✨ feat: Add streaming support (#125)

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>

packages/relay/internal/common.go

@@ -0,0 +1,100 @@
+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
+}