fix audio del 4 byte on start fix webrtc add options ice and port, webrtc.MediaEngine{}
This commit is contained in:
Andrey Semochkin
parent
f00d1c189a
commit
8d167fd1c0
@ -272,9 +272,6 @@ func (element *Muxer) WritePacket(pkt av.Packet, GOP bool) (bool, []byte, error)
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if stream.lastpkt != nil {
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ts = pkt.Time - stream.lastpkt.Time
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}
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if stream.CodecData.Type().IsAudio() {
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pkt.Data = pkt.Data[4:]
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}
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got, buf, err := stream.writePacketV2(pkt, ts, 5)
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stream.lastpkt = &pkt
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if err != nil {
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@ -522,7 +522,6 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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client.BufferRtpPacket.Truncate(0)
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client.BufferRtpPacket.Reset()
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}
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nalRaw, _ := h264parser.SplitNALUs(content[offset:end])
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var retmap []*av.Packet
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for _, nal := range nalRaw {
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@ -559,6 +558,22 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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if isEnd {
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client.fuStarted = false
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naluTypef := client.BufferRtpPacket.Bytes()[0] & 0x1f
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if naluTypef == 7 {
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bufered, _ := h264parser.SplitNALUs(append([]byte{0, 0, 0, 1}, client.BufferRtpPacket.Bytes()...))
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for _, v := range bufered {
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naluTypefs := v[0] & 0x1f
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switch {
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case naluTypefs == 5:
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client.BufferRtpPacket.Reset()
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client.BufferRtpPacket.Write(v)
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naluTypef = 5
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case naluTypefs == 7:
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client.CodecUpdateSPS(v)
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case naluTypefs == 8:
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client.CodecUpdatePPS(v)
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}
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}
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}
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(client.BufferRtpPacket.Len()), client.BufferRtpPacket.Bytes()...),
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CompositionTime: time.Duration(1) * time.Millisecond,
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@ -591,7 +606,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
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client.AudioTimeLine += duration
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(len(nal)), nal...),
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Data: nal,
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: duration,
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Idx: client.audioIDX,
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@ -602,7 +617,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
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client.AudioTimeLine += duration
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(len(nal)), nal...),
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Data: nal,
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: duration,
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Idx: client.audioIDX,
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@ -613,7 +628,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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duration = time.Duration(20) * time.Millisecond
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client.AudioTimeLine += duration
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(len(nal)), nal...),
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Data: nal,
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: duration,
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Idx: client.audioIDX,
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@ -638,7 +653,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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duration = time.Duration((float32(1024)/float32(client.AudioTimeScale))*1000) * time.Millisecond
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client.AudioTimeLine += duration
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(len(frame)), frame...),
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Data: frame,
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: duration,
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Idx: client.audioIDX,
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@ -7,6 +7,7 @@ import (
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"log"
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"time"
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"github.com/pion/interceptor"
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"github.com/pion/webrtc/v3"
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"github.com/deepch/vdk/av"
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@ -29,19 +30,44 @@ type Muxer struct {
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pc *webrtc.PeerConnection
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ClientACK *time.Timer
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StreamACK *time.Timer
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Options Options
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}
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type Stream struct {
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codec av.CodecData
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ts time.Duration
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track *webrtc.TrackLocalStaticSample
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}
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type Options struct {
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ICEServers []string
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PortMin uint16
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PortMax uint16
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}
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func NewMuxer() *Muxer {
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tmp := Muxer{ClientACK: time.NewTimer(time.Second * 20), StreamACK: time.NewTimer(time.Second * 20), streams: make(map[int8]*Stream)}
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func NewMuxer(options Options) *Muxer {
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tmp := Muxer{Options: options, ClientACK: time.NewTimer(time.Second * 20), StreamACK: time.NewTimer(time.Second * 20), streams: make(map[int8]*Stream)}
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go tmp.WaitCloser()
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return &tmp
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}
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func (element *Muxer) NewPeerConnection(configuration webrtc.Configuration) (*webrtc.PeerConnection, error) {
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if len(element.Options.ICEServers) > 0 {
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log.Println("Set ICEServers", element.Options.ICEServers)
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configuration.ICEServers = append(configuration.ICEServers, webrtc.ICEServer{URLs: element.Options.ICEServers})
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}
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m := &webrtc.MediaEngine{}
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if err := m.RegisterDefaultCodecs(); err != nil {
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return nil, err
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}
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i := &interceptor.Registry{}
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if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
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return nil, err
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}
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s := webrtc.SettingEngine{}
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if element.Options.PortMin > 0 && element.Options.PortMax > 0 && element.Options.PortMax > element.Options.PortMin {
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s.SetEphemeralUDPPortRange(element.Options.PortMin, element.Options.PortMax)
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log.Println("Set UDP ports to", element.Options.PortMin, "..", element.Options.PortMax)
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}
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api := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i), webrtc.WithSettingEngine(s))
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return api.NewPeerConnection(configuration)
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}
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func (element *Muxer) WriteHeader(streams []av.CodecData, sdp64 string) (string, error) {
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var WriteHeaderSuccess bool
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if len(streams) == 0 {
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@ -55,7 +81,7 @@ func (element *Muxer) WriteHeader(streams []av.CodecData, sdp64 string) (string,
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Type: webrtc.SDPTypeOffer,
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SDP: string(sdpB),
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}
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peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{})
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peerConnection, err := element.NewPeerConnection(webrtc.Configuration{})
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if err != nil {
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return "", err
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}
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@ -169,9 +195,6 @@ func (element *Muxer) WritePacket(pkt av.Packet) (err error) {
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}
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if tmp, ok := element.streams[pkt.Idx]; ok {
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element.StreamACK.Reset(10 * time.Second)
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if tmp.ts == 0 {
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tmp.ts = pkt.Time
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}
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switch tmp.codec.Type() {
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case av.H264:
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codec := tmp.codec.(h264parser.CodecData)
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@ -180,18 +203,20 @@ func (element *Muxer) WritePacket(pkt av.Packet) (err error) {
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} else {
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pkt.Data = pkt.Data[4:]
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}
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case av.PCM_MULAW:
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pkt.Data = pkt.Data[4:]
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case av.PCM_ALAW:
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pkt.Data = pkt.Data[4:]
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case av.OPUS:
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pkt.Data = pkt.Data[4:]
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case av.PCM_ALAW:
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case av.AAC:
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//TODO: NEED ADD DECODER AND ENCODER
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return ErrorCodecNotSupported
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case av.PCM:
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//TODO: NEED ADD ENCODER
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return ErrorCodecNotSupported
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default:
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return ErrorCodecNotSupported
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}
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err = tmp.track.WriteSample(media.Sample{Data: pkt.Data, Duration: pkt.Time - element.streams[pkt.Idx].ts})
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err = tmp.track.WriteSample(media.Sample{Data: pkt.Data, Duration: pkt.Duration})
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if err == nil {
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element.streams[pkt.Idx].ts = pkt.Time
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WritePacketSuccess = true
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}
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return err
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