fix logging for WebRTC.
This commit is contained in:
parent
8ef91be6ad
commit
29fc67aff9
@ -120,10 +120,15 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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if err != nil {
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if err != nil {
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return "", manager.config.ICELite, manager.config.ICEServers, err
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return "", manager.config.ICELite, manager.config.ICEServers, err
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}
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}
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negotiated := true
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negotiated := true
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connection.CreateDataChannel("data", &webrtc.DataChannelInit{
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_, err = connection.CreateDataChannel("data", &webrtc.DataChannelInit{
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Negotiated: &negotiated,
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Negotiated: &negotiated,
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})
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})
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if err != nil {
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return "", manager.config.ICELite, manager.config.ICEServers, err
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}
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connection.OnDataChannel(func(d *webrtc.DataChannel) {
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connection.OnDataChannel(func(d *webrtc.DataChannel) {
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d.OnMessage(func(msg webrtc.DataChannelMessage) {
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d.OnMessage(func(msg webrtc.DataChannelMessage) {
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if err = manager.handle(id, msg); err != nil {
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if err = manager.handle(id, msg); err != nil {
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@ -135,7 +140,9 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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// Set the handler for ICE connection state
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// Set the handler for ICE connection state
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// This will notify you when the peer has connected/disconnected
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// This will notify you when the peer has connected/disconnected
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connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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fmt.Printf("Connection State has changed %s \n", connectionState.String())
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manager.logger.Info().
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Str("connection_state", connectionState.String()).
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Msg("connection state has changed")
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})
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})
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rtpSender, viderr := connection.AddTrack(manager.videoTrack)
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rtpSender, viderr := connection.AddTrack(manager.videoTrack)
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@ -154,7 +161,7 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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err = connection.SetLocalDescription(description)
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err = connection.SetLocalDescription(description)
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if err != nil {
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if err != nil {
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panic(err)
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return "", manager.config.ICELite, manager.config.ICEServers, err
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}
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}
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connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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@ -175,46 +182,50 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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})
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})
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connection.OnICECandidate(func(i *webrtc.ICECandidate) {
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connection.OnICECandidate(func(i *webrtc.ICECandidate) {
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if i != nil {
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if i == nil {
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candidateString, err := json.Marshal(i.ToJSON())
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manager.logger.Info().Msg("sent all ICECandidates")
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if err != nil {
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return
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manager.logger.Info().Msg("error")
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}
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return
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}
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if err = session.SignalCandidate(string(candidateString));err != nil {
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candidateString, err := json.Marshal(i.ToJSON())
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manager.logger.Info().Msg("err")
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if err != nil {
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return
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manager.logger.Warn().Err(err).Msg("converting ICECandidate to json failed")
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}
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return
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}
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if err := session.SignalCandidate(string(candidateString)); err != nil {
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manager.logger.Warn().Err(err).Msg("sending SignalCandidate failed")
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return
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}
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}
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})
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})
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// Read incoming RTCP packets
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// Read incoming RTCP packets
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// Before these packets are retuned they are processed by interceptors. For things
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// Before these packets are retuned they are processed by interceptors. For things
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// like NACK this needs to be called.
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// like NACK this needs to be called.
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go func() {
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go func() {
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rtcpBuf := make([]byte, 1500)
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rtcpBuf := make([]byte, 1500)
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for {
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for {
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n, _, rtcpErr := rtpSender.Read(rtcpBuf)
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n, _, err := rtpSender.Read(rtcpBuf)
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if rtcpErr != nil {
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if err != nil {
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return
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return
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}
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}
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ps, err := rtcp.Unmarshal(rtcpBuf[:n])
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ps, err := rtcp.Unmarshal(rtcpBuf[:n])
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if err != nil {
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if err != nil {
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log.Printf("Unmarshal RTCP: %v", err)
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manager.logger.Warn().Err(err).Msg("unmarshal RTCP failed")
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continue
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continue
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}
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}
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for _, p := range ps {
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for _, p := range ps {
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switch p.(type) {
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switch p.(type) {
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case *rtcp.TransportLayerNack:
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case *rtcp.TransportLayerNack:
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manager.logger.Info().Msg("got a nack")
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manager.logger.Warn().Msg("got a nack")
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}
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}
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}
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}
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}
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}
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}()
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}()
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if err := session.SetPeer(&Peer{
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if err := session.SetPeer(&Peer{
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id: id,
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id: id,
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api: api,
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api: api,
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@ -232,31 +243,30 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
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func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
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var codec webrtc.RTPCodecParameters
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var codec webrtc.RTPCodecParameters
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var fb []webrtc.RTCPFeedback
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var fba []webrtc.RTCPFeedback
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fba := []webrtc.RTCPFeedback{}
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fb = []webrtc.RTCPFeedback{
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fbv := []webrtc.RTCPFeedback{
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{"goog-remb", ""},
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{"goog-remb", ""},
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{"nack", ""},
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{"nack", ""},
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{"nack", "pli"},
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{"nack", "pli"},
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{"ccm", "fir"},
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{"ccm", "fir"},
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}
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}
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fba = []webrtc.RTCPFeedback{}
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switch codecName {
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switch codecName {
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case "VP8":
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case "VP8":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fbv}, PayloadType: 96}
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case "VP9":
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case "VP9":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fbv}, PayloadType: 98}
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case "H264":
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case "H264":
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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,}
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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: fbv}, PayloadType: 102}
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case "Opus":
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case "Opus":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111}
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case "G722":
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case "G722":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 9,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 9}
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case "PCMU":
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case "PCMU":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 0,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 0}
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case "PCMA":
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case "PCMA":
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codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8,}
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8}
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default:
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default:
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return nil, codec, fmt.Errorf("unknown codec %s", codecName)
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return nil, codec, fmt.Errorf("unknown codec %s", codecName)
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}
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}
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