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323ac7204e
2、Track增加离线状态，当发布者离线时，Track状态改为离线状态，当发布者重新上线时，Track状态改为在线状态。 3、Track在恢复在线后，记录时间戳的差值，保持后续时间戳和之前的连续。 4、进一步优化订阅者读取音视频同步逻辑。
121 lines
3.2 KiB
Go
121 lines
3.2 KiB
Go
package track
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import (
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"github.com/pion/rtp"
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"go.uber.org/zap"
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. "m7s.live/engine/v4/common"
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"m7s.live/engine/v4/config"
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"m7s.live/engine/v4/util"
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)
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const RTPMTU = 1400
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func (av *Media) UnmarshalRTPPacket(p *rtp.Packet) (frame *RTPFrame) {
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if av.PayloadType != p.PayloadType {
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av.Warn("RTP PayloadType error", zap.Uint8("want", av.PayloadType), zap.Uint8("got", p.PayloadType))
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return
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}
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frame = &RTPFrame{
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Packet: *p,
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}
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av.Value.BytesIn += len(p.Payload) + 12
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return av.recorderRTP(frame)
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}
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func (av *Media) UnmarshalRTP(raw []byte) (frame *RTPFrame) {
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var p rtp.Packet
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err := p.Unmarshal(raw)
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if err != nil {
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av.Warn("RTP Unmarshal error", zap.Error(err))
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return
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}
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return av.UnmarshalRTPPacket(&p)
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}
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func (av *Media) writeRTPFrame(frame *RTPFrame) {
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av.Value.RTP.PushValue(*frame)
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av.WriteRTPFrame(frame)
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if frame.Marker {
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av.SpesificTrack.generateTimestamp(frame.Timestamp)
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av.SpesificTrack.Flush()
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}
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}
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// WriteRTPPack 写入已反序列化的RTP包
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func (av *Media) WriteRTPPack(p *rtp.Packet) {
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for frame := av.UnmarshalRTPPacket(p); frame != nil; frame = av.nextRTPFrame() {
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av.writeRTPFrame(frame)
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}
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}
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// WriteRTP 写入未反序列化的RTP包
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func (av *Media) WriteRTP(raw []byte) {
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for frame := av.UnmarshalRTP(raw); frame != nil; frame = av.nextRTPFrame() {
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av.writeRTPFrame(frame)
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}
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}
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// https://www.cnblogs.com/moonwalk/p/15903760.html
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// Packetize packetizes the payload of an RTP packet and returns one or more RTP packets
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func (av *Media) PacketizeRTP(payloads ...[][]byte) {
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packetCount := len(payloads)
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for i, pp := range payloads {
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av.rtpSequence++
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rtpItem := av.rtpPool.Get()
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packet := &rtpItem.Value
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if packet.Payload == nil {
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packet.Payload = make([]byte, 0, RTPMTU)
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packet.Version = 2
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packet.PayloadType = av.PayloadType
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packet.SSRC = av.SSRC
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}
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packet.Payload = packet.Payload[:0]
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packet.SequenceNumber = av.rtpSequence
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packet.Timestamp = av.Value.PTS
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packet.Marker = i == packetCount-1
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for _, p := range pp {
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packet.Payload = append(packet.Payload, p...)
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}
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av.Value.RTP.Push(rtpItem)
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}
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}
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type RTPDemuxer struct {
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lastSeq uint16 //上一个收到的序号，用于乱序重排
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lastSeq2 uint16 //记录上上一个收到的序列号
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乱序重排 util.RTPReorder[*RTPFrame]
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}
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// 获取缓存中下一个rtpFrame
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func (av *RTPDemuxer) nextRTPFrame() (frame *RTPFrame) {
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if config.Global.RTPReorder {
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return av.乱序重排.Pop()
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}
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return
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}
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// 对RTP包乱序重排
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func (av *RTPDemuxer) recorderRTP(frame *RTPFrame) *RTPFrame {
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if config.Global.RTPReorder {
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return av.乱序重排.Push(frame.SequenceNumber, frame)
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} else {
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if av.lastSeq == 0 {
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av.lastSeq = frame.SequenceNumber
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} else if frame.SequenceNumber == av.lastSeq2+1 { // 本次序号是上上次的序号+1 说明中间隔了一个错误序号（某些rtsp流中的rtcp包写成了rtp包导致的）
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av.lastSeq = frame.SequenceNumber
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} else {
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av.lastSeq2 = av.lastSeq
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av.lastSeq = frame.SequenceNumber
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if av.lastSeq != av.lastSeq2+1 { //序号不连续
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// av.Stream.Warn("RTP SequenceNumber error", av.lastSeq2, av.lastSeq)
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return frame
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}
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}
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return frame
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}
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}
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type RTPMuxer struct {
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rtpSequence uint16 //用于生成下一个rtp包的序号
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}
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