golanggohlsrtmpwebrtcmedia-serverobs-studiortcprtmp-proxyrtmp-serverrtprtsprtsp-proxyrtsp-relayrtsp-serversrtstreamingwebrtc-proxy
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363 lines
6.8 KiB
363 lines
6.8 KiB
package sourcertmp |
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import ( |
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"fmt" |
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"net" |
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"sync" |
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"sync/atomic" |
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"time" |
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"github.com/aler9/gortsplib" |
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"github.com/aler9/gortsplib/pkg/rtcpsender" |
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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/notedit/rtmp/codec/h264" |
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"github.com/notedit/rtmp/format/rtmp" |
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"github.com/aler9/rtsp-simple-server/internal/logger" |
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"github.com/aler9/rtsp-simple-server/internal/stats" |
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) |
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const ( |
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retryPause = 5 * time.Second |
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analyzeTimeout = 8 * time.Second |
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) |
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// Parent is implemeneted by path.Path. |
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type Parent interface { |
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Log(logger.Level, string, ...interface{}) |
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OnSourceSetReady(gortsplib.Tracks) |
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OnSourceSetNotReady() |
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OnFrame(int, gortsplib.StreamType, []byte) |
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} |
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// Source is a RTMP source. |
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type Source struct { |
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ur string |
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wg *sync.WaitGroup |
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stats *stats.Stats |
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parent Parent |
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// in |
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terminate chan struct{} |
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} |
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// New allocates a Source. |
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func New(ur string, |
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wg *sync.WaitGroup, |
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stats *stats.Stats, |
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parent Parent) *Source { |
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s := &Source{ |
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ur: ur, |
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wg: wg, |
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stats: stats, |
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parent: parent, |
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terminate: make(chan struct{}), |
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} |
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atomic.AddInt64(s.stats.CountSourcesRtmp, +1) |
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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 *Source) Close() { |
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atomic.AddInt64(s.stats.CountSourcesRtmpRunning, -1) |
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s.log(logger.Info, "stopped") |
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close(s.terminate) |
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} |
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// IsSource implements path.source. |
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func (s *Source) IsSource() {} |
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// IsSourceExternal implements path.sourceExternal. |
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func (s *Source) IsSourceExternal() {} |
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func (s *Source) 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 *Source) run() { |
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defer s.wg.Done() |
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for { |
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ok := func() bool { |
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ok := s.runInner() |
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if !ok { |
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return false |
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} |
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t := time.NewTimer(retryPause) |
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defer t.Stop() |
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select { |
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case <-t.C: |
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return true |
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case <-s.terminate: |
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return false |
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} |
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}() |
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if !ok { |
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break |
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} |
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} |
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} |
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func (s *Source) runInner() bool { |
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s.log(logger.Info, "connecting") |
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var conn *rtmp.Conn |
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var nconn net.Conn |
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var err error |
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dialDone := make(chan struct{}, 1) |
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go func() { |
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defer close(dialDone) |
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conn, nconn, err = rtmp.NewClient().Dial(s.ur, rtmp.PrepareReading) |
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}() |
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select { |
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case <-s.terminate: |
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return false |
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case <-dialDone: |
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} |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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// gather video and audio features |
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var h264Sps []byte |
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var h264Pps []byte |
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var aacConfig []byte |
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confDone := make(chan struct{}) |
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confClose := uint32(0) |
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go func() { |
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defer close(confDone) |
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for { |
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var pkt av.Packet |
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pkt, err = conn.ReadPacket() |
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if err != nil { |
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return |
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} |
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if atomic.LoadUint32(&confClose) > 0 { |
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return |
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} |
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switch pkt.Type { |
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case av.H264DecoderConfig: |
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codec, err := h264.FromDecoderConfig(pkt.Data) |
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if err != nil { |
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panic(err) |
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} |
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h264Sps, h264Pps = codec.SPS[0], codec.PPS[0] |
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if aacConfig != nil { |
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return |
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} |
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case av.AACDecoderConfig: |
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aacConfig = pkt.Data |
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if h264Sps != nil { |
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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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timer := time.NewTimer(analyzeTimeout) |
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defer timer.Stop() |
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select { |
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case <-confDone: |
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case <-timer.C: |
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atomic.StoreUint32(&confClose, 1) |
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<-confDone |
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} |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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var tracks gortsplib.Tracks |
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var videoTrack *gortsplib.Track |
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var videoRTCPSender *rtcpsender.RTCPSender |
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var h264Encoder *rtph264.Encoder |
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var audioTrack *gortsplib.Track |
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var audioRTCPSender *rtcpsender.RTCPSender |
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var aacEncoder *rtpaac.Encoder |
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if h264Sps != nil { |
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videoTrack, err = gortsplib.NewTrackH264(96, h264Sps, h264Pps) |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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clockRate, _ := videoTrack.ClockRate() |
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videoRTCPSender = rtcpsender.New(clockRate) |
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h264Encoder, err = rtph264.NewEncoder(96) |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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tracks = append(tracks, videoTrack) |
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} |
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if aacConfig != nil { |
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audioTrack, err = gortsplib.NewTrackAAC(96, aacConfig) |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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clockRate, _ := audioTrack.ClockRate() |
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audioRTCPSender = rtcpsender.New(clockRate) |
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aacEncoder, err = rtpaac.NewEncoder(96, clockRate) |
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if err != nil { |
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s.log(logger.Info, "ERR: %s", err) |
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return true |
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} |
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tracks = append(tracks, audioTrack) |
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} |
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if len(tracks) == 0 { |
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s.log(logger.Info, "ERR: no tracks found") |
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return true |
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} |
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for i, t := range tracks { |
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t.ID = i |
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} |
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s.log(logger.Info, "ready") |
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s.parent.OnSourceSetReady(tracks) |
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defer s.parent.OnSourceSetNotReady() |
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rtcpTerminate := make(chan struct{}) |
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rtcpDone := make(chan struct{}) |
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go func() { |
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close(rtcpDone) |
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t := time.NewTicker(10 * time.Second) |
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defer t.Stop() |
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for { |
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select { |
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case <-t.C: |
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now := time.Now() |
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if videoRTCPSender != nil { |
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r := videoRTCPSender.Report(now) |
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if r != nil { |
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s.parent.OnFrame(videoTrack.ID, gortsplib.StreamTypeRTCP, r) |
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} |
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} |
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if audioRTCPSender != nil { |
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r := audioRTCPSender.Report(now) |
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if r != nil { |
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s.parent.OnFrame(audioTrack.ID, gortsplib.StreamTypeRTCP, r) |
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} |
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} |
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case <-rtcpTerminate: |
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return |
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} |
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} |
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}() |
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readerDone := make(chan error) |
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go func() { |
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for { |
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pkt, err := conn.ReadPacket() |
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if err != nil { |
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readerDone <- err |
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return |
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} |
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switch pkt.Type { |
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case av.H264: |
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if h264Sps == nil { |
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readerDone <- fmt.Errorf("rtmp source ERR: received an H264 frame, but track is not setup up") |
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return |
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} |
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// decode from AVCC format |
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nalus, typ := h264.SplitNALUs(pkt.Data) |
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if typ != h264.NALU_AVCC { |
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readerDone <- fmt.Errorf("invalid NALU format (%d)", typ) |
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return |
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} |
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// encode into RTP/H264 format |
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frames, err := h264Encoder.Write(pkt.Time+pkt.CTime, nalus) |
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if err != nil { |
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readerDone <- err |
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return |
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} |
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for _, f := range frames { |
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videoRTCPSender.ProcessFrame(time.Now(), gortsplib.StreamTypeRTP, f) |
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s.parent.OnFrame(videoTrack.ID, gortsplib.StreamTypeRTP, f) |
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} |
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case av.AAC: |
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if aacConfig == nil { |
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readerDone <- fmt.Errorf("rtmp source ERR: received an AAC frame, but track is not setup up") |
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return |
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} |
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frames, err := aacEncoder.Write(pkt.Time+pkt.CTime, pkt.Data) |
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if err != nil { |
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readerDone <- err |
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return |
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} |
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for _, f := range frames { |
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audioRTCPSender.ProcessFrame(time.Now(), gortsplib.StreamTypeRTP, f) |
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s.parent.OnFrame(audioTrack.ID, gortsplib.StreamTypeRTP, f) |
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} |
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default: |
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readerDone <- fmt.Errorf("rtmp source ERR: unexpected packet: %v", pkt.Type) |
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return |
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} |
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} |
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}() |
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for { |
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select { |
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case <-s.terminate: |
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nconn.Close() |
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<-readerDone |
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close(rtcpTerminate) |
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<-rtcpDone |
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return false |
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case err := <-readerDone: |
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nconn.Close() |
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s.log(logger.Info, "ERR: %s", err) |
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close(rtcpTerminate) |
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<-rtcpDone |
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return true |
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} |
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} |
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}
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