neko/server/internal/webrtc/webrtc.go

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package webrtc
import (
"fmt"
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"strings"
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"time"
"github.com/pion/webrtc/v2"
"github.com/rs/zerolog"
"github.com/rs/zerolog/log"
"n.eko.moe/neko/internal/gst"
"n.eko.moe/neko/internal/types"
"n.eko.moe/neko/internal/types/config"
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"n.eko.moe/neko/internal/xorg"
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)
func New(sessions types.SessionManager, config *config.WebRTC) *WebRTCManager {
logger := log.With().Str("module", "webrtc").Logger()
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settings := webrtc.SettingEngine{
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LoggerFactory: loggerFactory{
logger: logger,
},
}
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settings.SetLite(true)
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settings.SetNetworkTypes([]webrtc.NetworkType{webrtc.NetworkTypeUDP4})
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settings.SetEphemeralUDPPortRange(config.EphemeralMin, config.EphemeralMax)
settings.SetNAT1To1IPs(config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
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return &WebRTCManager{
logger: logger,
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settings: settings,
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cleanup: time.NewTicker(1 * time.Second),
shutdown: make(chan bool),
sessions: sessions,
config: config,
configuration: &webrtc.Configuration{
SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
},
}
}
type WebRTCManager struct {
logger zerolog.Logger
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settings webrtc.SettingEngine
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sessions types.SessionManager
videoPipeline *gst.Pipeline
audioPipeline *gst.Pipeline
cleanup *time.Ticker
config *config.WebRTC
shutdown chan bool
configuration *webrtc.Configuration
}
func (m *WebRTCManager) Start() {
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xorg.Display(m.config.Display)
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videoPipeline, err := gst.CreatePipeline(
m.config.VideoCodec,
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m.config.Display,
m.config.VideoParams,
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)
if err != nil {
m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
}
audioPipeline, err := gst.CreatePipeline(
m.config.AudioCodec,
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m.config.Device,
m.config.AudioParams,
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)
if err != nil {
m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
}
m.videoPipeline = videoPipeline
m.audioPipeline = audioPipeline
videoPipeline.Start()
audioPipeline.Start()
go func() {
defer func() {
m.logger.Info().Msg("shutdown")
}()
for {
select {
case <-m.shutdown:
return
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case sample := <-m.videoPipeline.Sample:
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if err := m.sessions.WriteVideoSample(sample); err != nil {
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m.logger.Warn().Err(err).Msg("video pipeline failed to write")
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}
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case sample := <-m.audioPipeline.Sample:
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if err := m.sessions.WriteAudioSample(sample); err != nil {
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m.logger.Warn().Err(err).Msg("audio pipeline failed to write")
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}
case <-m.cleanup.C:
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xorg.CheckKeys(time.Second * 10)
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}
}
}()
m.sessions.OnHostCleared(func(id string) {
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xorg.ResetKeys()
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})
m.sessions.OnCreated(func(id string, session types.Session) {
m.logger.Debug().Str("id", id).Msg("session created")
})
m.sessions.OnDestroy(func(id string) {
m.logger.Debug().Str("id", id).Msg("session destroyed")
})
// TODO: log resolution, bit rate and codec parameters
m.logger.Info().
Str("video_display", m.config.Display).
Str("video_codec", m.config.VideoCodec).
Str("audio_device", m.config.Device).
Str("audio_codec", m.config.AudioCodec).
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Str("ephemeral_port_range", fmt.Sprintf("%d-%d", m.config.EphemeralMin, m.config.EphemeralMax)).
Str("nat_ips", strings.Join(m.config.NAT1To1IPs, ",")).
Str("audio_pipeline_src", audioPipeline.Src).
Str("video_pipeline_src", videoPipeline.Src).
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Msgf("webrtc streaming")
}
func (m *WebRTCManager) Shutdown() error {
m.logger.Info().Msgf("webrtc shutting down")
m.videoPipeline.Stop()
m.audioPipeline.Stop()
m.cleanup.Stop()
m.shutdown <- true
return nil
}
func (m *WebRTCManager) CreatePeer(id string, sdp string) (string, types.Peer, error) {
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// Create SessionDescription
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description := webrtc.SessionDescription{
SDP: sdp,
Type: webrtc.SDPTypeOffer,
}
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// Create MediaEngine based off sdp
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engine := webrtc.MediaEngine{}
engine.PopulateFromSDP(description)
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// Create API with MediaEngine and SettingEngine
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api := webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(m.settings))
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// Create new peer connection
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connection, err := api.NewPeerConnection(*m.configuration)
if err != nil {
return "", nil, err
}
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// Create video track
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video, err := m.createVideoTrack(engine)
if err != nil {
return "", nil, err
}
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_, err = connection.AddTransceiverFromTrack(video, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
})
if err != nil {
return "", nil, err
}
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// Create audio track
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audio, err := m.createAudioTrack(engine)
if err != nil {
return "", nil, err
}
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_, err = connection.AddTransceiverFromTrack(audio, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
})
if err != nil {
return "", nil, err
}
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// Set remote description
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connection.SetRemoteDescription(description)
answer, err := connection.CreateAnswer(nil)
if err != nil {
return "", nil, err
}
if err = connection.SetLocalDescription(answer); err != nil {
return "", nil, err
}
connection.OnDataChannel(func(d *webrtc.DataChannel) {
d.OnMessage(func(msg webrtc.DataChannelMessage) {
if err = m.handle(id, msg); err != nil {
m.logger.Warn().Err(err).Msg("data handle failed")
}
})
})
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
switch state {
case webrtc.PeerConnectionStateDisconnected:
case webrtc.PeerConnectionStateFailed:
m.logger.Info().Str("id", id).Msg("peer disconnected")
m.sessions.Destroy(id)
break
case webrtc.PeerConnectionStateConnected:
m.logger.Info().Str("id", id).Msg("peer connected")
break
}
})
return answer.SDP, &Peer{
id: id,
api: api,
engine: engine,
video: video,
audio: audio,
connection: connection,
}, nil
}
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func (m *WebRTCManager) ChangeScreenSize(width int, height int, rate int) error {
m.videoPipeline.Stop()
if err := xorg.ChangeScreenSize(width, height, rate); err != nil {
return err
}
videoPipeline, err := gst.CreatePipeline(
m.config.VideoCodec,
m.config.Display,
m.config.VideoParams,
)
if err != nil {
m.logger.Panic().Err(err).Msg("unable to create new video pipeline")
}
m.videoPipeline = videoPipeline
m.videoPipeline.Start()
return nil
}