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:
42
server/vendor/github.com/pion/rtcp/doc.go
generated
vendored
Normal file
42
server/vendor/github.com/pion/rtcp/doc.go
generated
vendored
Normal file
@@ -0,0 +1,42 @@
|
||||
// SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
|
||||
// SPDX-License-Identifier: MIT
|
||||
|
||||
/*
|
||||
Package rtcp implements encoding and decoding of RTCP packets according to RFCs 3550 and 5506.
|
||||
|
||||
RTCP is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality
|
||||
and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control
|
||||
information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data,
|
||||
but does not transport any media data itself.
|
||||
|
||||
The primary function of RTCP is to provide feedback on the quality of service (QoS)
|
||||
in media distribution by periodically sending statistics information such as transmitted octet
|
||||
and packet counts, packet loss, packet delay variation, and round-trip delay time to participants
|
||||
in a streaming multimedia session. An application may use this information to control quality of
|
||||
service parameters, perhaps by limiting flow, or using a different codec.
|
||||
|
||||
Decoding RTCP packets:
|
||||
|
||||
pkts, err := rtcp.Unmarshal(rtcpData)
|
||||
// ...
|
||||
for _, pkt := range pkts {
|
||||
switch p := pkt.(type) {
|
||||
case *rtcp.CompoundPacket:
|
||||
...
|
||||
case *rtcp.PictureLossIndication:
|
||||
...
|
||||
default:
|
||||
...
|
||||
}
|
||||
}
|
||||
|
||||
Encoding RTCP packets:
|
||||
|
||||
pkt := &rtcp.PictureLossIndication{
|
||||
SenderSSRC: senderSSRC,
|
||||
MediaSSRC: mediaSSRC
|
||||
}
|
||||
pliData, err := pkt.Marshal()
|
||||
// ...
|
||||
*/
|
||||
package rtcp
|
||||
Reference in New Issue
Block a user