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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h | 86 |
1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h new file mode 100644 index 0000000000..1e407fd71b --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h @@ -0,0 +1,86 @@ +/* + * Copyright 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <Foundation/Foundation.h> + +#import "RTCMacros.h" +#import "RTCMediaStreamTrack.h" +#import "RTCRtpParameters.h" + +NS_ASSUME_NONNULL_BEGIN + +/** Represents the media type of the RtpReceiver. */ +typedef NS_ENUM(NSInteger, RTCRtpMediaType) { + RTCRtpMediaTypeAudio, + RTCRtpMediaTypeVideo, + RTCRtpMediaTypeData, + RTCRtpMediaTypeUnsupported, +}; + +@class RTC_OBJC_TYPE(RTCRtpReceiver); + +RTC_OBJC_EXPORT +@protocol RTC_OBJC_TYPE +(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 + : (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType + : (RTCRtpMediaType)mediaType; + +@end + +RTC_OBJC_EXPORT +@protocol RTC_OBJC_TYPE +(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) RTC_OBJC_TYPE(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) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track; + +/** The delegate for this RtpReceiver. */ +@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate; + +@end + +RTC_OBJC_EXPORT +@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)> + +- (instancetype)init NS_UNAVAILABLE; + +@end + +NS_ASSUME_NONNULL_END |