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Obj-C SDK Cleanup

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This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
This commit is contained in:
Anders Carlsson
2018-08-30 09:30:29 +02:00
committed by Commit Bot
parent 9ea5765f78
commit 7bca8ca4e2
470 changed files with 7255 additions and 5258 deletions
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240
sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
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@ -8,242 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCAudioSessionErrorDomain;
/** Method that requires lock was called without lock. */
extern NSInteger const kRTCAudioSessionErrorLockRequired;
/** Unknown configuration error occurred. */
extern NSInteger const kRTCAudioSessionErrorConfiguration;
@class RTCAudioSession;
@class RTCAudioSessionConfiguration;
// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
// from AVAudioSession and handle them before calling these delegate methods,
// at which point applications can perform additional processing if required.
RTC_EXPORT
@protocol RTCAudioSessionDelegate <NSObject>
@optional
/** Called on a system notification thread when AVAudioSession starts an
* interruption event.
*/
- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
/** Called on a system notification thread when AVAudioSession ends an
* interruption event.
*/
- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
shouldResumeSession:(BOOL)shouldResumeSession;
/** Called on a system notification thread when AVAudioSession changes the
* route.
*/
- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
reason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
/** Called on a system notification thread when AVAudioSession media server
* terminates.
*/
- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
/** Called on a system notification thread when AVAudioSession media server
* restarts.
*/
- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
/** Called on a WebRTC thread when the audio device is notified to begin
* playback or recording.
*/
- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
/** Called on a WebRTC thread when the audio device is notified to stop
* playback or recording.
*/
- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
/** Called when the AVAudioSession output volume value changes. */
- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
/** Called when the audio device detects a playout glitch. The argument is the
* number of glitches detected so far in the current audio playout session.
*/
- (void)audioSession:(RTCAudioSession *)audioSession
didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
/** Called when the audio session is about to change the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
/** Called after the audio session sucessfully changed the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
/** Called after the audio session failed to change the active state.
*/
- (void)audioSession:(RTCAudioSession *)audioSession
failedToSetActive:(BOOL)active
error:(NSError *)error;
@end
/** This is a protocol used to inform RTCAudioSession when the audio session
* activation state has changed outside of RTCAudioSession. The current known use
* case of this is when CallKit activates the audio session for the application
*/
RTC_EXPORT
@protocol RTCAudioSessionActivationDelegate <NSObject>
/** Called when the audio session is activated outside of the app by iOS. */
- (void)audioSessionDidActivate:(AVAudioSession *)session;
/** Called when the audio session is deactivated outside of the app by iOS. */
- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
@end
/** Proxy class for AVAudioSession that adds a locking mechanism similar to
* AVCaptureDevice. This is used to that interleaving configurations between
* WebRTC and the application layer are avoided.
*
* RTCAudioSession also coordinates activation so that the audio session is
* activated only once. See |setActive:error:|.
*/
RTC_EXPORT
@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
/** Convenience property to access the AVAudioSession singleton. Callers should
* not call setters on AVAudioSession directly, but other method invocations
* are fine.
*/
@property(nonatomic, readonly) AVAudioSession *session;
/** Our best guess at whether the session is active based on results of calls to
* AVAudioSession.
*/
@property(nonatomic, readonly) BOOL isActive;
/** Whether RTCAudioSession is currently locked for configuration. */
@property(nonatomic, readonly) BOOL isLocked;
/** If YES, WebRTC will not initialize the audio unit automatically when an
* audio track is ready for playout or recording. Instead, applications should
* call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
* as soon as an audio track is ready for playout or recording.
*/
@property(nonatomic, assign) BOOL useManualAudio;
/** This property is only effective if useManualAudio is YES.
* Represents permission for WebRTC to initialize the VoIP audio unit.
* When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
* stopped and uninitialized. This will stop incoming and outgoing audio.
* When set to YES, WebRTC will initialize and start the audio unit when it is
* needed (e.g. due to establishing an audio connection).
* This property was introduced to work around an issue where if an AVPlayer is
* playing audio while the VoIP audio unit is initialized, its audio would be
* either cut off completely or played at a reduced volume. By preventing
* the audio unit from being initialized until after the audio has completed,
* we are able to prevent the abrupt cutoff.
*/
@property(nonatomic, assign) BOOL isAudioEnabled;
// Proxy properties.
@property(readonly) NSString *category;
@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
@property(readonly) NSString *mode;
@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
@property(readonly) AVAudioSessionRouteDescription *currentRoute;
@property(readonly) NSInteger maximumInputNumberOfChannels;
@property(readonly) NSInteger maximumOutputNumberOfChannels;
@property(readonly) float inputGain;
@property(readonly) BOOL inputGainSettable;
@property(readonly) BOOL inputAvailable;
@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
@property(readonly) double sampleRate;
@property(readonly) double preferredSampleRate;
@property(readonly) NSInteger inputNumberOfChannels;
@property(readonly) NSInteger outputNumberOfChannels;
@property(readonly) float outputVolume;
@property(readonly) NSTimeInterval inputLatency;
@property(readonly) NSTimeInterval outputLatency;
@property(readonly) NSTimeInterval IOBufferDuration;
@property(readonly) NSTimeInterval preferredIOBufferDuration;
/** Default constructor. */
+ (instancetype)sharedInstance;
- (instancetype)init NS_UNAVAILABLE;
/** Adds a delegate, which is held weakly. */
- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Removes an added delegate. */
- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Request exclusive access to the audio session for configuration. This call
* will block if the lock is held by another object.
*/
- (void)lockForConfiguration;
/** Relinquishes exclusive access to the audio session. */
- (void)unlockForConfiguration;
/** If |active|, activates the audio session if it isn't already active.
* Successful calls must be balanced with a setActive:NO when activation is no
* longer required. If not |active|, deactivates the audio session if one is
* active and this is the last balanced call. When deactivating, the
* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
* AVAudioSession.
*/
- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
// The following methods are proxies for the associated methods on
// AVAudioSession. |lockForConfiguration| must be called before using them
// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
- (BOOL)setCategory:(NSString *)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
@end
@interface RTCAudioSession (Configuration)
/** Applies the configuration to the current session. Attempts to set all
* properties even if previous ones fail. Only the last error will be
* returned.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
/** Convenience method that calls both setConfiguration and setActive.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
active:(BOOL)active
error:(NSError **)outError;
@end
NS_ASSUME_NONNULL_END
#import "components/audio/RTCAudioSession.h"

39
sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
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@ -8,41 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import "WebRTC/RTCMacros.h"
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
// Struct to hold configuration values.
RTC_EXPORT
@interface RTCAudioSessionConfiguration : NSObject
@property(nonatomic, strong) NSString *category;
@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
@property(nonatomic, strong) NSString *mode;
@property(nonatomic, assign) double sampleRate;
@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
@property(nonatomic, assign) NSInteger inputNumberOfChannels;
@property(nonatomic, assign) NSInteger outputNumberOfChannels;
/** Initializes configuration to defaults. */
- (instancetype)init NS_DESIGNATED_INITIALIZER;
/** Returns the current configuration of the audio session. */
+ (instancetype)currentConfiguration;
/** Returns the configuration that WebRTC needs. */
+ (instancetype)webRTCConfiguration;
/** Provide a way to override the default configuration. */
+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
@end
NS_ASSUME_NONNULL_END
#import "components/audio/RTCAudioSessionConfiguration.h"

23
sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
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@ -8,25 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCAudioSource : RTCMediaSource
- (instancetype)init NS_UNAVAILABLE;
// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
// [0, 10].
// Temporary fix to be able to modify volume of remote audio tracks.
// TODO(kthelgason): Property stays here temporarily until a proper volume-api
// is available on the surface exposed by webrtc.
@property(nonatomic, assign) double volume;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCAudioSource.h"

19
sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSource;
RTC_EXPORT
@interface RTCAudioTrack : RTCMediaStreamTrack
- (instancetype)init NS_UNAVAILABLE;
/** The audio source for this audio track. */
@property(nonatomic, readonly) RTCAudioSource *source;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCAudioTrack.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCVPixelBuffer.h Normal file
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@ -0,0 +1,11 @@
/*
* 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 "components/video_frame_buffer/RTCCVPixelBuffer.h"

26
sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h
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@ -8,28 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCLogging.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and forwards them to a registered block.
// This class is not threadsafe.
RTC_EXPORT
@interface RTCCallbackLogger : NSObject
// The severity level to capture. The default is kRTCLoggingSeverityInfo.
@property(nonatomic, assign) RTCLoggingSeverity severity;
// The callback 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 void (^)(NSString*))callback;
- (void)stop;
@end
NS_ASSUME_NONNULL_END
#import "api/logging/RTCCallbackLogger.h"

21
sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
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@ -8,23 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
@class AVCaptureSession;
/** RTCCameraPreviewView is a view that renders local video from an
* AVCaptureSession.
*/
RTC_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
#import "helpers/RTCCameraPreviewView.h"

47
sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
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@ -8,49 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_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
#import "components/capturer/RTCCameraVideoCapturer.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_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
#import "api/peerconnection/RTCCertificate.h"

165
sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
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@ -8,167 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCCertificate.h>
#import <WebRTC/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_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;
/** 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;
/** 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;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCConfiguration.h"

121
sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
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@ -8,123 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_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_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_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
#import "api/peerconnection/RTCDataChannel.h"

43
sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
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@ -8,45 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_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
#import "api/peerconnection/RTCDataChannelConfiguration.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoDecoderFactory.h Normal file
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@ -0,0 +1,11 @@
/*
* 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 "components/video_codec/RTCDefaultVideoDecoderFactory.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoEncoderFactory.h Normal file
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@ -0,0 +1,11 @@
/*
* 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 "components/video_codec/RTCDefaultVideoEncoderFactory.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
// Main dispatcher queue.
RTCDispatcherTypeMain,
// Used for starting/stopping AVCaptureSession, and assigning
// capture session to AVCaptureVideoPreviewLayer.
RTCDispatcherTypeCaptureSession,
// Used for operations on AVAudioSession.
RTCDispatcherTypeAudioSession,
};
/** Dispatcher that asynchronously dispatches blocks to a specific
* shared dispatch queue.
*/
RTC_EXPORT
@interface RTCDispatcher : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Dispatch the block asynchronously on the queue for dispatchType.
* @param dispatchType The queue type to dispatch on.
* @param block The block to dispatch asynchronously.
*/
+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
/** Returns YES if run on queue for the dispatchType otherwise NO.
* Useful for asserting that a method is run on a correct queue.
*/
+ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
@end
#import "helpers/RTCDispatcher.h"

61
sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
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@ -8,63 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@protocol RTCDtmfSender <NSObject>
/**
* Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
* returns false. To be able to send DTMF, the associated RTCRtpSender must be
* able to send packets, and a "telephone-event" codec must be negotiated.
*/
@property(nonatomic, readonly) BOOL canInsertDtmf;
/**
* Queues a task that sends the DTMF tones. The tones parameter is treated
* as a series of characters. The characters 0 through 9, A through D, #, and *
* generate the associated DTMF tones. The characters a to d are equivalent
* to A to D. The character ',' indicates a delay of 2 seconds before
* processing the next character in the tones parameter.
*
* Unrecognized characters are ignored.
*
* @param duration The parameter indicates the duration to use for each
* character passed in the tones parameter. The duration cannot be more
* than 6000 or less than 70 ms.
*
* @param interToneGap The parameter indicates the gap between tones.
* This parameter must be at least 50 ms but should be as short as
* possible.
*
* If InsertDtmf is called on the same object while an existing task for this
* object to generate DTMF is still running, the previous task is canceled.
* Returns true on success and false on failure.
*/
- (BOOL)insertDtmf:(nonnull NSString *)tones
duration:(NSTimeInterval)duration
interToneGap:(NSTimeInterval)interToneGap;
/** The tones remaining to be played out */
- (nonnull NSString *)remainingTones;
/**
* The current tone duration value. This value will be the value last set via the
* insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
*/
- (NSTimeInterval)duration;
/**
* The current value of the between-tone gap. This value will be the value last set
* via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
* called.
*/
- (NSTimeInterval)interToneGap;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCDtmfSender.h"

35
sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
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@ -8,37 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCVideoViewShading.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCEAGLVideoView;
RTC_EXPORT
@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
@end
/**
* RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
* bounds using OpenGLES 2.0 or OpenGLES 3.0.
*/
RTC_EXPORT
NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
- (instancetype)initWithFrame:(CGRect)frame
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithCoder:(NSCoder *)aDecoder
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCEAGLVideoView.h"

37
sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
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@ -8,39 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
RTC_EXTERN NSString * const kRTCFieldTrialImprovedBitrateEstimateKey;
RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
/** The valid value for field trials above. */
RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
/** Use a string returned by RTCFieldTrialMedianSlopeFilterValue as the value. */
RTC_EXTERN NSString * const kRTCFieldTrialMedianSlopeFilterKey;
RTC_EXTERN NSString *RTCFieldTrialMedianSlopeFilterValue(
size_t windowSize, double thresholdGain);
/** Use a string returned by RTCFieldTrialTrendlineFilterValue as the value. */
RTC_EXTERN NSString * const kRTCFieldTrialTrendlineFilterKey;
/** Returns a valid value for kRTCFieldTrialTrendlineFilterKey. */
RTC_EXTERN NSString *RTCFieldTrialTrendlineFilterValue(
size_t windowSize, double smoothingCoeff, double thresholdGain);
/** Initialize field trials using a dictionary mapping field trial keys to their values. See above
* for valid keys and values.
* Must be called before any other call into WebRTC. See:
* webrtc/system_wrappers/include/field_trial_default.h
*/
RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
#import "api/peerconnection/RTCFieldTrials.h"

65
sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
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@ -8,67 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
RTCFileLoggerSeverityVerbose,
RTCFileLoggerSeverityInfo,
RTCFileLoggerSeverityWarning,
RTCFileLoggerSeverityError
};
typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
RTCFileLoggerTypeCall,
RTCFileLoggerTypeApp,
};
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and saves them to a file. The file size
// will not exceed the given maximum bytesize. When the maximum bytesize is
// reached, logs are rotated according to the rotationType specified.
// For kRTCFileLoggerTypeCall, logs from the beginning and the end
// are preserved while the middle section is overwritten instead.
// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
// This class is not threadsafe.
RTC_EXPORT
@interface RTCFileLogger : NSObject
// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
@property(nonatomic, assign) RTCFileLoggerSeverity severity;
// The rotation type for this file logger. The default is
// kRTCFileLoggerTypeCall.
@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
// Disables buffering disk writes. Should be set before |start|. Buffering
// is enabled by default for performance.
@property(nonatomic, assign) BOOL shouldDisableBuffering;
// Default constructor provides default settings for dir path, file size and
// rotation type.
- (instancetype)init;
// Create file logger with default rotation type.
- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
- (instancetype)initWithDirPath:(NSString *)dirPath
maxFileSize:(NSUInteger)maxFileSize
rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
// Starts writing WebRTC logs to disk if not already started. Overwrites any
// existing file(s).
- (void)start;
// Stops writing WebRTC logs to disk. This method is also called on dealloc.
- (void)stop;
// Returns the current contents of the logs, or nil if start has been called
// without a stop.
- (nullable NSData *)logData;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCFileLogger.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
View File

@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
/**
* Error passing block.
*/
typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
/**
* Captures buffers from bundled video file.
*
* See @c RTCVideoCapturer for more info on capturers.
*/
RTC_EXPORT
NS_CLASS_AVAILABLE_IOS(10)
@interface RTCFileVideoCapturer : RTCVideoCapturer
/**
* Starts asynchronous capture of frames from video file.
*
* Capturing is not started if error occurs. Underlying error will be
* relayed in the errorBlock if one is provided.
* Successfully captured video frames will be passed to the delegate.
*
* @param nameOfFile The name of the bundled video file to be read.
* @errorBlock block to be executed upon error.
*/
- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
/**
* Immediately stops capture.
*/
- (void)stopCapture;
@end
NS_ASSUME_NONNULL_END
#import "components/capturer/RTCFileVideoCapturer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCH264ProfileLevelId.h Normal file
View File

@ -0,0 +1,11 @@
/*
* 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 "components/video_codec/RTCH264ProfileLevelId.h"

40
sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
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@ -8,42 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCIceCandidate : NSObject
/**
* If present, the identifier of the "media stream identification" for the media
* component this candidate is associated with.
*/
@property(nonatomic, readonly, nullable) NSString *sdpMid;
/**
* The index (starting at zero) of the media description this candidate is
* associated with in the SDP.
*/
@property(nonatomic, readonly) int sdpMLineIndex;
/** The SDP string for this candidate. */
@property(nonatomic, readonly) NSString *sdp;
/** The URL of the ICE server which this candidate is gathered from. */
@property(nonatomic, readonly, nullable) NSString *serverUrl;
- (instancetype)init NS_UNAVAILABLE;
/**
* Initialize an RTCIceCandidate from SDP.
*/
- (instancetype)initWithSdp:(NSString *)sdp
sdpMLineIndex:(int)sdpMLineIndex
sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIceCandidate.h"

105
sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
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@ -8,107 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
RTCTlsCertPolicySecure,
RTCTlsCertPolicyInsecureNoCheck
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCIceServer : NSObject
/** URI(s) for this server represented as NSStrings. */
@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
/** Username to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *username;
/** Credential to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *credential;
/**
* TLS certificate policy to use if this RTCIceServer object is a TURN server.
*/
@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
/**
If the URIs in |urls| only contain IP addresses, this field can be used
to indicate the hostname, which may be necessary for TLS (using the SNI
extension). If |urls| itself contains the hostname, this isn't necessary.
*/
@property(nonatomic, readonly, nullable) NSString *hostname;
/** List of protocols to be used in the TLS ALPN extension. */
@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
/**
List elliptic curves to be used in the TLS elliptic curves extension.
Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
*/
@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
- (nonnull instancetype)init NS_UNAVAILABLE;
/** Convenience initializer for a server with no authentication (e.g. STUN). */
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and credentialType.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and TLS cert policy.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy and hostname.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname and ALPN protocols.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname, ALPN protocols and
* elliptic curves.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIceServer.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
View File

@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
NS_ASSUME_NONNULL_BEGIN
@interface RTCIntervalRange : NSObject
@property(nonatomic, readonly) NSInteger min;
@property(nonatomic, readonly) NSInteger max;
- (instancetype)init;
- (instancetype)initWithMin:(NSInteger)min max:(NSInteger)max NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCIntervalRange.h"

28
sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** This does not currently conform to the spec. */
RTC_EXPORT
@interface RTCLegacyStatsReport : NSObject
/** Time since 1970-01-01T00:00:00Z in milliseconds. */
@property(nonatomic, readonly) CFTimeInterval timestamp;
/** The type of stats held by this object. */
@property(nonatomic, readonly) NSString *type;
/** The identifier for this object. */
@property(nonatomic, readonly) NSString *reportId;
/** A dictionary holding the actual stats. */
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCLegacyStatsReport.h"

57
sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
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@ -8,59 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
// Subset of rtc::LoggingSeverity.
typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
RTCLoggingSeverityVerbose,
RTCLoggingSeverityInfo,
RTCLoggingSeverityWarning,
RTCLoggingSeverityError,
};
// Wrapper for C++ RTC_LOG(sev) macros.
// Logs the log string to the webrtc logstream for the given severity.
RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
// Wrapper for rtc::LogMessage::LogToDebug.
// Sets the minimum severity to be logged to console.
RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
// Returns the filename with the path prefix removed.
RTC_EXTERN NSString* RTCFileName(const char* filePath);
// Some convenience macros.
#define RTCLogString(format, ...) \
[NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
__LINE__, __FUNCTION__, ##__VA_ARGS__]
#define RTCLogFormat(severity, format, ...) \
do { \
NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
RTCLogEx(severity, log_string); \
} while (false)
#define RTCLogVerbose(format, ...) \
RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
#define RTCLogInfo(format, ...) \
RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
#define RTCLogWarning(format, ...) \
RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
#define RTCLogError(format, ...) \
RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
#if !defined(NDEBUG)
#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
#else
#define RTCLogDebug(format, ...) \
do { \
} while (false)
#endif
#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
#import "base/RTCLogging.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
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@ -8,13 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Cocoa/Cocoa.h>
#import "WebRTC/RTCVideoRenderer.h"
NS_AVAILABLE_MAC(10.11)
@interface RTCMTLNSVideoView : NSView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
+ (BOOL)isMetalAvailable;
@end
#import "components/renderer/metal/RTCMTLNSVideoView.h"

42
sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
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@ -8,44 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import "WebRTC/RTCVideoFrame.h"
#import "WebRTC/RTCVideoRenderer.h"
// Check if metal is supported in WebRTC.
// NOTE: Currently arm64 == Metal.
#if defined(__aarch64__)
#define RTC_SUPPORTS_METAL
#endif
NS_ASSUME_NONNULL_BEGIN
/**
* RTCMTLVideoView is thin wrapper around MTKView.
*
* It has id<RTCVideoRenderer> property that renders video frames in the view's
* bounds using Metal.
* NOTE: always check if metal is available on the running device via
* RTC_SUPPORTS_METAL macro before initializing this class.
*/
NS_CLASS_AVAILABLE_IOS(9)
RTC_EXPORT
@interface RTCMTLVideoView : UIView<RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
@property(nonatomic) UIViewContentMode videoContentMode;
/** @abstract Enables/disables rendering.
*/
@property(nonatomic, getter=isEnabled) BOOL enabled;
/** @abstract Wrapped RTCVideoRotation, or nil.
*/
@property(nonatomic, nullable) NSValue* rotationOverride;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/metal/RTCMTLVideoView.h"

19
sdk/objc/Framework/Headers/WebRTC/RTCMacros.h
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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#define SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#define RTC_EXPORT __attribute__((visibility("default")))
#if defined(__cplusplus)
#define RTC_EXTERN extern "C" RTC_EXPORT
#else
#define RTC_EXTERN extern RTC_EXPORT
#endif
#ifdef __OBJC__
#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
#else
#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
#endif
#endif // SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
#import "base/RTCMacros.h"

45
sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
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@ -8,47 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Constraint keys for media sources. */
RTC_EXTERN NSString *const kRTCMediaConstraintsMinAspectRatio;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxAspectRatio;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxWidth;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinWidth;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxHeight;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinHeight;
RTC_EXTERN NSString *const kRTCMediaConstraintsMaxFrameRate;
RTC_EXTERN NSString *const kRTCMediaConstraintsMinFrameRate;
/** The value for this key should be a base64 encoded string containing
* the data from the serialized configuration proto.
*/
RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
/** Constraint keys for generating offers and answers. */
RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
/** Constraint values for Boolean parameters. */
RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
RTC_EXPORT
@interface RTCMediaConstraints : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Initialize with mandatory and/or optional constraints. */
- (instancetype)
initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaConstraints.h"

25
sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
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@ -8,27 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCSourceState) {
RTCSourceStateInitializing,
RTCSourceStateLive,
RTCSourceStateEnded,
RTCSourceStateMuted,
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCMediaSource : NSObject
/** The current state of the RTCMediaSource. */
@property(nonatomic, readonly) RTCSourceState state;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaSource.h"

40
sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
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@ -8,42 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioTrack;
@class RTCPeerConnectionFactory;
@class RTCVideoTrack;
RTC_EXPORT
@interface RTCMediaStream : NSObject
/** The audio tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTCAudioTrack *> *audioTracks;
/** The video tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTCVideoTrack *> *videoTracks;
/** An identifier for this media stream. */
@property(nonatomic, readonly) NSString *streamId;
- (instancetype)init NS_UNAVAILABLE;
/** Adds the given audio track to this media stream. */
- (void)addAudioTrack:(RTCAudioTrack *)audioTrack;
/** Adds the given video track to this media stream. */
- (void)addVideoTrack:(RTCVideoTrack *)videoTrack;
/** Removes the given audio track to this media stream. */
- (void)removeAudioTrack:(RTCAudioTrack *)audioTrack;
/** Removes the given video track to this media stream. */
- (void)removeVideoTrack:(RTCVideoTrack *)videoTrack;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaStream.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
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@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the state of the track. This exposes the same states in C++.
*/
typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
RTCMediaStreamTrackStateLive,
RTCMediaStreamTrackStateEnded
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
RTC_EXPORT
@interface RTCMediaStreamTrack : NSObject
/**
* The kind of track. For example, "audio" if this track represents an audio
* track and "video" if this track represents a video track.
*/
@property(nonatomic, readonly) NSString *kind;
/** An identifier string. */
@property(nonatomic, readonly) NSString *trackId;
/** The enabled state of the track. */
@property(nonatomic, assign) BOOL isEnabled;
/** The state of the track. */
@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMediaStreamTrack.h"

14
sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
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@ -8,16 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMetricsSampleInfo.h>
/**
* Enables gathering of metrics (which can be fetched with
* RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
*/
RTC_EXTERN void RTCEnableMetrics(void);
/** Gets and clears native histograms. */
RTC_EXTERN NSArray<RTCMetricsSampleInfo*>* RTCGetAndResetMetrics(void);
#import "api/peerconnection/RTCMetrics.h"

39
sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
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@ -8,41 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCMetricsSampleInfo : NSObject
/**
* Example of RTCMetricsSampleInfo:
* name: "WebRTC.Video.InputFramesPerSecond"
* min: 1
* max: 100
* bucketCount: 50
* samples: [29]:2 [30]:1
*/
/** The name of the histogram. */
@property(nonatomic, readonly) NSString *name;
/** The minimum bucket value. */
@property(nonatomic, readonly) int min;
/** The maximum bucket value. */
@property(nonatomic, readonly) int max;
/** The number of buckets. */
@property(nonatomic, readonly) int bucketCount;
/** A dictionary holding the samples <value, # of events>. */
@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCMetricsSampleInfo.h"

30
sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
View File

@ -8,32 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#if !TARGET_OS_IPHONE
#import <AppKit/NSOpenGLView.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCVideoViewShading.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCNSGLVideoView;
@protocol RTCNSGLVideoViewDelegate <RTCVideoViewDelegate>
@end
@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
- (instancetype)initWithFrame:(NSRect)frameRect
pixelFormat:(NSOpenGLPixelFormat *)format
shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#endif
#import "components/renderer/opengl/RTCNSGLVideoView.h"

307
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
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@ -8,309 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
@class RTCConfiguration;
@class RTCDataChannel;
@class RTCDataChannelConfiguration;
@class RTCIceCandidate;
@class RTCMediaConstraints;
@class RTCMediaStream;
@class RTCMediaStreamTrack;
@class RTCPeerConnectionFactory;
@class RTCRtpReceiver;
@class RTCRtpSender;
@class RTCRtpTransceiver;
@class RTCRtpTransceiverInit;
@class RTCSessionDescription;
@class RTCLegacyStatsReport;
typedef NS_ENUM(NSInteger, RTCRtpMediaType);
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCPeerConnectionErrorDomain;
extern int const kRTCSessionDescriptionErrorCode;
/** Represents the signaling state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCSignalingState) {
RTCSignalingStateStable,
RTCSignalingStateHaveLocalOffer,
RTCSignalingStateHaveLocalPrAnswer,
RTCSignalingStateHaveRemoteOffer,
RTCSignalingStateHaveRemotePrAnswer,
// Not an actual state, represents the total number of states.
RTCSignalingStateClosed,
};
/** Represents the ice connection state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
RTCIceConnectionStateNew,
RTCIceConnectionStateChecking,
RTCIceConnectionStateConnected,
RTCIceConnectionStateCompleted,
RTCIceConnectionStateFailed,
RTCIceConnectionStateDisconnected,
RTCIceConnectionStateClosed,
RTCIceConnectionStateCount,
};
/** Represents the ice gathering state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
RTCIceGatheringStateNew,
RTCIceGatheringStateGathering,
RTCIceGatheringStateComplete,
};
/** Represents the stats output level. */
typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
RTCStatsOutputLevelStandard,
RTCStatsOutputLevelDebug,
};
@class RTCPeerConnection;
RTC_EXPORT
@protocol RTCPeerConnectionDelegate <NSObject>
/** Called when the SignalingState changed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeSignalingState:(RTCSignalingState)stateChanged;
/** Called when media is received on a new stream from remote peer. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream;
/** Called when a remote peer closes a stream.
* This is not called when RTCSdpSemanticsUnifiedPlan is specified.
*/
- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream;
/** Called when negotiation is needed, for example ICE has restarted. */
- (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection;
/** Called any time the IceConnectionState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeIceConnectionState:(RTCIceConnectionState)newState;
/** Called any time the IceGatheringState changes. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didChangeIceGatheringState:(RTCIceGatheringState)newState;
/** New ice candidate has been found. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didGenerateIceCandidate:(RTCIceCandidate *)candidate;
/** Called when a group of local Ice candidates have been removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didRemoveIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
/** New data channel has been opened. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didOpenDataChannel:(RTCDataChannel *)dataChannel;
/** Called when signaling indicates a transceiver will be receiving media from
* the remote endpoint.
* This is only called with RTCSdpSemanticsUnifiedPlan specified.
*/
@optional
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver;
/** Called when a receiver and its track are created. */
@optional
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didAddReceiver:(RTCRtpReceiver *)rtpReceiver
streams:(NSArray<RTCMediaStream *> *)mediaStreams;
/** Called when the receiver and its track are removed. */
- (void)peerConnection:(RTCPeerConnection *)peerConnection
didRemoveReceiver:(RTCRtpReceiver *)rtpReceiver;
@end
RTC_EXPORT
@interface RTCPeerConnection : NSObject
/** The object that will be notifed about events such as state changes and
* streams being added or removed.
*/
@property(nonatomic, weak, nullable) id<RTCPeerConnectionDelegate> delegate;
/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
* |senders| instead.
*/
@property(nonatomic, readonly) NSArray<RTCMediaStream *> *localStreams;
@property(nonatomic, readonly, nullable) RTCSessionDescription *localDescription;
@property(nonatomic, readonly, nullable) RTCSessionDescription *remoteDescription;
@property(nonatomic, readonly) RTCSignalingState signalingState;
@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
@property(nonatomic, readonly, copy) RTCConfiguration *configuration;
/** Gets all RTCRtpSenders associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpSender.
* Use isEqual: instead of == to compare RTCRtpSender instances.
*/
@property(nonatomic, readonly) NSArray<RTCRtpSender *> *senders;
/** Gets all RTCRtpReceivers associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpReceiver.
* Use isEqual: instead of == to compare RTCRtpReceiver instances.
*/
@property(nonatomic, readonly) NSArray<RTCRtpReceiver *> *receivers;
/** Gets all RTCRtpTransceivers associated with this peer connection.
* Note: reading this property returns different instances of
* RTCRtpTransceiver. Use isEqual: instead of == to compare RTCRtpTransceiver
* instances.
* This is only available with RTCSdpSemanticsUnifiedPlan specified.
*/
@property(nonatomic, readonly) NSArray<RTCRtpTransceiver *> *transceivers;
- (instancetype)init NS_UNAVAILABLE;
/** Sets the PeerConnection's global configuration to |configuration|.
* Any changes to STUN/TURN servers or ICE candidate policy will affect the
* next gathering phase, and cause the next call to createOffer to generate
* new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
* cannot be changed with this method.
*/
- (BOOL)setConfiguration:(RTCConfiguration *)configuration;
/** Terminate all media and close the transport. */
- (void)close;
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTCIceCandidate *)candidate;
/** Remove a group of remote candidates from the ICE Agent. */
- (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
/** Add a new media stream to be sent on this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTrack instead.
*/
- (void)addStream:(RTCMediaStream *)stream;
/** Remove the given media stream from this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* removeTrack instead.
*/
- (void)removeStream:(RTCMediaStream *)stream;
/** Add a new media stream track to be sent on this peer connection, and return
* the newly created RTCRtpSender. The RTCRtpSender will be associated with
* the streams specified in the |streamIds| list.
*
* Errors: If an error occurs, returns nil. An error can occur if:
* - A sender already exists for the track.
* - The peer connection is closed.
*/
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds;
/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
*
* With UnifiedPlan semantics, sets sender's track to null and removes the
* send component from the associated RTCRtpTransceiver's direction.
*
* Returns YES on success.
*/
- (BOOL)removeTrack:(RTCRtpSender *)sender;
/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
* transceivers. Adding a transceiver will cause future calls to CreateOffer
* to add a media description for the corresponding transceiver.
*
* The initial value of |mid| in the returned transceiver is nil. Setting a
* new session description may change it to a non-nil value.
*
* https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
*
* Optionally, an RtpTransceiverInit structure can be specified to configure
* the transceiver from construction. If not specified, the transceiver will
* default to having a direction of kSendRecv and not be part of any streams.
*
* These methods are only available when Unified Plan is enabled (see
* RTCConfiguration).
*/
/** Adds a transceiver with a sender set to transmit the given track. The kind
* of the transceiver (and sender/receiver) will be derived from the kind of
* the track.
*/
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track;
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
init:(RTCRtpTransceiverInit *)init;
/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
* or RTCRtpMediaTypeVideo.
*/
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTCRtpTransceiverInit *)init;
/** Generate an SDP offer. */
- (void)offerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
NSError *_Nullable error))completionHandler;
/** Generate an SDP answer. */
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the local description. */
- (void)setLocalDescription:(RTCSessionDescription *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Apply the supplied RTCSessionDescription as the remote description. */
- (void)setRemoteDescription:(RTCSessionDescription *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
/** Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Nil parameters will be unchanged. Setting
* |currentBitrateBps| will force the available bitrate estimate to the given
* value. Returns YES if the parameters were successfully updated.
*/
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
/** Start or stop recording an Rtc EventLog. */
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
- (void)stopRtcEventLog;
@end
@interface RTCPeerConnection (Media)
/** Create an RTCRtpSender with the specified kind and media stream ID.
* See RTCMediaStreamTrack.h for available kinds.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTransceiver instead.
*/
- (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
@end
@interface RTCPeerConnection (DataChannel)
/** Create a new data channel with the given label and configuration. */
- (nullable RTCDataChannel *)dataChannelForLabel:(NSString *)label
configuration:(RTCDataChannelConfiguration *)configuration;
@end
@interface RTCPeerConnection (Stats)
/** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
* statistics are gathered for all tracks.
*/
- (void)statsForTrack:(nullable RTCMediaStreamTrack *)mediaStreamTrack
statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
completionHandler:(nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnection.h"

72
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
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@ -8,74 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCAudioSource;
@class RTCAudioTrack;
@class RTCConfiguration;
@class RTCMediaConstraints;
@class RTCMediaStream;
@class RTCPeerConnection;
@class RTCVideoSource;
@class RTCVideoTrack;
@class RTCPeerConnectionFactoryOptions;
@protocol RTCPeerConnectionDelegate;
@protocol RTCVideoDecoderFactory;
@protocol RTCVideoEncoderFactory;
RTC_EXPORT
@interface RTCPeerConnectionFactory : NSObject
/* Initialize object with default H264 video encoder/decoder factories */
- (instancetype)init;
/* Initialize object with injectable video encoder/decoder factories */
- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory;
/** Initialize an RTCAudioSource with constraints. */
- (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints;
/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source with no
* constraints.
*/
- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId;
/** Initialize an RTCAudioTrack with a source and an id. */
- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source trackId:(NSString *)trackId;
/** Initialize a generic RTCVideoSource. The RTCVideoSource should be passed to a RTCVideoCapturer
* implementation, e.g. RTCCameraVideoCapturer, in order to produce frames.
*/
- (RTCVideoSource *)videoSource;
/** Initialize an RTCVideoTrack with a source and an id. */
- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source trackId:(NSString *)trackId;
/** Initialize an RTCMediaStream with an id. */
- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:
(nullable id<RTCPeerConnectionDelegate>)delegate;
/** Set the options to be used for subsequently created RTCPeerConnections */
- (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options;
/** Start an AecDump recording. This API call will likely change in the future. */
- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
/* Stop an active AecDump recording */
- (void)stopAecDump;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactory.h"

33
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
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@ -8,35 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCPeerConnectionFactoryOptions : NSObject
@property(nonatomic, assign) BOOL disableEncryption;
@property(nonatomic, assign) BOOL disableNetworkMonitor;
@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
@property(nonatomic, assign) BOOL enableAes128Sha1_32CryptoCipher;
@property(nonatomic, assign) BOOL enableGcmCryptoSuites;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactoryOptions.h"

21
sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
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@ -8,23 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtcpParameters : NSObject
/** The Canonical Name used by RTCP. */
@property(nonatomic, readonly, copy) NSString *cname;
/** Whether reduced size RTCP is configured or compound RTCP. */
@property(nonatomic, assign) BOOL isReducedSize;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtcpParameters.h"

64
sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
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@ -8,66 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const NSString *const kRTCRtxCodecName;
RTC_EXTERN const NSString *const kRTCRedCodecName;
RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
RTC_EXTERN const NSString *const kRTCOpusCodecName;
RTC_EXTERN const NSString *const kRTCIsacCodecName;
RTC_EXTERN const NSString *const kRTCL16CodecName;
RTC_EXTERN const NSString *const kRTCG722CodecName;
RTC_EXTERN const NSString *const kRTCIlbcCodecName;
RTC_EXTERN const NSString *const kRTCPcmuCodecName;
RTC_EXTERN const NSString *const kRTCPcmaCodecName;
RTC_EXTERN const NSString *const kRTCDtmfCodecName;
RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
RTC_EXTERN const NSString *const kRTCVp8CodecName;
RTC_EXTERN const NSString *const kRTCVp9CodecName;
RTC_EXTERN const NSString *const kRTCH264CodecName;
/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
RTC_EXPORT
@interface RTCRtpCodecParameters : NSObject
/** The RTP payload type. */
@property(nonatomic, assign) int payloadType;
/**
* The codec MIME subtype. Valid types are listed in:
* http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
*
* Several supported types are represented by the constants above.
*/
@property(nonatomic, readonly, nonnull) NSString *name;
/**
* The media type of this codec. Equivalent to MIME top-level type.
*
* Valid values are kRTCMediaStreamTrackKindAudio and
* kRTCMediaStreamTrackKindVideo.
*/
@property(nonatomic, readonly, nonnull) NSString *kind;
/** The codec clock rate expressed in Hertz. */
@property(nonatomic, readonly, nullable) NSNumber *clockRate;
/**
* The number of channels (mono=1, stereo=2).
* Set to null for video codecs.
**/
@property(nonatomic, readonly, nullable) NSNumber *numChannels;
/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpCodecParameters.h"

33
sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
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@ -8,35 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpEncodingParameters : NSObject
/** Controls whether the encoding is currently transmitted. */
@property(nonatomic, assign) BOOL isActive;
/** The maximum bitrate to use for the encoding, or nil if there is no
* limit.
*/
@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
/** The minimum bitrate to use for the encoding, or nil if there is no
* limit.
*
* Not implemented.
*/
@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
/** The SSRC being used by this encoding. */
@property(nonatomic, readonly, nullable) NSNumber *ssrc;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpEncodingParameters.h"

24
sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h
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@ -8,26 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpHeaderExtension : NSObject
/** The URI of the RTP header extension, as defined in RFC5285. */
@property(nonatomic, readonly, copy) NSString *uri;
/** The value put in the RTP packet to identify the header extension. */
@property(nonatomic, readonly) int id;
/** Whether the header extension is encrypted or not. */
@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpHeaderExtension.h"

34
sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
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@ -8,36 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtcpParameters.h>
#import <WebRTC/RTCRtpCodecParameters.h>
#import <WebRTC/RTCRtpEncodingParameters.h>
#import <WebRTC/RTCRtpHeaderExtension.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCRtpParameters : NSObject
/** A unique identifier for the last set of parameters applied. */
@property(nonatomic, copy) NSString *transactionId;
/** Parameters used for RTCP. */
@property(nonatomic, readonly, copy) RTCRtcpParameters *rtcp;
/** An array containing parameters for RTP header extensions. */
@property(nonatomic, readonly, copy) NSArray<RTCRtpHeaderExtension *> *headerExtensions;
/** The currently active encodings in the order of preference. */
@property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;
/** The negotiated set of send codecs in order of preference. */
@property(nonatomic, copy) NSArray<RTCRtpCodecParameters *> *codecs;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpParameters.h"

73
sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
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@ -8,75 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
/** Represents the media type of the RtpReceiver. */
typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
RTCRtpMediaTypeAudio,
RTCRtpMediaTypeVideo,
RTCRtpMediaTypeData,
};
@class RTCRtpReceiver;
RTC_EXPORT
@protocol RTCRtpReceiverDelegate <NSObject>
/** Called when the first RTP packet is received.
*
* Note: Currently if there are multiple RtpReceivers of the same media type,
* they will all call OnFirstPacketReceived at once.
*
* For example, if we create three audio receivers, A/B/C, they will listen to
* the same signal from the underneath network layer. Whenever the first audio packet
* is received, the underneath signal will be fired. All the receivers A/B/C will be
* notified and the callback of the receiver's delegate will be called.
*
* The process is the same for video receivers.
*/
- (void)rtpReceiver:(RTCRtpReceiver *)rtpReceiver
didReceiveFirstPacketForMediaType:(RTCRtpMediaType)mediaType;
@end
RTC_EXPORT
@protocol RTCRtpReceiver <NSObject>
/** A unique identifier for this receiver. */
@property(nonatomic, readonly) NSString *receiverId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
*
* The WebRTC specification only defines RTCRtpParameters in terms of senders,
* but this API also applies them to receivers, similar to ORTC:
* http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
*/
@property(nonatomic, readonly) RTCRtpParameters *parameters;
/** The RTCMediaStreamTrack associated with the receiver.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, readonly, nullable) RTCMediaStreamTrack *track;
/** The delegate for this RtpReceiver. */
@property(nonatomic, weak) id<RTCRtpReceiverDelegate> delegate;
@end
RTC_EXPORT
@interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpReceiver.h"

41
sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
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@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCDtmfSender.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@protocol RTCRtpSender <NSObject>
/** A unique identifier for this sender. */
@property(nonatomic, readonly) NSString *senderId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
*/
@property(nonatomic, copy) RTCRtpParameters *parameters;
/** The RTCMediaStreamTrack associated with the sender.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, copy, nullable) RTCMediaStreamTrack *track;
/** The RTCDtmfSender accociated with the RTP sender. */
@property(nonatomic, readonly, nullable) id<RTCDtmfSender> dtmfSender;
@end
RTC_EXPORT
@interface RTCRtpSender : NSObject <RTCRtpSender>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpSender.h"

120
sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
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@ -8,122 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtpReceiver.h>
#import <WebRTC/RTCRtpSender.h>
NS_ASSUME_NONNULL_BEGIN
/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
RTCRtpTransceiverDirectionSendRecv,
RTCRtpTransceiverDirectionSendOnly,
RTCRtpTransceiverDirectionRecvOnly,
RTCRtpTransceiverDirectionInactive,
};
/** Structure for initializing an RTCRtpTransceiver in a call to
* RTCPeerConnection.addTransceiver.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
*/
RTC_EXPORT
@interface RTCRtpTransceiverInit : NSObject
/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The added RTCRtpTransceiver will be added to these streams. */
@property(nonatomic) NSArray<NSString *> *streamIds;
/** TODO(bugs.webrtc.org/7600): Not implemented. */
@property(nonatomic) NSArray<RTCRtpEncodingParameters *> *sendEncodings;
@end
@class RTCRtpTransceiver;
/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
* specification. A transceiver represents a combination of an RTCRtpSender
* and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
* RTCRtpTransceiver is said to be associated with a media description if its
* mid property is non-nil; otherwise, it is said to be disassociated.
* JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
*
* Note that RTCRtpTransceivers are only supported when using
* RTCPeerConnection with Unified Plan SDP.
*
* WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
*/
RTC_EXPORT
@protocol RTCRtpTransceiver <NSObject>
/** Media type of the transceiver. The sender and receiver will also have this
* type.
*/
@property(nonatomic, readonly) RTCRtpMediaType mediaType;
/** The mid attribute is the mid negotiated and present in the local and
* remote descriptions. Before negotiation is complete, the mid value may be
* nil. After rollbacks, the value may change from a non-nil value to nil.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
*/
@property(nonatomic, readonly) NSString *mid;
/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
* media that may be sent with the transceiver's mid. The sender is always
* present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
*/
@property(nonatomic, readonly) RTCRtpSender *sender;
/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
* media that may be received with the transceiver's mid. The receiver is
* always present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
*/
@property(nonatomic, readonly) RTCRtpReceiver *receiver;
/** The isStopped attribute indicates that the sender of this transceiver will
* no longer send, and that the receiver will no longer receive. It is true if
* either stop has been called or if setting the local or remote description
* has caused the RTCRtpTransceiver to be stopped.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
*/
@property(nonatomic, readonly) BOOL isStopped;
/** The direction attribute indicates the preferred direction of this
* transceiver, which will be used in calls to createOffer and createAnswer.
* An update of directionality does not take effect immediately. Instead,
* future calls to createOffer and createAnswer mark the corresponding media
* descriptions as sendrecv, sendonly, recvonly, or inactive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The currentDirection attribute indicates the current direction negotiated
* for this transceiver. If this transceiver has never been represented in an
* offer/answer exchange, or if the transceiver is stopped, the value is not
* present and this method returns NO.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
*/
- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
* this transceiver will no longer send, the receiver will no longer receive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
*/
- (void)stop;
@end
RTC_EXPORT
@interface RTCRtpTransceiver : NSObject <RTCRtpTransceiver>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCRtpTransceiver.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
View File

@ -8,13 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Initialize and clean up the SSL library. Failure is fatal. These call the
* corresponding functions in webrtc/rtc_base/ssladapter.h.
*/
RTC_EXTERN BOOL RTCInitializeSSL(void);
RTC_EXTERN BOOL RTCCleanupSSL(void);
#import "api/peerconnection/RTCSSLAdapter.h"

38
sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
View File

@ -8,40 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the session description type. This exposes the same types that are
* in C++, which doesn't include the rollback type that is in the W3C spec.
*/
typedef NS_ENUM(NSInteger, RTCSdpType) {
RTCSdpTypeOffer,
RTCSdpTypePrAnswer,
RTCSdpTypeAnswer,
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCSessionDescription : NSObject
/** The type of session description. */
@property(nonatomic, readonly) RTCSdpType type;
/** The SDP string representation of this session description. */
@property(nonatomic, readonly) NSString *sdp;
- (instancetype)init NS_UNAVAILABLE;
/** Initialize a session description with a type and SDP string. */
- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+ (NSString *)stringForType:(RTCSdpType)type;
+ (RTCSdpType)typeForString:(NSString *)string;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCSessionDescription.h"

12
sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
View File

@ -8,14 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
RTC_EXTERN void RTCSetupInternalTracer(void);
/** Starts capture to specified file. Must be a valid writable path.
* Returns YES if capture starts.
*/
RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
RTC_EXTERN void RTCStopInternalCapture(void);
RTC_EXTERN void RTCShutdownInternalTracer(void);
#import "api/peerconnection/RTCTracing.h"

22
sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
View File

@ -8,24 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCVideoCapturer;
RTC_EXPORT
@protocol RTCVideoCapturerDelegate <NSObject>
- (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame;
@end
RTC_EXPORT
@interface RTCVideoCapturer : NSObject
@property(nonatomic, weak) id<RTCVideoCapturerDelegate> delegate;
- (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoCapturer.h"

186
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
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@ -8,179 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT extern NSString *const kRTCVideoCodecVp8Name;
RTC_EXPORT extern NSString *const kRTCVideoCodecVp9Name;
RTC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
/** Represents an encoded frame's type. */
typedef NS_ENUM(NSUInteger, RTCFrameType) {
RTCFrameTypeEmptyFrame = 0,
RTCFrameTypeAudioFrameSpeech = 1,
RTCFrameTypeAudioFrameCN = 2,
RTCFrameTypeVideoFrameKey = 3,
RTCFrameTypeVideoFrameDelta = 4,
};
typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
RTCVideoContentTypeUnspecified,
RTCVideoContentTypeScreenshare,
};
/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
RTC_EXPORT
@interface RTCEncodedImage : NSObject
@property(nonatomic, strong) NSData *buffer;
@property(nonatomic, assign) int32_t encodedWidth;
@property(nonatomic, assign) int32_t encodedHeight;
@property(nonatomic, assign) uint32_t timeStamp;
@property(nonatomic, assign) int64_t captureTimeMs;
@property(nonatomic, assign) int64_t ntpTimeMs;
@property(nonatomic, assign) uint8_t flags;
@property(nonatomic, assign) int64_t encodeStartMs;
@property(nonatomic, assign) int64_t encodeFinishMs;
@property(nonatomic, assign) RTCFrameType frameType;
@property(nonatomic, assign) RTCVideoRotation rotation;
@property(nonatomic, assign) BOOL completeFrame;
@property(nonatomic, strong) NSNumber *qp;
@property(nonatomic, assign) RTCVideoContentType contentType;
@end
/** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
RTC_EXPORT
@interface RTCRtpFragmentationHeader : NSObject
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationLength;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationTimeDiff;
@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationPlType;
@end
/** Implement this protocol to pass codec specific info from the encoder.
* Corresponds to webrtc::CodecSpecificInfo.
*/
RTC_EXPORT
@protocol RTCCodecSpecificInfo <NSObject>
@end
/** Callback block for encoder. */
typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
id<RTCCodecSpecificInfo> info,
RTCRtpFragmentationHeader *header);
/** Callback block for decoder. */
typedef void (^RTCVideoDecoderCallback)(RTCVideoFrame *frame);
typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
RTCVideoCodecModeRealtimeVideo,
RTCVideoCodecModeScreensharing,
};
/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
RTC_EXPORT
@interface RTCVideoCodecInfo : NSObject <NSCoding>
- (instancetype)init NS_UNAVAILABLE;
- (instancetype)initWithName:(NSString *)name;
- (instancetype)initWithName:(NSString *)name
parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
NS_DESIGNATED_INITIALIZER;
- (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info;
@property(nonatomic, readonly) NSString *name;
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
@end
/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
RTC_EXPORT
@interface RTCVideoEncoderSettings : NSObject
@property(nonatomic, strong) NSString *name;
@property(nonatomic, assign) unsigned short width;
@property(nonatomic, assign) unsigned short height;
@property(nonatomic, assign) unsigned int startBitrate; // kilobits/sec.
@property(nonatomic, assign) unsigned int maxBitrate;
@property(nonatomic, assign) unsigned int minBitrate;
@property(nonatomic, assign) unsigned int targetBitrate;
@property(nonatomic, assign) uint32_t maxFramerate;
@property(nonatomic, assign) unsigned int qpMax;
@property(nonatomic, assign) RTCVideoCodecMode mode;
@end
/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
RTC_EXPORT
@interface RTCVideoEncoderQpThresholds : NSObject
- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
@property(nonatomic, readonly) NSInteger low;
@property(nonatomic, readonly) NSInteger high;
@end
/** Protocol for encoder implementations. */
RTC_EXPORT
@protocol RTCVideoEncoder <NSObject>
- (void)setCallback:(RTCVideoEncoderCallback)callback;
- (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores;
- (NSInteger)releaseEncoder;
- (NSInteger)encode:(RTCVideoFrame *)frame
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
frameTypes:(NSArray<NSNumber *> *)frameTypes;
- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
- (NSString *)implementationName;
/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
* keep the QP from the encoded images within the given range. Returning nil from this function
* disables quality scaling. */
- (RTCVideoEncoderQpThresholds *)scalingSettings;
@end
/** Protocol for decoder implementations. */
RTC_EXPORT
@protocol RTCVideoDecoder <NSObject>
- (void)setCallback:(RTCVideoDecoderCallback)callback;
- (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores
DEPRECATED_MSG_ATTRIBUTE("use startDecodeWithNumberOfCores: instead");
- (NSInteger)releaseDecoder;
- (NSInteger)decode:(RTCEncodedImage *)encodedImage
missingFrames:(BOOL)missingFrames
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
renderTimeMs:(int64_t)renderTimeMs;
- (NSString *)implementationName;
// TODO(andersc): Make non-optional when `startDecodeWithSettings:numberOfCores:` is removed.
@optional
- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
@end
NS_ASSUME_NONNULL_END
#import "api/video_codec/RTCVideoCodecConstants.h"
#import "base/RTCCodecSpecificInfo.h"
#import "base/RTCEncodedImage.h"
#import "base/RTCRtpFragmentationHeader.h"
#import "base/RTCVideoCodecInfo.h"
#import "base/RTCVideoDecoder.h"
#import "base/RTCVideoEncoder.h"
#import "base/RTCVideoEncoderQpThresholds.h"
#import "base/RTCVideoEncoderSettings.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"

48
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
View File

@ -8,47 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
RTC_EXPORT
@protocol RTCVideoEncoderFactory <NSObject>
- (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@end
/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
RTC_EXPORT
@protocol RTCVideoDecoderFactory <NSObject>
- (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@end
#pragma mark - Default factories
/** These codec factories include support for all codecs bundled with WebRTC. If using custom
* codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
*/
RTC_EXPORT
@interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
@property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
+ (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;
@end
RTC_EXPORT
@interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoDecoderFactory.h"
#import "base/RTCVideoEncoderFactory.h"
#import "components/video_codec/RTCDefaultVideoDecoderFactory.h"
#import "components/video_codec/RTCDefaultVideoEncoderFactory.h"

87
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
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@ -8,84 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodecFactory.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_EXPORT
@interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
@end
/** H264 Profiles and levels. */
typedef NS_ENUM(NSUInteger, RTCH264Profile) {
RTCH264ProfileConstrainedBaseline,
RTCH264ProfileBaseline,
RTCH264ProfileMain,
RTCH264ProfileConstrainedHigh,
RTCH264ProfileHigh,
};
typedef NS_ENUM(NSUInteger, RTCH264Level) {
RTCH264Level1_b = 0,
RTCH264Level1 = 10,
RTCH264Level1_1 = 11,
RTCH264Level1_2 = 12,
RTCH264Level1_3 = 13,
RTCH264Level2 = 20,
RTCH264Level2_1 = 21,
RTCH264Level2_2 = 22,
RTCH264Level3 = 30,
RTCH264Level3_1 = 31,
RTCH264Level3_2 = 32,
RTCH264Level4 = 40,
RTCH264Level4_1 = 41,
RTCH264Level4_2 = 42,
RTCH264Level5 = 50,
RTCH264Level5_1 = 51,
RTCH264Level5_2 = 52
};
RTC_EXPORT
@interface RTCH264ProfileLevelId : NSObject
@property(nonatomic, readonly) RTCH264Profile profile;
@property(nonatomic, readonly) RTCH264Level level;
@property(nonatomic, readonly) NSString *hexString;
- (instancetype)initWithHexString:(NSString *)hexString;
- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
@end
/** Encoder. */
RTC_EXPORT
@interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
@end
/** Decoder. */
RTC_EXPORT
@interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
@end
/** Encoder factory. */
RTC_EXPORT
@interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
@end
/** Decoder factory. */
RTC_EXPORT
@interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
@end
#import "components/video_codec/RTCCodecSpecificInfoH264.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoDecoderH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderH264.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecInfo.h Normal file
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@ -0,0 +1,11 @@
/*
* 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 "base/RTCVideoCodecInfo.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoDecoderVP8 : NSObject
/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoDecoder>)vp8Decoder;
@end
#import "api/video_codec/RTCVideoDecoderVP8.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoDecoderVP9 : NSObject
/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoDecoder>)vp9Decoder;
@end
#import "api/video_codec/RTCVideoDecoderVP9.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoEncoderVP8 : NSObject
/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoEncoder>)vp8Encoder;
@end
#import "api/video_codec/RTCVideoEncoderVP8.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
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@ -8,18 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodec.h>
RTC_EXPORT
@interface RTCVideoEncoderVP9 : NSObject
/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTCVideoEncoder>)vp9Encoder;
@end
#import "api/video_codec/RTCVideoEncoderVP9.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
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@ -8,78 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef NS_ENUM(NSInteger, RTCVideoRotation) {
RTCVideoRotation_0 = 0,
RTCVideoRotation_90 = 90,
RTCVideoRotation_180 = 180,
RTCVideoRotation_270 = 270,
};
@protocol RTCVideoFrameBuffer;
// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
RTC_EXPORT
@interface RTCVideoFrame : NSObject
/** Width without rotation applied. */
@property(nonatomic, readonly) int width;
/** Height without rotation applied. */
@property(nonatomic, readonly) int height;
@property(nonatomic, readonly) RTCVideoRotation rotation;
/** Timestamp in nanoseconds. */
@property(nonatomic, readonly) int64_t timeStampNs;
/** Timestamp 90 kHz. */
@property(nonatomic, assign) int32_t timeStamp;
@property(nonatomic, readonly) id<RTCVideoFrameBuffer> buffer;
- (instancetype)init NS_UNAVAILABLE;
- (instancetype) new NS_UNAVAILABLE;
/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
* Deprecated - initialize with a RTCCVPixelBuffer instead
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
* scaling. Cropping will be applied first on the pixel buffer, followed by
* scaling to the final resolution of scaledWidth x scaledHeight.
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
scaledWidth:(int)scaledWidth
scaledHeight:(int)scaledHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
*/
- (instancetype)initWithBuffer:(id<RTCVideoFrameBuffer>)frameBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs;
/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
* the YUV data on it.
*/
- (RTCVideoFrame *)newI420VideoFrame;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoFrame.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
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@ -8,109 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTCI420Buffer;
// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
RTC_EXPORT
@protocol RTCVideoFrameBuffer <NSObject>
@property(nonatomic, readonly) int width;
@property(nonatomic, readonly) int height;
- (id<RTCI420Buffer>)toI420;
@end
/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
@protocol RTCYUVPlanarBuffer <RTCVideoFrameBuffer>
@property(nonatomic, readonly) int chromaWidth;
@property(nonatomic, readonly) int chromaHeight;
@property(nonatomic, readonly) const uint8_t *dataY;
@property(nonatomic, readonly) const uint8_t *dataU;
@property(nonatomic, readonly) const uint8_t *dataV;
@property(nonatomic, readonly) int strideY;
@property(nonatomic, readonly) int strideU;
@property(nonatomic, readonly) int strideV;
- (instancetype)initWithWidth:(int)width
height:(int)height
dataY:(const uint8_t *)dataY
dataU:(const uint8_t *)dataU
dataV:(const uint8_t *)dataV;
- (instancetype)initWithWidth:(int)width height:(int)height;
- (instancetype)initWithWidth:(int)width
height:(int)height
strideY:(int)strideY
strideU:(int)strideU
strideV:(int)strideV;
@end
/** Extension of the YUV planar data buffer with mutable data access */
@protocol RTCMutableYUVPlanarBuffer <RTCYUVPlanarBuffer>
@property(nonatomic, readonly) uint8_t *mutableDataY;
@property(nonatomic, readonly) uint8_t *mutableDataU;
@property(nonatomic, readonly) uint8_t *mutableDataV;
@end
/** Protocol for RTCYUVPlanarBuffers containing I420 data */
@protocol RTCI420Buffer <RTCYUVPlanarBuffer>
@end
/** Extension of the I420 buffer with mutable data access */
@protocol RTCMutableI420Buffer <RTCI420Buffer, RTCMutableYUVPlanarBuffer>
@end
/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
RTC_EXPORT
@interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
@property(nonatomic, readonly) int cropX;
@property(nonatomic, readonly) int cropY;
@property(nonatomic, readonly) int cropWidth;
@property(nonatomic, readonly) int cropHeight;
+ (NSSet<NSNumber *> *)supportedPixelFormats;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
adaptedWidth:(int)adaptedWidth
adaptedHeight:(int)adaptedHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY;
- (BOOL)requiresCropping;
- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
* bufferSizeForCroppingAndScalingToWidth:height: method.
* If that size is 0, the |tmpBuffer| may be nil.
*/
- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
withTempBuffer:(nullable uint8_t *)tmpBuffer;
@end
/** RTCI420Buffer implements the RTCI420Buffer protocol */
RTC_EXPORT
@interface RTCI420Buffer : NSObject <RTCI420Buffer>
@end
/** Mutable version of RTCI420Buffer */
RTC_EXPORT
@interface RTCMutableI420Buffer : RTCI420Buffer <RTCMutableI420Buffer>
@end
NS_ASSUME_NONNULL_END
#import "api/video_frame_buffer/RTCI420Buffer.h"
#import "api/video_frame_buffer/RTCMutableI420Buffer.h"
#import "base/RTCI420Buffer.h"
#import "base/RTCMutableI420Buffer.h"
#import "base/RTCMutableYUVPlanarBuffer.h"
#import "base/RTCVideoFrameBuffer.h"
#import "base/RTCYUVPlanarBuffer.h"
#import "components/video_frame_buffer/RTCCVPixelBuffer.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
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@ -8,33 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#if TARGET_OS_IPHONE
#import <UIKit/UIKit.h>
#endif
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTCVideoFrame;
RTC_EXPORT
@protocol RTCVideoRenderer <NSObject>
/** The size of the frame. */
- (void)setSize:(CGSize)size;
/** The frame to be displayed. */
- (void)renderFrame:(nullable RTCVideoFrame *)frame;
@end
RTC_EXPORT
@protocol RTCVideoViewDelegate
- (void)videoView:(id<RTCVideoRenderer>)videoView didChangeVideoSize:(CGSize)size;
@end
NS_ASSUME_NONNULL_END
#import "base/RTCVideoRenderer.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXPORT
@interface RTCVideoSource : RTCMediaSource <RTCVideoCapturerDelegate>
- (instancetype)init NS_UNAVAILABLE;
/**
* Calling this function will cause frames to be scaled down to the
* requested resolution. Also, frames will be cropped to match the
* requested aspect ratio, and frames will be dropped to match the
* requested fps. The requested aspect ratio is orientation agnostic and
* will be adjusted to maintain the input orientation, so it doesn't
* matter if e.g. 1280x720 or 720x1280 is requested.
*/
- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCVideoSource.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
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@ -8,30 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTCVideoRenderer;
@class RTCPeerConnectionFactory;
@class RTCVideoSource;
RTC_EXPORT
@interface RTCVideoTrack : RTCMediaStreamTrack
/** The video source for this video track. */
@property(nonatomic, readonly) RTCVideoSource *source;
- (instancetype)init NS_UNAVAILABLE;
/** Register a renderer that will render all frames received on this track. */
- (void)addRenderer:(id<RTCVideoRenderer>)renderer;
/** Deregister a renderer. */
- (void)removeRenderer:(id<RTCVideoRenderer>)renderer;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCVideoTrack.h"

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sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
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@ -8,34 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/**
* RTCVideoViewShading provides a way for apps to customize the OpenGL(ES) shaders used in
* rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
*/
RTC_EXPORT
@protocol RTCVideoViewShading <NSObject>
/** Callback for I420 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth:(int)width
height:(int)height
rotation:(RTCVideoRotation)rotation
yPlane:(GLuint)yPlane
uPlane:(GLuint)uPlane
vPlane:(GLuint)vPlane;
/** Callback for NV12 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth:(int)width
height:(int)height
rotation:(RTCVideoRotation)rotation
yPlane:(GLuint)yPlane
uvPlane:(GLuint)uvPlane;
@end
NS_ASSUME_NONNULL_END
#import "components/renderer/opengl/RTCVideoViewShading.h"

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sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
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@ -8,74 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import <UIKit/UIKit.h>
typedef NS_ENUM(NSInteger, RTCDeviceType) {
RTCDeviceTypeUnknown,
RTCDeviceTypeIPhone1G,
RTCDeviceTypeIPhone3G,
RTCDeviceTypeIPhone3GS,
RTCDeviceTypeIPhone4,
RTCDeviceTypeIPhone4Verizon,
RTCDeviceTypeIPhone4S,
RTCDeviceTypeIPhone5GSM,
RTCDeviceTypeIPhone5GSM_CDMA,
RTCDeviceTypeIPhone5CGSM,
RTCDeviceTypeIPhone5CGSM_CDMA,
RTCDeviceTypeIPhone5SGSM,
RTCDeviceTypeIPhone5SGSM_CDMA,
RTCDeviceTypeIPhone6Plus,
RTCDeviceTypeIPhone6,
RTCDeviceTypeIPhone6S,
RTCDeviceTypeIPhone6SPlus,
RTCDeviceTypeIPhone7,
RTCDeviceTypeIPhone7Plus,
RTCDeviceTypeIPhoneSE,
RTCDeviceTypeIPhone8,
RTCDeviceTypeIPhone8Plus,
RTCDeviceTypeIPhoneX,
RTCDeviceTypeIPodTouch1G,
RTCDeviceTypeIPodTouch2G,
RTCDeviceTypeIPodTouch3G,
RTCDeviceTypeIPodTouch4G,
RTCDeviceTypeIPodTouch5G,
RTCDeviceTypeIPodTouch6G,
RTCDeviceTypeIPad,
RTCDeviceTypeIPad2Wifi,
RTCDeviceTypeIPad2GSM,
RTCDeviceTypeIPad2CDMA,
RTCDeviceTypeIPad2Wifi2,
RTCDeviceTypeIPadMiniWifi,
RTCDeviceTypeIPadMiniGSM,
RTCDeviceTypeIPadMiniGSM_CDMA,
RTCDeviceTypeIPad3Wifi,
RTCDeviceTypeIPad3GSM_CDMA,
RTCDeviceTypeIPad3GSM,
RTCDeviceTypeIPad4Wifi,
RTCDeviceTypeIPad4GSM,
RTCDeviceTypeIPad4GSM_CDMA,
RTCDeviceTypeIPad5,
RTCDeviceTypeIPad6,
RTCDeviceTypeIPadAirWifi,
RTCDeviceTypeIPadAirCellular,
RTCDeviceTypeIPadAirWifiCellular,
RTCDeviceTypeIPadAir2,
RTCDeviceTypeIPadMini2GWifi,
RTCDeviceTypeIPadMini2GCellular,
RTCDeviceTypeIPadMini2GWifiCellular,
RTCDeviceTypeIPadMini3,
RTCDeviceTypeIPadMini4,
RTCDeviceTypeIPadPro9Inch,
RTCDeviceTypeIPadPro12Inch,
RTCDeviceTypeIPadPro12Inch2,
RTCDeviceTypeIPadPro10Inch,
RTCDeviceTypeSimulatori386,
RTCDeviceTypeSimulatorx86_64,
};
@interface UIDevice (RTCDevice)
+ (RTCDeviceType)deviceType;
+ (BOOL)isIOS11OrLater;
@end
#import "helpers/UIDevice+RTCDevice.h"