mirror of
https://github.com/nestriness/nestri.git
synced 2025-12-12 16:55:37 +02:00
This adds: - [x] Keyboard and mouse handling on the frontend - [x] Video and audio streaming from the backend to the frontend - [x] Input server that works with Websockets Update - 17/11 - [ ] Master docker container to run this - [ ] Steam runtime - [ ] Entrypoint.sh --------- Co-authored-by: Kristian Ollikainen <14197772+DatCaptainHorse@users.noreply.github.com> Co-authored-by: Kristian Ollikainen <DatCaptainHorse@users.noreply.github.com>
101 lines
2.8 KiB
Go
101 lines
2.8 KiB
Go
package relay
|
|
|
|
import (
|
|
"github.com/pion/interceptor"
|
|
"github.com/pion/webrtc/v4"
|
|
"log"
|
|
)
|
|
|
|
var globalWebRTCAPI *webrtc.API
|
|
var globalWebRTCConfig = webrtc.Configuration{
|
|
ICETransportPolicy: webrtc.ICETransportPolicyAll,
|
|
BundlePolicy: webrtc.BundlePolicyBalanced,
|
|
SDPSemantics: webrtc.SDPSemanticsUnifiedPlan,
|
|
}
|
|
|
|
func InitWebRTCAPI() error {
|
|
var err error
|
|
flags := GetFlags()
|
|
|
|
// Media engine
|
|
mediaEngine := &webrtc.MediaEngine{}
|
|
|
|
// Default codecs cover most of our needs
|
|
err = mediaEngine.RegisterDefaultCodecs()
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
// Add H.265 for special cases
|
|
videoRTCPFeedback := []webrtc.RTCPFeedback{{"goog-remb", ""}, {"ccm", "fir"}, {"nack", ""}, {"nack", "pli"}}
|
|
for _, codec := range []webrtc.RTPCodecParameters{
|
|
{
|
|
RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH265, ClockRate: 90000, RTCPFeedback: videoRTCPFeedback},
|
|
PayloadType: 48,
|
|
},
|
|
{
|
|
RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeRTX, ClockRate: 90000, SDPFmtpLine: "apt=48"},
|
|
PayloadType: 49,
|
|
},
|
|
} {
|
|
if err := mediaEngine.RegisterCodec(codec, webrtc.RTPCodecTypeVideo); err != nil {
|
|
return err
|
|
}
|
|
}
|
|
|
|
// Interceptor registry
|
|
interceptorRegistry := &interceptor.Registry{}
|
|
|
|
// Use default set
|
|
err = webrtc.RegisterDefaultInterceptors(mediaEngine, interceptorRegistry)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
// Setting engine
|
|
settingEngine := webrtc.SettingEngine{}
|
|
|
|
// New in v4, reduces CPU usage and latency when enabled
|
|
settingEngine.EnableSCTPZeroChecksum(true)
|
|
|
|
// Set the UDP port range used by WebRTC
|
|
err = settingEngine.SetEphemeralUDPPortRange(uint16(flags.WebRTCUDPStart), uint16(flags.WebRTCUDPEnd))
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
// Create a new API object with our customized settings
|
|
globalWebRTCAPI = webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine), webrtc.WithSettingEngine(settingEngine), webrtc.WithInterceptorRegistry(interceptorRegistry))
|
|
|
|
return nil
|
|
}
|
|
|
|
// GetWebRTCAPI returns the global WebRTC API
|
|
func GetWebRTCAPI() *webrtc.API {
|
|
return globalWebRTCAPI
|
|
}
|
|
|
|
// CreatePeerConnection sets up a new peer connection
|
|
func CreatePeerConnection(onClose func()) (*webrtc.PeerConnection, error) {
|
|
pc, err := globalWebRTCAPI.NewPeerConnection(globalWebRTCConfig)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
// Log connection state changes and handle failed/disconnected connections
|
|
pc.OnConnectionStateChange(func(connectionState webrtc.PeerConnectionState) {
|
|
// Close PeerConnection in cases
|
|
if connectionState == webrtc.PeerConnectionStateFailed ||
|
|
connectionState == webrtc.PeerConnectionStateDisconnected ||
|
|
connectionState == webrtc.PeerConnectionStateClosed {
|
|
err := pc.Close()
|
|
if err != nil {
|
|
log.Printf("Error closing PeerConnection: %s\n", err.Error())
|
|
}
|
|
onClose()
|
|
}
|
|
})
|
|
|
|
return pc, nil
|
|
}
|