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重构输出流转发, TransStream不再持有Sink
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yangjiechina
@@ -17,136 +17,87 @@ const (
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OverTcpMagic = 0x24
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)
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// rtsp传输流封装
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// TranStream rtsp传输流封装
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// 低延迟是rtsp特性, 所以不考虑实现GOP缓存
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type tranStream struct {
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type TranStream struct {
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stream.BaseTransStream
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addr net.IPAddr
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addrType string
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urlFormat string
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rtpTracks []*rtspTrack
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rtpTracks []*Track
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sdp string
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buffer *stream.ReceiveBuffer
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}
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// rtpMuxer申请输出流内存的回调
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// 无论是tcp/udp拉流, 均使用同一块内存, 并且给tcp预留4字节的包长.
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func (t *tranStream) onAllocBuffer(params interface{}) []byte {
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return t.rtpTracks[params.(int)].buffer[OverTcpHeaderSize:]
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}
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// onRtpPacket 所有封装后的RTP流都将回调于此
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func (t *tranStream) onRtpPacket(data []byte, timestamp uint32, params interface{}) {
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//params传递track索引
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index := params.(int)
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track := t.rtpTracks[index]
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//保存带有sps和ssp等编码信息的rtp包, 对所有sink通用
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if track.cache && track.extraDataBuffer == nil {
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bytes := make([]byte, OverTcpHeaderSize+len(data))
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copy(bytes[OverTcpHeaderSize:], data)
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track.tmpExtraDataBuffer = append(track.tmpExtraDataBuffer, bytes)
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t.overTCP(bytes, index)
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return
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}
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//将rtp包发送给各个sink
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for _, value := range t.Sinks {
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sink_ := value.(*sink)
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if !sink_.isConnected(index) {
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continue
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}
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//为刚刚连接上的sink, 发送sps和pps等rtp包
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if sink_.pktCount(index) < 1 && utils.AVMediaTypeVideo == track.mediaType {
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seq := binary.BigEndian.Uint16(data[2:])
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count := len(track.extraDataBuffer)
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for i, rtp := range track.extraDataBuffer {
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//回滚rtp包的序号
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librtp.RollbackSeq(rtp[OverTcpHeaderSize:], int(seq)-(count-i-1))
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if sink_.tcp {
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sink_.input(index, rtp, 0)
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} else {
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sink_.input(index, rtp[OverTcpHeaderSize:], timestamp)
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}
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}
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}
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end := OverTcpHeaderSize + len(data)
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t.overTCP(track.buffer[:end], index)
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//发送rtp包
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if sink_.tcp {
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sink_.input(index, track.buffer[:end], 0)
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} else {
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sink_.input(index, data, timestamp)
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}
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}
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}
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func (t *tranStream) overTCP(data []byte, channel int) {
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func (t *TranStream) OverTCP(data []byte, channel int) {
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data[0] = OverTcpMagic
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data[1] = byte(channel)
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binary.BigEndian.PutUint16(data[2:], uint16(len(data)-4))
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}
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func (t *tranStream) Input(packet utils.AVPacket) error {
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stream_ := t.rtpTracks[packet.Index()]
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func (t *TranStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
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t.ClearOutStreamBuffer()
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var ts uint32
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track := t.rtpTracks[packet.Index()]
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track.seq = track.muxer.GetHeader().Seq
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if utils.AVMediaTypeAudio == packet.MediaType() {
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stream_.muxer.Input(packet.Data(), uint32(packet.ConvertPts(stream_.rate)))
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ts = uint32(packet.ConvertPts(track.rate))
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t.PackRtpPayload(track.muxer, packet.Index(), packet.Data(), ts)
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} else if utils.AVMediaTypeVideo == packet.MediaType() {
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ts = uint32(packet.ConvertPts(track.rate))
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data := libavc.RemoveStartCode(packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()]))
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t.PackRtpPayload(track.muxer, packet.Index(), data, ts)
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}
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//将sps和pps按照单一模式打包
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if stream_.extraDataBuffer == nil {
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if !packet.KeyFrame() {
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return nil
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}
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return t.OutBuffer[:t.OutBufferSize], int64(ts), utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame(), nil
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}
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stream_.cache = true
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parameters := t.BaseTransStream.Tracks[packet.Index()].CodecParameters()
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if utils.AVCodecIdH265 == packet.CodecId() {
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bytes := parameters.(*utils.HEVCCodecData).VPS()
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stream_.muxer.Input(bytes[0], uint32(packet.ConvertPts(stream_.rate)))
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}
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spsBytes := parameters.SPS()
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ppsBytes := parameters.PPS()
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stream_.muxer.Input(spsBytes[0], uint32(packet.ConvertPts(stream_.rate)))
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stream_.muxer.Input(ppsBytes[0], uint32(packet.ConvertPts(stream_.rate)))
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stream_.extraDataBuffer = stream_.tmpExtraDataBuffer
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func (t *TranStream) ReadExtraData(ts int64) ([][]byte, int64, error) {
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// 返回视频编码数据的rtp包
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for _, track := range t.rtpTracks {
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if utils.AVMediaTypeVideo != track.mediaType {
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continue
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}
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data := libavc.RemoveStartCode(packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()]))
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stream_.muxer.Input(data, uint32(packet.ConvertPts(stream_.rate)))
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// 回滚序号和时间戳
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index := int(track.seq) - len(track.extraDataBuffer)
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for i, bytes := range track.extraDataBuffer {
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librtp.RollbackSeq(bytes[OverTcpHeaderSize:], index+i+1)
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binary.BigEndian.PutUint32(bytes[OverTcpHeaderSize+4:], uint32(ts))
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}
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return track.extraDataBuffer, ts, nil
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}
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return nil
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return nil, ts, nil
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}
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func (t *tranStream) AddSink(sink_ stream.Sink) error {
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sink_.(*sink).setSenderCount(len(t.BaseTransStream.Tracks))
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if err := sink_.(*sink).SendHeader([]byte(t.sdp)); err != nil {
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return err
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}
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return t.BaseTransStream.AddSink(sink_)
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func (t *TranStream) PackRtpPayload(muxer librtp.Muxer, channel int, data []byte, timestamp uint32) {
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var index int
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muxer.Input(data, timestamp, func() []byte {
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index = t.buffer.Index()
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block := t.buffer.GetBlock()
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return block[OverTcpHeaderSize:]
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}, func(bytes []byte) {
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packet := t.buffer.Get(index)[:OverTcpHeaderSize+len(bytes)]
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t.OverTCP(packet, channel)
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t.AppendOutStreamBuffer(packet)
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})
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}
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func (t *tranStream) AddTrack(stream utils.AVStream) error {
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func (t *TranStream) AddTrack(stream utils.AVStream) error {
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if err := t.BaseTransStream.AddTrack(stream); err != nil {
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return err
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}
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payloadType, ok := librtp.CodecIdPayloads[stream.CodecId()]
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if !ok {
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return fmt.Errorf("no payload type was found for codecid:%d", stream.CodecId())
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return fmt.Errorf("no payload type was found for codecid: %d", stream.CodecId())
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}
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//创建RTP封装
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// 创建RTP封装器
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var muxer librtp.Muxer
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if utils.AVCodecIdH264 == stream.CodecId() {
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muxer = librtp.NewH264Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
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@@ -158,25 +109,51 @@ func (t *tranStream) AddTrack(stream utils.AVStream) error {
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muxer = librtp.NewMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
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}
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muxer.SetAllocHandler(t.onAllocBuffer)
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muxer.SetWriteHandler(t.onRtpPacket)
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t.rtpTracks = append(t.rtpTracks, NewRTSPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, stream.Type()))
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muxer.SetParams(len(t.rtpTracks) - 1)
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index := len(t.rtpTracks) - 1
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// 将sps和pps按照单一模式打包
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bufferIndex := t.buffer.Index()
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if utils.AVMediaTypeVideo == stream.Type() {
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parameters := stream.CodecParameters()
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if utils.AVCodecIdH265 == stream.CodecId() {
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bytes := parameters.(*utils.HEVCCodecData).VPS()
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t.PackRtpPayload(muxer, index, bytes[0], 0)
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}
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spsBytes := parameters.SPS()
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ppsBytes := parameters.PPS()
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t.PackRtpPayload(muxer, index, spsBytes[0], 0)
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t.PackRtpPayload(muxer, index, ppsBytes[0], 0)
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// 拷贝扩展数据的rtp包
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size := t.buffer.Index() - bufferIndex
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extraRtpBuffer := make([][]byte, size)
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for i := 0; i < size; i++ {
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src := t.buffer.Get(bufferIndex + i)
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dst := make([]byte, len(src))
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copy(dst, src)
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extraRtpBuffer[i] = dst[:OverTcpHeaderSize+binary.BigEndian.Uint16(dst[2:])]
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}
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t.rtpTracks[index].extraDataBuffer = extraRtpBuffer
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}
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return nil
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}
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func (t *tranStream) Close() error {
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func (t *TranStream) Close() ([][]byte, int64, error) {
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for _, track := range t.rtpTracks {
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if track != nil {
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track.Close()
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}
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}
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return nil
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return nil, 0, nil
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}
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func (t *tranStream) WriteHeader() error {
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func (t *TranStream) WriteHeader() error {
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description := sdp.SessionDescription{
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Version: 0,
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Origin: sdp.Origin{
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@@ -261,9 +238,11 @@ func (t *tranStream) WriteHeader() error {
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}
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func NewTransStream(addr net.IPAddr, urlFormat string) stream.TransStream {
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t := &tranStream{
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t := &TranStream{
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addr: addr,
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urlFormat: urlFormat,
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// 在将AVPacket打包rtp时, 会使用多个buffer块, 回环覆盖多个rtp块, 如果是TCP拉流并且网络不好, 推流的数据会错乱.
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buffer: stream.NewReceiveBuffer(1500, 1024),
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}
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if addr.IP.To4() != nil {
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