seperate remote desktop from webrtc

This commit is contained in:
Craig 2020-04-05 22:34:51 +00:00
parent 6de731b9bb
commit 26c6cfbe1e
18 changed files with 625 additions and 459 deletions

View file

@ -3,133 +3,80 @@ package webrtc
import (
"fmt"
"io"
"math/rand"
"strings"
"time"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
"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"
"n.eko.moe/neko/internal/xorg"
)
func New(sessions types.SessionManager, config *config.WebRTC) *WebRTCManager {
func New(sessions types.SessionManager, remote types.RemoteManager, config *config.WebRTC) *WebRTCManager {
return &WebRTCManager{
logger: log.With().Str("module", "webrtc").Logger(),
cleanup: time.NewTicker(1 * time.Second),
shutdown: make(chan bool),
remote: remote,
sessions: sessions,
config: config,
}
}
type WebRTCManager struct {
logger zerolog.Logger
videoTrack *webrtc.Track
audioTrack *webrtc.Track
videoPipeline *gst.Pipeline
audioPipeline *gst.Pipeline
videoCodec *webrtc.RTPCodec
audioCodec *webrtc.RTPCodec
sessions types.SessionManager
cleanup *time.Ticker
config *config.WebRTC
shutdown chan bool
logger zerolog.Logger
videoTrack *webrtc.Track
audioTrack *webrtc.Track
videoCodec *webrtc.RTPCodec
audioCodec *webrtc.RTPCodec
sessions types.SessionManager
remote types.RemoteManager
config *config.WebRTC
}
func (m *WebRTCManager) Start() {
// Set display and change to default resolution
xorg.Display(m.config.Display)
if !xorg.ValidScreenSize(m.config.ScreenWidth, m.config.ScreenHeight, m.config.ScreenRate) {
m.logger.Warn().Msgf("invalid screen option %dx%d@%d", m.config.ScreenWidth, m.config.ScreenHeight, m.config.ScreenRate)
} else {
if err := xorg.ChangeScreenSize(m.config.ScreenWidth, m.config.ScreenHeight, m.config.ScreenRate); err != nil {
m.logger.Warn().Err(err).Msg("unable to change screen size")
}
}
func (manager *WebRTCManager) Start() {
var err error
m.videoPipeline, m.videoTrack, m.videoCodec, err = m.createTrack(m.config.VideoCodec, m.config.Display, m.config.VideoParams)
manager.audioTrack, manager.audioCodec, err = manager.createTrack(manager.remote.AudioCodec())
if err != nil {
m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
manager.logger.Panic().Err(err).Msg("unable to create audio track")
}
m.audioPipeline, m.audioTrack, m.audioCodec, err = m.createTrack(m.config.AudioCodec, m.config.Device, m.config.AudioParams)
if err != nil {
m.logger.Panic().Err(err).Msg("unable to start webrtc manager")
}
go func() {
defer func() {
m.logger.Info().Msg("shutdown")
}()
for {
select {
case <-m.shutdown:
return
case sample := <-m.videoPipeline.Sample:
if err := m.videoTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
m.logger.Warn().Err(err).Msg("video pipeline failed to write")
}
case sample := <-m.audioPipeline.Sample:
if err := m.audioTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
m.logger.Warn().Err(err).Msg("audio pipeline failed to write")
}
case <-m.cleanup.C:
xorg.CheckKeys(time.Second * 10)
}
manager.remote.OnAudioFrame(func(sample types.Sample) {
if err := manager.audioTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
manager.logger.Warn().Err(err).Msg("audio pipeline failed to write")
}
}()
m.videoPipeline.Start()
m.audioPipeline.Start()
m.sessions.OnHostCleared(func(id string) {
xorg.ResetKeys()
})
m.sessions.OnCreated(func(id string, session types.Session) {
m.logger.Debug().Str("id", id).Msg("session created")
manager.videoTrack, manager.videoCodec, err = manager.createTrack(manager.remote.VideoCodec())
if err != nil {
manager.logger.Panic().Err(err).Msg("unable to create video track")
}
manager.remote.OnVideoFrame(func(sample types.Sample) {
if err := manager.videoTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
manager.logger.Warn().Err(err).Msg("video pipeline failed to write")
}
})
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).
Str("audio_pipeline_src", m.audioPipeline.Src).
Str("video_pipeline_src", m.videoPipeline.Src).
Str("ice_lite", fmt.Sprintf("%t", m.config.ICELite)).
Str("ice_servers", strings.Join(m.config.ICEServers, ",")).
Str("ephemeral_port_range", fmt.Sprintf("%d-%d", m.config.EphemeralMin, m.config.EphemeralMax)).
Str("nat_ips", strings.Join(m.config.NAT1To1IPs, ",")).
Msgf("webrtc streaming")
manager.logger.Info().
Str("ice_lite", fmt.Sprintf("%t", manager.config.ICELite)).
Str("ice_servers", strings.Join(manager.config.ICEServers, ",")).
Str("ephemeral_port_range", fmt.Sprintf("%d-%d", manager.config.EphemeralMin, manager.config.EphemeralMax)).
Str("nat_ips", strings.Join(manager.config.NAT1To1IPs, ",")).
Msgf("webrtc starting")
}
func (m *WebRTCManager) Shutdown() error {
m.logger.Info().Msgf("webrtc shutting down")
m.videoPipeline.Stop()
m.audioPipeline.Stop()
m.cleanup.Stop()
m.shutdown <- true
func (manager *WebRTCManager) Shutdown() error {
manager.logger.Info().Msgf("webrtc shutting down")
return nil
}
func (m *WebRTCManager) CreatePeer(id string, session types.Session) (string, bool, []string, error) {
func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (string, bool, []string, error) {
configuration := &webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: m.config.ICEServers,
URLs: manager.config.ICEServers,
},
},
SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
@ -137,25 +84,25 @@ func (m *WebRTCManager) CreatePeer(id string, session types.Session) (string, bo
settings := webrtc.SettingEngine{
LoggerFactory: loggerFactory{
logger: m.logger,
logger: manager.logger,
},
}
if m.config.ICELite {
if manager.config.ICELite {
configuration = &webrtc.Configuration{
SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
}
settings.SetLite(true)
}
settings.SetEphemeralUDPPortRange(m.config.EphemeralMin, m.config.EphemeralMax)
settings.SetNAT1To1IPs(m.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
// Create MediaEngine based off sdp
engine := webrtc.MediaEngine{}
// engine.RegisterDefaultCodecs()
engine.RegisterCodec(m.audioCodec)
engine.RegisterCodec(m.videoCodec)
engine.RegisterCodec(manager.audioCodec)
engine.RegisterCodec(manager.videoCodec)
// Create API with MediaEngine and SettingEngine
api := webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(settings))
@ -163,30 +110,30 @@ func (m *WebRTCManager) CreatePeer(id string, session types.Session) (string, bo
// Create new peer connection
connection, err := api.NewPeerConnection(*configuration)
if err != nil {
return "", m.config.ICELite, m.config.ICEServers, err
return "", manager.config.ICELite, manager.config.ICEServers, err
}
if _, err = connection.AddTransceiverFromTrack(m.videoTrack, webrtc.RtpTransceiverInit{
if _, err = connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
Direction: webrtc.RTPTransceiverDirectionSendonly,
}); err != nil {
return "", m.config.ICELite, m.config.ICEServers, err
return "", manager.config.ICELite, manager.config.ICEServers, err
}
if _, err = connection.AddTransceiverFromTrack(m.audioTrack, webrtc.RtpTransceiverInit{
if _, err = connection.AddTransceiverFromTrack(manager.audioTrack, webrtc.RtpTransceiverInit{
Direction: webrtc.RTPTransceiverDirectionSendonly,
}); err != nil {
return "", m.config.ICELite, m.config.ICEServers, err
return "", manager.config.ICELite, manager.config.ICEServers, err
}
description, err := connection.CreateOffer(nil)
if err != nil {
return "", m.config.ICELite, m.config.ICEServers, err
return "", manager.config.ICELite, manager.config.ICEServers, 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")
if err = manager.handle(id, msg); err != nil {
manager.logger.Warn().Err(err).Msg("data handle failed")
}
})
})
@ -196,14 +143,14 @@ func (m *WebRTCManager) CreatePeer(id string, session types.Session) (string, bo
switch state {
case webrtc.PeerConnectionStateDisconnected:
case webrtc.PeerConnectionStateFailed:
m.logger.Info().Str("id", id).Msg("peer disconnected")
m.sessions.Destroy(id)
manager.logger.Info().Str("id", id).Msg("peer disconnected")
manager.sessions.Destroy(id)
break
case webrtc.PeerConnectionStateConnected:
m.logger.Info().Str("id", id).Msg("peer connected")
manager.logger.Info().Str("id", id).Msg("peer connected")
if err = session.SetConnected(true); err != nil {
m.logger.Warn().Err(err).Msg("unable to set connected on peer")
m.sessions.Destroy(id)
manager.logger.Warn().Err(err).Msg("unable to set connected on peer")
manager.sessions.Destroy(id)
}
break
}
@ -213,43 +160,42 @@ func (m *WebRTCManager) CreatePeer(id string, session types.Session) (string, bo
id: id,
api: api,
engine: &engine,
manager: m,
manager: manager,
settings: &settings,
connection: connection,
configuration: configuration,
}); err != nil {
return "", m.config.ICELite, m.config.ICEServers, err
return "", manager.config.ICELite, manager.config.ICEServers, err
}
return description.SDP, m.config.ICELite, m.config.ICEServers, nil
return description.SDP, manager.config.ICELite, manager.config.ICEServers, nil
}
func (m *WebRTCManager) ChangeScreenSize(width int, height int, rate int) error {
if !xorg.ValidScreenSize(width, height, rate) {
return fmt.Errorf("unknown configuration")
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.Track, *webrtc.RTPCodec, error) {
var codec *webrtc.RTPCodec
switch codecName {
case webrtc.VP8:
codec = webrtc.NewRTPVP8Codec(webrtc.DefaultPayloadTypeVP8, 90000)
case webrtc.VP9:
codec = webrtc.NewRTPVP9Codec(webrtc.DefaultPayloadTypeVP9, 90000)
case webrtc.H264:
codec = webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
case webrtc.Opus:
codec = webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000)
case webrtc.G722:
codec = webrtc.NewRTPG722Codec(webrtc.DefaultPayloadTypeG722, 8000)
case webrtc.PCMU:
codec = webrtc.NewRTPPCMUCodec(webrtc.DefaultPayloadTypePCMU, 8000)
case webrtc.PCMA:
codec = webrtc.NewRTPPCMACodec(webrtc.DefaultPayloadTypePCMA, 8000)
default:
return nil, nil, fmt.Errorf("unknown codec %s", codecName)
}
m.videoPipeline.Stop()
defer func() {
m.videoPipeline.Start()
m.logger.Info().Msg("starting pipeline")
}()
if err := xorg.ChangeScreenSize(width, height, rate); err != nil {
return err
}
videoPipeline, err := gst.CreatePipeline(
m.config.VideoCodec,
m.config.Display,
m.config.VideoParams,
)
track, err := webrtc.NewTrack(codec.PayloadType, rand.Uint32(), "stream", "stream", codec)
if err != nil {
m.logger.Panic().Err(err).Msg("unable to create new video pipeline")
return nil, nil, err
}
m.videoPipeline = videoPipeline
return nil
return track, codec, nil
}