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neko-custom/server/internal/webrtc/webrtc.go
2021-02-14 16:30:24 +00:00

271 lines
8.7 KiB
Go

package webrtc
import (
"encoding/json"
"fmt"
"io"
"strings"
"github.com/pion/interceptor"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
"github.com/rs/zerolog"
"github.com/rs/zerolog/log"
"n.eko.moe/neko/internal/types"
"n.eko.moe/neko/internal/types/config"
)
func New(sessions types.SessionManager, remote types.RemoteManager, config *config.WebRTC) *WebRTCManager {
return &WebRTCManager{
logger: log.With().Str("module", "webrtc").Logger(),
remote: remote,
sessions: sessions,
config: config,
}
}
type WebRTCManager struct {
logger zerolog.Logger
videoTrack *webrtc.TrackLocalStaticSample
audioTrack *webrtc.TrackLocalStaticSample
videoCodec webrtc.RTPCodecParameters
audioCodec webrtc.RTPCodecParameters
sessions types.SessionManager
remote types.RemoteManager
config *config.WebRTC
}
func (manager *WebRTCManager) Start() {
var err error
manager.audioTrack, manager.audioCodec, err = manager.createTrack(manager.remote.AudioCodec())
if err != nil {
manager.logger.Panic().Err(err).Msg("unable to create audio track")
}
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")
}
})
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")
}
})
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 (manager *WebRTCManager) Shutdown() error {
manager.logger.Info().Msgf("webrtc shutting down")
return nil
}
func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (string, bool, []string, error) {
configuration := &webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: manager.config.ICEServers,
},
},
SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
}
settings := webrtc.SettingEngine{
LoggerFactory: loggerFactory{
logger: manager.logger,
},
}
if manager.config.ICELite {
configuration = &webrtc.Configuration{
SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
}
settings.SetLite(true)
}
settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
settings.SetSRTPReplayProtectionWindow(512)
// Create MediaEngine based off sdp
engine := webrtc.MediaEngine{}
engine.RegisterCodec(manager.audioCodec, webrtc.RTPCodecTypeAudio)
engine.RegisterCodec(manager.videoCodec, webrtc.RTPCodecTypeVideo)
i := &interceptor.Registry{}
if err := webrtc.RegisterDefaultInterceptors(&engine, i); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
// Create API with MediaEngine and SettingEngine
api := webrtc.NewAPI(webrtc.WithMediaEngine(&engine), webrtc.WithSettingEngine(settings), webrtc.WithInterceptorRegistry(i))
// Create new peer connection
connection, err := api.NewPeerConnection(*configuration)
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
negotiated := true
connection.CreateDataChannel("data", &webrtc.DataChannelInit{
Negotiated: &negotiated,
})
connection.OnDataChannel(func(d *webrtc.DataChannel) {
d.OnMessage(func(msg webrtc.DataChannelMessage) {
if err = manager.handle(id, msg); err != nil {
manager.logger.Warn().Err(err).Msg("data handle failed")
}
})
})
// Set the handler for ICE connection state
// This will notify you when the peer has connected/disconnected
connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
fmt.Printf("Connection State has changed %s \n", connectionState.String())
})
rtpSender, viderr := connection.AddTrack(manager.videoTrack)
if viderr != nil {
return "", manager.config.ICELite, manager.config.ICEServers, viderr
}
if _, err = connection.AddTrack(manager.audioTrack); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
description, err := connection.CreateOffer(nil)
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
err = connection.SetLocalDescription(description)
if err != nil {
panic(err)
}
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
switch state {
case webrtc.PeerConnectionStateDisconnected:
case webrtc.PeerConnectionStateFailed:
manager.logger.Info().Str("id", id).Msg("peer disconnected")
manager.sessions.Destroy(id)
break
case webrtc.PeerConnectionStateConnected:
manager.logger.Info().Str("id", id).Msg("peer connected")
if err = session.SetConnected(true); err != nil {
manager.logger.Warn().Err(err).Msg("unable to set connected on peer")
manager.sessions.Destroy(id)
}
break
}
})
connection.OnICECandidate(func(i *webrtc.ICECandidate) {
if i != nil {
candidateString, err := json.Marshal(i.ToJSON())
if err != nil {
manager.logger.Info().Msg("error")
return
}
if err = session.SignalCandidate(string(candidateString));err != nil {
manager.logger.Info().Msg("err")
return
}
}
})
// Read incoming RTCP packets
// Before these packets are retuned they are processed by interceptors. For things
// like NACK this needs to be called.
go func() {
rtcpBuf := make([]byte, 1500)
for {
n, _, rtcpErr := rtpSender.Read(rtcpBuf)
if rtcpErr != nil {
return
}
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
if err != nil {
log.Printf("Unmarshal RTCP: %v", err)
continue
}
for _, p := range ps {
switch p.(type) {
case *rtcp.TransportLayerNack:
manager.logger.Info().Msg("got a nack")
}
}
}
}()
if err := session.SetPeer(&Peer{
id: id,
api: api,
engine: &engine,
manager: manager,
settings: &settings,
connection: connection,
configuration: configuration,
}); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
return description.SDP, manager.config.ICELite, manager.config.ICEServers, nil
}
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
var codec webrtc.RTPCodecParameters
var fb []webrtc.RTCPFeedback
var fba []webrtc.RTCPFeedback
fb = []webrtc.RTCPFeedback{
{"goog-remb", ""},
{"nack", ""},
{"nack", "pli"},
{"ccm", "fir"},
}
fba = []webrtc.RTCPFeedback{}
switch codecName {
case "VP8":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96,}
case "VP9":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98,}
case "H264":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/H264", ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", RTCPFeedback: fb}, PayloadType: 102,}
case "Opus":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111,}
case "G722":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 9,}
case "PCMU":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 0,}
case "PCMA":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8,}
default:
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
}
track, err := webrtc.NewTrackLocalStaticSample(codec.RTPCodecCapability, "stream", "stream")
if err != nil {
return nil, codec, err
}
return track, codec, nil
}