTCCodecSpecificInfo <NSObject>
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,27 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCCodecSpecificInfo.h"
|
||||
#import "RTCMacros.h"
|
||||
|
||||
/** Class for H264 specific config. */
|
||||
typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
|
||||
RTCH264PacketizationModeNonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
|
||||
RTCH264PacketizationModeSingleNalUnit // Mode 0 - only single NALU allowed
|
||||
};
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
@interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
|
||||
|
||||
@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
|
||||
|
||||
@end
|
||||
@ -1,230 +0,0 @@
|
||||
/*
|
||||
* 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 RTCIceServer;
|
||||
@class RTCIntervalRange;
|
||||
|
||||
/**
|
||||
* 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) {
|
||||
RTCSdpSemanticsPlanB,
|
||||
RTCSdpSemanticsUnifiedPlan,
|
||||
};
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
@interface RTCConfiguration : NSObject
|
||||
|
||||
/** An array of Ice Servers available to be used by ICE. */
|
||||
@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
|
||||
|
||||
/** An RTCCertificate for 're' use. */
|
||||
@property(nonatomic, nullable) 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.
|
||||
* Default is NO.
|
||||
*/
|
||||
@property(nonatomic, assign) BOOL disableIPV6;
|
||||
|
||||
/** 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;
|
||||
|
||||
/** ICE Periodic Regathering
|
||||
* If set, WebRTC will periodically create and propose candidates without
|
||||
* starting a new ICE generation. The regathering happens continuously with
|
||||
* interval specified in milliseconds by the uniform distribution [a, b].
|
||||
*/
|
||||
@property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
|
||||
|
||||
/** Configure the SDP semantics used by this PeerConnection. Note that the
|
||||
* WebRTC 1.0 specification requires UnifiedPlan semantics. The
|
||||
* RTCRtpTransceiver API is only available with UnifiedPlan semantics.
|
||||
*
|
||||
* PlanB 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.
|
||||
*
|
||||
* UnifiedPlan 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.
|
||||
*
|
||||
* For users who wish to send multiple audio/video streams and need to stay
|
||||
* interoperable with legacy WebRTC implementations or use legacy APIs,
|
||||
* specify PlanB.
|
||||
*
|
||||
* For all other users, specify UnifiedPlan.
|
||||
*/
|
||||
@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 MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
|
||||
* that it should use the MediaTransportInterface.
|
||||
*/
|
||||
@property(nonatomic, assign) BOOL useMediaTransport;
|
||||
|
||||
/**
|
||||
* If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
|
||||
* that it should use the MediaTransportInterface for data channels.
|
||||
*/
|
||||
@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
|
||||
|
||||
/**
|
||||
* 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) RTCCryptoOptions *cryptoOptions;
|
||||
|
||||
/**
|
||||
* Time interval between audio RTCP reports.
|
||||
*/
|
||||
@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
|
||||
|
||||
/**
|
||||
* Time interval between video RTCP reports.
|
||||
*/
|
||||
@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
|
||||
|
||||
- (instancetype)init;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,63 +0,0 @@
|
||||
/*
|
||||
* 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 "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 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
|
||||
* SRTP_AES128_CM_SHA1_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
|
||||
@ -1,130 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
@interface 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 RTCDataChannel;
|
||||
RTC_OBJC_EXPORT
|
||||
@protocol RTCDataChannelDelegate <NSObject>
|
||||
|
||||
/** The data channel state changed. */
|
||||
- (void)dataChannelDidChangeState:(RTCDataChannel *)dataChannel;
|
||||
|
||||
/** The data channel successfully received a data buffer. */
|
||||
- (void)dataChannel:(RTCDataChannel *)dataChannel
|
||||
didReceiveMessageWithBuffer:(RTCDataBuffer *)buffer;
|
||||
|
||||
@optional
|
||||
/** The data channel's |bufferedAmount| changed. */
|
||||
- (void)dataChannel:(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 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<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:(RTCDataBuffer *)data;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,52 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
|
||||
NS_ASSUME_NONNULL_BEGIN
|
||||
|
||||
RTC_OBJC_EXPORT
|
||||
@interface 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
|
||||
@ -1,25 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
#import "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 RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,30 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
#import "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 RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
|
||||
|
||||
@property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
|
||||
|
||||
+ (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;
|
||||
|
||||
@end
|
||||
|
||||
NS_ASSUME_NONNULL_END
|
||||
@ -1,44 +0,0 @@
|
||||
/*
|
||||
* 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 "RTCMacros.h"
|
||||
|
||||
typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
|
||||
// Main dispatcher queue.
< | ||||