直播:后台 JWT 推流、前台画中画;WebRTC 服务与 Nginx WebSocket 代理
Made-with: Cursor
This commit is contained in:
20
server/vendor/github.com/pion/webrtc/v3/webrtc.go
generated
vendored
Normal file
20
server/vendor/github.com/pion/webrtc/v3/webrtc.go
generated
vendored
Normal file
@@ -0,0 +1,20 @@
|
||||
// SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
|
||||
// SPDX-License-Identifier: MIT
|
||||
|
||||
// Package webrtc implements the WebRTC 1.0 as defined in W3C WebRTC specification document.
|
||||
package webrtc
|
||||
|
||||
// SSRC represents a synchronization source
|
||||
// A synchronization source is a randomly chosen
|
||||
// value meant to be globally unique within a particular
|
||||
// RTP session. Used to identify a single stream of media.
|
||||
//
|
||||
// https://tools.ietf.org/html/rfc3550#section-3
|
||||
type SSRC uint32
|
||||
|
||||
// PayloadType identifies the format of the RTP payload and determines
|
||||
// its interpretation by the application. Each codec in a RTP Session
|
||||
// will have a different PayloadType
|
||||
//
|
||||
// https://tools.ietf.org/html/rfc3550#section-3
|
||||
type PayloadType uint8
|
||||
Reference in New Issue
Block a user