apps/plugin-rtsp
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适配引擎修改

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dexter
2023-02-26 10:38:53 +08:00
parent 344ab6f463
commit beda2dd9e0
3 changed files with 5 additions and 5 deletions
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2
README.md
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@@ -24,7 +24,7 @@ rtsp://localhost/live/test
例如通过ffmpeg向m7s进行推流 例如通过ffmpeg向m7s进行推流
```bash ```bash
ffmpeg -i [视频源] -c:v h264 -f rtsp rtsp://localhost/live/test ffmpeg -i [视频源] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
``` ```
会在m7s内部形成一个名为live/test的流 会在m7s内部形成一个名为live/test的流

4
client.go
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@@ -92,9 +92,9 @@ type RTSPPusher struct {
func (p *RTSPPusher) OnEvent(event any) { func (p *RTSPPusher) OnEvent(event any) {
switch v := event.(type) { switch v := event.(type) {
case engine.VideoRTP: case engine.VideoRTP:
p.Client.WritePacketRTP(p.videoTrack, &v.Packet) p.Client.WritePacketRTP(p.videoTrack, v.Packet)
case engine.AudioRTP: case engine.AudioRTP:
p.Client.WritePacketRTP(p.audioTrack, &v.Packet) p.Client.WritePacketRTP(p.audioTrack, v.Packet)
default: default:
p.RTSPSubscriber.OnEvent(event) p.RTSPSubscriber.OnEvent(event)
} }

4
subscriber.go
View File

@@ -91,9 +91,9 @@ func (s *RTSPSubscriber) OnEvent(event any) {
case ISubscriber: case ISubscriber:
s.stream = gortsplib.NewServerStream(s.tracks) s.stream = gortsplib.NewServerStream(s.tracks)
case VideoRTP: case VideoRTP:
s.stream.WritePacketRTP(s.videoTrack, &v.Packet) s.stream.WritePacketRTP(s.videoTrack, v.Packet)
case AudioRTP: case AudioRTP:
s.stream.WritePacketRTP(s.audioTrack, &v.Packet) s.stream.WritePacketRTP(s.audioTrack, v.Packet)
default: default:
s.Subscriber.OnEvent(event) s.Subscriber.OnEvent(event)
} }