mirror of
https://github.com/sipeed/NanoKVM.git
synced 2026-09-11 00:22:56 -05:00
0e30a26db4
- H.264 WebRTC: Refactored to significantly reduce video latency - H.264 Direct: Optimized data transmission; data parsing now continues correctly even when the tab is in the background - MJPEG: Refactored to ensure the correct data length is sent Bug Fixes - Fixed an issue where certain keyboard modifier keys were not recognized - Fixed vertical mouse cursor drift when the page is zoomed in or out Optimizations - Optimized HID write logic and updated the HID reset mechanism - Added support for deleting images - Added a Swap Memory option to the Tailscale page - Added a confirmation dialog when uninstalling Tailscale to prevent accidental removal - Improved the logic for updating the web page title - Improved the UI for the Clipboard, Settings, Image Mounting, and App Update pages Security - Added a mandatory delay after failed login attempts to prevent brute-force attacks - Updated dependencies to patch known security vulnerabilities
113 lines
2.1 KiB
Go
113 lines
2.1 KiB
Go
package webrtc
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import (
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"encoding/json"
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"errors"
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"github.com/gorilla/websocket"
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"github.com/pion/rtp"
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"github.com/pion/rtp/codecs"
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"github.com/pion/webrtc/v4"
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log "github.com/sirupsen/logrus"
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"sync"
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)
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func NewClient(ws *websocket.Conn, videoConn *webrtc.PeerConnection) *Client {
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return &Client{
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ws: ws,
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video: videoConn,
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mutex: sync.Mutex{},
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}
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}
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func (c *Client) WriteMessage(event string, data string) error {
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c.mutex.Lock()
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defer c.mutex.Unlock()
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message := &Message{
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Event: event,
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Data: data,
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}
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if err := c.ws.WriteJSON(message); err != nil {
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log.Errorf("failed to send message %s: %v", event, err)
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return err
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}
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log.Debugf("sent message %s", event)
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return nil
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}
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func (c *Client) ReadMessage() (*Message, error) {
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_, raw, err := c.ws.ReadMessage()
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if err != nil {
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log.Errorf("failed to read message: %v", err)
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return nil, err
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}
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var message Message
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if err := json.Unmarshal(raw, &message); err != nil {
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log.Errorf("failed to unmarshal message: %v", err)
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return nil, nil
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}
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return &message, nil
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}
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func (c *Client) AddTrack() error {
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// video track
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videoTrack, err := webrtc.NewTrackLocalStaticRTP(
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webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264},
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"video",
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"pion-video",
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)
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if err != nil {
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log.Errorf("failed to create video track: %s", err)
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return err
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}
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videoPacketizer := rtp.NewPacketizer(
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1200,
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100,
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0x1234ABCD,
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&codecs.H264Payloader{},
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rtp.NewRandomSequencer(),
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90000,
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)
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if videoPacketizer == nil {
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err := errors.New("failed to create rtp packetizer")
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log.Error(err)
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return err
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}
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videoSender, err := c.video.AddTrack(videoTrack)
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if err != nil {
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log.Errorf("failed to add video track: %s", err)
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return err
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}
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go startRTCPReader(videoSender)
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track := &Track{
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videoPacketizer: videoPacketizer,
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video: videoTrack,
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}
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track.updateExtension()
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c.mutex.Lock()
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c.track = track
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c.mutex.Unlock()
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return nil
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}
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func startRTCPReader(sender *webrtc.RTPSender) {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, err := sender.Read(rtcpBuf); err != nil {
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log.Debugf("RTCP reader error: %v", err)
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return
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}
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}
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}
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