apps/plugin-webrtc
1
0
Fork 0
mirror of https://github.com/Monibuca/plugin-webrtc.git synced 2025-10-05 14:56:56 +08:00
Code Issues Actions Packages Projects Releases Wiki Activity

加入测试页面

Browse Source
This commit is contained in:
dexter
2023-02-21 21:29:53 +08:00
parent 5526f1e44c
commit 89ebf28810
3 changed files with 70 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
README.md
View File

@@ -50,6 +50,11 @@ Body: `SDP`
Content-Type: `application/sdp`
Response Body: `SDP`
### 推流测试页面
`/webrtc/test/publish`
## WHIP
WebRTC-HTTP ingestion protocol
用于WebRTC交换SDP信息的规范

11
main.go
View File

@@ -10,8 +10,9 @@ import (
"go.uber.org/zap"
"m7s.live/engine/v4"
"github.com/pion/interceptor"
_ "embed"
"github.com/pion/interceptor"
. "github.com/pion/webrtc/v3"
"m7s.live/engine/v4/config"
"m7s.live/plugin/webrtc/v4/webrtc"
@@ -43,6 +44,10 @@ import (
// conn *net.UDPConn
// port int
// }
//go:embed publish.html
var publishHTML []byte
var (
reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
)
@@ -168,6 +173,10 @@ func (conf *WebRTCConfig) Push_(w http.ResponseWriter, r *http.Request) {
}
}
func (conf *WebRTCConfig) Test_Publish(w http.ResponseWriter, r *http.Request) {
w.Write(publishHTML)
}
var webrtcConfig WebRTCConfig
var WebRTCPlugin = engine.InstallPlugin(&webrtcConfig)

55
publish.html Normal file
View File

@@ -0,0 +1,55 @@
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta http-equiv="X-UA-Compatible" content="IE=edge">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<title>测试WebRTC推流</title>
</head>
<body>
<video id="video" width="640" height="480" autoplay muted />
</body>
<script>
(async () => {
const mediaStream = await navigator.mediaDevices.getUserMedia({
video: true,
audio: true,
});
document.getElementById('video').srcObject = mediaStream;
const pc = new RTCPeerConnection();
pc.addTransceiver('video', { direction: 'sendonly' });
pc.addTransceiver('audio', { direction: 'sendonly' });
pc.oniceconnectionstatechange = () => {
console.log('oniceconnectionstatechange', pc.iceConnectionState);
};
pc.onicecandidate = (e) => {
console.log('onicecandidate', e.candidate);
};
mediaStream.getTracks().forEach((t) => {
pc.addTrack(t, mediaStream);
});
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
const result = await fetch(
`/webrtc/push/live/webrtc`,
{
method: 'POST',
mode: 'cors',
cache: 'no-cache',
credentials: 'include',
redirect: 'follow',
referrerPolicy: 'no-referrer',
headers: { 'Content-Type': 'application/sdp' },
body: offer.sdp,
},
);
const remoteSdp = await result.text();
await pc.setRemoteDescription(
new RTCSessionDescription({ type: 'answer', sdp: remoteSdp }),
);
})()
</script>
</html>