golanggohlsrtmpwebrtcmedia-serverobs-studiortcprtmp-proxyrtmp-serverrtprtsprtsp-proxyrtsp-relayrtsp-serversrtstreamingwebrtc-proxy
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255 lines
5.6 KiB
255 lines
5.6 KiB
package core |
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import ( |
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"context" |
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"fmt" |
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"sync" |
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"time" |
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"github.com/aler9/gortsplib" |
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"github.com/aler9/gortsplib/pkg/h264" |
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"github.com/aler9/gortsplib/pkg/rtpaac" |
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"github.com/aler9/gortsplib/pkg/rtph264" |
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"github.com/notedit/rtmp/av" |
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"github.com/aler9/rtsp-simple-server/internal/conf" |
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"github.com/aler9/rtsp-simple-server/internal/logger" |
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"github.com/aler9/rtsp-simple-server/internal/rtmp" |
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) |
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const ( |
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rtmpSourceRetryPause = 5 * time.Second |
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) |
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type rtmpSourceParent interface { |
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log(logger.Level, string, ...interface{}) |
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onSourceStaticSetReady(req pathSourceStaticSetReadyReq) pathSourceStaticSetReadyRes |
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onSourceStaticSetNotReady(req pathSourceStaticSetNotReadyReq) |
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} |
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type rtmpSource struct { |
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ur string |
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readTimeout conf.StringDuration |
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writeTimeout conf.StringDuration |
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wg *sync.WaitGroup |
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parent rtmpSourceParent |
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ctx context.Context |
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ctxCancel func() |
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} |
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func newRTMPSource( |
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parentCtx context.Context, |
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ur string, |
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readTimeout conf.StringDuration, |
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writeTimeout conf.StringDuration, |
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wg *sync.WaitGroup, |
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parent rtmpSourceParent) *rtmpSource { |
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ctx, ctxCancel := context.WithCancel(parentCtx) |
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s := &rtmpSource{ |
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ur: ur, |
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readTimeout: readTimeout, |
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writeTimeout: writeTimeout, |
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wg: wg, |
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parent: parent, |
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ctx: ctx, |
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ctxCancel: ctxCancel, |
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} |
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s.log(logger.Info, "started") |
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s.wg.Add(1) |
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go s.run() |
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return s |
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} |
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// Close closes a Source. |
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func (s *rtmpSource) close() { |
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s.log(logger.Info, "stopped") |
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s.ctxCancel() |
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} |
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func (s *rtmpSource) log(level logger.Level, format string, args ...interface{}) { |
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s.parent.log(level, "[rtmp source] "+format, args...) |
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} |
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func (s *rtmpSource) run() { |
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defer s.wg.Done() |
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outer: |
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for { |
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ok := s.runInner() |
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if !ok { |
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break outer |
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} |
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select { |
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case <-time.After(rtmpSourceRetryPause): |
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case <-s.ctx.Done(): |
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break outer |
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} |
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} |
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s.ctxCancel() |
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} |
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func (s *rtmpSource) runInner() bool { |
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innerCtx, innerCtxCancel := context.WithCancel(s.ctx) |
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runErr := make(chan error) |
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go func() { |
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runErr <- func() error { |
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s.log(logger.Debug, "connecting") |
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ctx2, cancel2 := context.WithTimeout(innerCtx, time.Duration(s.readTimeout)) |
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defer cancel2() |
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conn, err := rtmp.DialContext(ctx2, s.ur) |
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if err != nil { |
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return err |
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} |
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readDone := make(chan error) |
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go func() { |
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readDone <- func() error { |
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conn.SetReadDeadline(time.Now().Add(time.Duration(s.readTimeout))) |
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conn.SetWriteDeadline(time.Now().Add(time.Duration(s.writeTimeout))) |
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err = conn.ClientHandshake() |
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if err != nil { |
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return err |
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} |
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conn.SetWriteDeadline(time.Time{}) |
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conn.SetReadDeadline(time.Now().Add(time.Duration(s.readTimeout))) |
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videoTrack, audioTrack, err := conn.ReadTracks() |
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if err != nil { |
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return err |
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} |
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var tracks gortsplib.Tracks |
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videoTrackID := -1 |
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audioTrackID := -1 |
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var h264Encoder *rtph264.Encoder |
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if videoTrack != nil { |
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h264Encoder = &rtph264.Encoder{PayloadType: 96} |
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h264Encoder.Init() |
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videoTrackID = len(tracks) |
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tracks = append(tracks, videoTrack) |
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} |
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var aacEncoder *rtpaac.Encoder |
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if audioTrack != nil { |
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aacEncoder = &rtpaac.Encoder{ |
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PayloadType: 97, |
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SampleRate: audioTrack.ClockRate(), |
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} |
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aacEncoder.Init() |
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audioTrackID = len(tracks) |
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tracks = append(tracks, audioTrack) |
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} |
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res := s.parent.onSourceStaticSetReady(pathSourceStaticSetReadyReq{ |
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source: s, |
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tracks: tracks, |
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}) |
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if res.err != nil { |
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return res.err |
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} |
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s.log(logger.Info, "ready") |
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defer func() { |
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s.parent.onSourceStaticSetNotReady(pathSourceStaticSetNotReadyReq{source: s}) |
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}() |
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for { |
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conn.SetReadDeadline(time.Now().Add(time.Duration(s.readTimeout))) |
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pkt, err := conn.ReadPacket() |
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if err != nil { |
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return err |
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} |
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switch pkt.Type { |
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case av.H264: |
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if videoTrack == nil { |
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return fmt.Errorf("received an H264 packet, but track is not set up") |
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} |
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nalus, err := h264.DecodeAVCC(pkt.Data) |
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if err != nil { |
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return err |
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} |
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var outNALUs [][]byte |
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for _, nalu := range nalus { |
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// remove SPS, PPS and AUD, not needed by RTSP / RTMP |
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typ := h264.NALUType(nalu[0] & 0x1F) |
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switch typ { |
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case h264.NALUTypeSPS, h264.NALUTypePPS, h264.NALUTypeAccessUnitDelimiter: |
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continue |
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} |
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outNALUs = append(outNALUs, nalu) |
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} |
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pkts, err := h264Encoder.Encode(outNALUs, pkt.Time+pkt.CTime) |
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if err != nil { |
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return fmt.Errorf("error while encoding H264: %v", err) |
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} |
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for _, pkt := range pkts { |
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res.stream.writePacketRTP(videoTrackID, pkt) |
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} |
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case av.AAC: |
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if audioTrack == nil { |
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return fmt.Errorf("received an AAC packet, but track is not set up") |
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} |
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pkts, err := aacEncoder.Encode([][]byte{pkt.Data}, pkt.Time+pkt.CTime) |
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if err != nil { |
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return fmt.Errorf("error while encoding AAC: %v", err) |
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} |
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for _, pkt := range pkts { |
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res.stream.writePacketRTP(audioTrackID, pkt) |
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} |
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} |
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} |
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}() |
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}() |
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select { |
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case err := <-readDone: |
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conn.Close() |
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return err |
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case <-innerCtx.Done(): |
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conn.Close() |
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<-readDone |
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return nil |
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} |
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}() |
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}() |
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select { |
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case err := <-runErr: |
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innerCtxCancel() |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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case <-s.ctx.Done(): |
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innerCtxCancel() |
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<-runErr |
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return false |
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} |
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} |
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// onSourceAPIDescribe implements source. |
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func (*rtmpSource) onSourceAPIDescribe() interface{} { |
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return struct { |
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Type string `json:"type"` |
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}{"rtmpSource"} |
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}
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