1、引入ClockRate, 用于RTP时间戳处理（MPEG2中的时间戳是90KHZ）

2、修复控制流Sleep传入的时间单位错误 3、优化TS格式解析内存复用，减少内存分配
2025-10-18 22:44:42 +08:00 · 2023-02-08 14:00:07 +08:00
parent 323ac7204e
commit eafd4f6aa2
11 changed files with 124 additions and 107 deletions
--- a/codec/mpegts/mpegts.go
+++ b/codec/mpegts/mpegts.go
@@ -102,10 +102,9 @@ const (
 //

 type MpegTsStream struct {
-	PAT         MpegTsPAT // PAT表信息
-	PMT         MpegTsPMT // PMT表信息
-	tsPktBuffer [][]byte  // TS包缓存
-	PESChan     chan *MpegTsPESPacket
+	PAT     MpegTsPAT // PAT表信息
+	PMT     MpegTsPMT // PMT表信息
+	PESChan chan *MpegTsPESPacket
 }

 // ios13818-1-CN.pdf 33/165
@@ -518,64 +517,60 @@ func (s *MpegTsStream) ReadPMT(packet *MpegTsPacket, pr io.Reader) (err error) {
 	}
 	return
 }
-func (s *MpegTsStream) Feed(ts io.Reader) error {
-	var frame int64
-	var tsPktArr []MpegTsPacket
-	var reader, pr bytes.Reader
-	defer func() {
-		s.tsPktBuffer = s.tsPktBuffer[:0]
-	}()
+func (s *MpegTsStream) Feed(ts io.Reader) (err error) {
+	var reader bytes.Reader
+	var lr io.LimitedReader
+	lr.R = &reader
+	var pesPkt *MpegTsPESPacket
+	var tsHeader MpegTsHeader
+	tsData := make([]byte, TS_PACKET_SIZE)
 	for {
-		var tsData []byte
-		if tsDataP := util.MallocSlice(&s.tsPktBuffer); tsDataP == nil {
-			tsData = make([]byte, TS_PACKET_SIZE)
-			s.tsPktBuffer = append(s.tsPktBuffer, tsData)
-		} else {
-			tsData = *tsDataP
-		}
-		_, err := io.ReadFull(ts, tsData)
-		reader.Reset(tsData)
+		_, err = io.ReadFull(ts, tsData)
 		if err == io.EOF {
 			// 文件结尾 把最后面的数据发出去
-			pesPkt, err := TsToPES(tsPktArr)
-			if err != nil {
-				return err
+			if pesPkt != nil {
+				s.PESChan <- pesPkt
 			}
-			s.PESChan <- &pesPkt
 			return nil
 		} else if err != nil {
-			return err
+			return
 		}
-		packet, err := ReadTsPacket(&reader)
-		if err != nil {
-			return err
+		reader.Reset(tsData)
+		lr.N = TS_PACKET_SIZE
+		if tsHeader, err = ReadTsHeader(&lr); err != nil {
+			return
 		}
-		pr.Reset(packet.Payload)
-		err = s.ReadPAT(&packet, &pr)
-		if err != nil {
-			return err
+		if tsHeader.SyncByte != 0x47 {
+			return errors.New("sync byte error")
 		}
-		err = s.ReadPMT(&packet, &pr)
-		if err != nil {
-			return err
+		if tsHeader.Pid == PID_PAT {
+			if s.PAT, err = ReadPAT(&lr); err != nil {
+				return
+			}
+			continue
 		}
-		// 在读取PMT中已经将所有的音视频PES的索引信息全部保存了起来
-		// 接着读取所有TS包里面的PID,找出PID==elementaryPID的TS包,就是音视频数据
-		for _, v := range s.PMT.Stream {
-			if v.ElementaryPID == packet.Header.Pid {
-				if packet.Header.PayloadUnitStartIndicator == 1 {
-					if frame != 0 {
-						pesPkt, err := TsToPES(tsPktArr)
-						if err != nil {
-							return err
-						}
-						s.PESChan <- &pesPkt
-						s.tsPktBuffer = s.tsPktBuffer[:0]
-						tsPktArr = tsPktArr[:0]
+		if len(s.PMT.Stream) == 0 {
+			for _, v := range s.PAT.Program {
+				if v.ProgramMapPID == tsHeader.Pid {
+					if s.PMT, err = ReadPMT(&lr); err != nil {
+						return
 					}
-					frame++
 				}
-				tsPktArr = append(tsPktArr, packet)
+				continue
+			}
+		}
+		for _, v := range s.PMT.Stream {
+			if v.ElementaryPID == tsHeader.Pid {
+				if tsHeader.PayloadUnitStartIndicator == 1 {
+					if pesPkt != nil {
+						s.PESChan <- pesPkt
+					}
+					pesPkt = &MpegTsPESPacket{}
+					if pesPkt.Header, err = ReadPESHeader(&lr); err != nil {
+						return
+					}
+				}
+				io.Copy(&pesPkt.Payload, &lr)
 			}
 		}
 	}
--- a/codec/mpegts/mpegts_pes.go
+++ b/codec/mpegts/mpegts_pes.go
@@ -25,7 +25,7 @@ type MpegTsPesStream struct {
 // 110x xxxx 为音频流(0xC0)
 type MpegTsPESPacket struct {
 	Header  MpegTsPESHeader
-	Payload []byte
+	Payload util.Buffer
 }

 type MpegTsPESHeader struct {
--- a/common/frame.go
+++ b/common/frame.go
@@ -72,7 +72,6 @@ func (av *AVFrame) WriteAVCC(ts uint32, frame util.BLL) {
 	}
 	// frame.Transfer(&av.AVCC)
 	// frame.ByteLength = 0
-	av.DTS = ts * 90
 }

 func (av *AVFrame) AppendMem(item *util.ListItem[util.Buffer]) {
--- a/publisher-rtpdump.go
+++ b/publisher-rtpdump.go
@@ -2,6 +2,7 @@ package engine

 import (
 	"os"
+	"time"

 	"github.com/pion/webrtc/v3/pkg/media/rtpdump"
 	"go.uber.org/zap"
@@ -47,8 +48,8 @@ func (t *RTPDumpPublisher) OnEvent(event any) {
 		case codec.CodecID_PCMU:
 			t.AudioTrack = track.NewG711(t.Publisher.Stream, false)
 		}
-		t.VideoTrack.SetSpeedLimit(500)
-		t.AudioTrack.SetSpeedLimit(500)
+		t.VideoTrack.SetSpeedLimit(500 * time.Millisecond)
+		t.AudioTrack.SetSpeedLimit(500 * time.Millisecond)
 		for {
 			packet, err := r.Next()
 			if err != nil {
--- a/publisher-ts.go
+++ b/publisher-ts.go
@@ -3,6 +3,7 @@ package engine
 import (
 	"go.uber.org/zap"
 	"m7s.live/engine/v4/codec/mpegts"
+	"m7s.live/engine/v4/common"
 	"m7s.live/engine/v4/track"
 )

@@ -38,15 +39,15 @@ func (t *TSPublisher) OnPmtStream(s mpegts.MpegTsPmtStream) {
 		}
 	case mpegts.STREAM_TYPE_AAC:
 		if t.AudioTrack == nil {
-			t.AudioTrack = track.NewAAC(t.Publisher.Stream)
+			t.AudioTrack = track.NewAAC(t.Publisher.Stream, uint32(90000))
 		}
 	case mpegts.STREAM_TYPE_G711A:
 		if t.AudioTrack == nil {
-			t.AudioTrack = track.NewG711(t.Publisher.Stream, true)
+			t.AudioTrack = track.NewG711(t.Publisher.Stream, true, uint32(90000))
 		}
 	case mpegts.STREAM_TYPE_G711U:
 		if t.AudioTrack == nil {
-			t.AudioTrack = track.NewG711(t.Publisher.Stream, false)
+			t.AudioTrack = track.NewG711(t.Publisher.Stream, false, uint32(90000))
 		}
 	default:
 		t.Warn("unsupport stream type:", zap.Uint8("type", s.StreamType))
@@ -72,11 +73,7 @@ func (t *TSPublisher) ReadPES() {
 						t.adts = append(t.adts, pes.Payload[:7]...)
 						t.AudioTrack.WriteADTS(t.adts)
 					}
-					current := t.AudioTrack.CurrentFrame()
-					current.PTS = uint32(pes.Header.Pts)
-					current.DTS = uint32(pes.Header.Dts)
 					remainLen := len(pes.Payload)
-					current.BytesIn += remainLen
 					for remainLen > 0 {
 						// AACFrameLength(13)
 						// xx xxxxxxxx xxx
@@ -88,12 +85,9 @@ func (t *TSPublisher) ReadPES() {
 						pes.Payload = pes.Payload[frameLen:remainLen]
 						remainLen -= frameLen
 					}
-					t.AudioTrack.Flush()
 				case *track.G711:
 					t.AudioTrack.WriteRaw(uint32(pes.Header.Pts), pes.Payload)
-					t.AudioTrack.Flush()
 				}
-
 			}
 		case mpegts.STREAM_ID_VIDEO:
 			if t.VideoTrack == nil {
@@ -102,7 +96,7 @@ func (t *TSPublisher) ReadPES() {
 				}
 			}
 			if t.VideoTrack != nil {
-				t.VideoTrack.WriteAnnexB(uint32(pes.Header.Pts), uint32(pes.Header.Dts), pes.Payload)
+				t.VideoTrack.WriteAnnexB(uint32(pes.Header.Pts), uint32(pes.Header.Dts), common.AnnexBFrame(pes.Payload))
 			}
 		}
 	}
--- a/subscriber.go
+++ b/subscriber.go
@@ -183,28 +183,37 @@ func (s *Subscriber) PlayBlock(subType byte) {
 	case SUBTYPE_RAW:
 		sendVideoFrame = func(frame *AVFrame) {
 			// println("v", frame.Sequence, frame.AbsTime, s.VideoReader.AbsTime, frame.IFrame)
-			spesic.OnEvent(VideoFrame{frame, s.VideoReader.AbsTime, frame.PTS - s.VideoReader.SkipTs*90, frame.DTS - s.VideoReader.SkipTs*90})
+			spesic.OnEvent(VideoFrame{frame, s.VideoReader.AbsTime, frame.PTS - s.VideoReader.SkipRTPTs, frame.DTS - s.VideoReader.SkipRTPTs})
 		}
 		sendAudioFrame = func(frame *AVFrame) {
 			// println("a", frame.Sequence, frame.AbsTime, s.AudioReader.AbsTime)
-			spesic.OnEvent(AudioFrame{frame, s.AudioReader.AbsTime, s.AudioReader.AbsTime * 90, s.AudioReader.AbsTime * 90})
+			spesic.OnEvent(AudioFrame{frame, s.AudioReader.AbsTime, s.AudioReader.Track.Ms2RTPTs(s.AudioReader.AbsTime), s.AudioReader.Track.Ms2RTPTs(s.AudioReader.AbsTime)})
 		}
 	case SUBTYPE_RTP:
 		var videoSeq, audioSeq uint16
 		sendVideoFrame = func(frame *AVFrame) {
 			frame.RTP.Range(func(vp RTPFrame) bool {
 				videoSeq++
-				vp.Header.Timestamp = vp.Header.Timestamp - s.VideoReader.SkipTs*90
+				vp.Header.Timestamp = vp.Header.Timestamp - s.VideoReader.SkipRTPTs
 				vp.Header.SequenceNumber = videoSeq
 				spesic.OnEvent((VideoRTP)(vp))
 				return true
 			})
 		}
+		createTimestamp := func(ts uint32) uint32 {
+			return ts - s.AudioReader.SkipRTPTs
+		}
+		// RTP需要转换时间戳调整为采样率
+		if s.Audio.SampleRate != s.Audio.ClockRate {
+			createTimestamp = func(ts uint32) uint32 {
+				return uint32(uint64(ts-s.AudioReader.SkipRTPTs) * uint64(s.Audio.SampleRate) / uint64(s.Audio.ClockRate))
+			}
+		}
 		sendAudioFrame = func(frame *AVFrame) {
 			frame.RTP.Range(func(ap RTPFrame) bool {
 				audioSeq++
 				ap.Header.SequenceNumber = audioSeq
-				ap.Header.Timestamp = ap.Header.Timestamp - s.AudioReader.SkipTs*90
+				ap.Header.Timestamp = createTimestamp(ap.Header.Timestamp)
 				spesic.OnEvent((AudioRTP)(ap))
 				return true
 			})
@@ -221,11 +230,11 @@ func (s *Subscriber) PlayBlock(subType byte) {
 			spesic.OnEvent(append(result, util.PutBE(flvHeadCache[11:15], dataSize+11)))
 		}
 		sendVideoDecConf = func() {
-			sendFlvFrame(codec.FLV_TAG_TYPE_VIDEO, 0, s.VideoReader.Track.SequenceHead)
+			sendFlvFrame(codec.FLV_TAG_TYPE_VIDEO, s.VideoReader.AbsTime, s.VideoReader.Track.SequenceHead)
 			// spesic.OnEvent(FLVFrame(copyBuffers(s.Video.Track.DecoderConfiguration.FLV)))
 		}
 		sendAudioDecConf = func() {
-			sendFlvFrame(codec.FLV_TAG_TYPE_AUDIO, 0, s.AudioReader.Track.SequenceHead)
+			sendFlvFrame(codec.FLV_TAG_TYPE_AUDIO, s.AudioReader.AbsTime, s.AudioReader.Track.SequenceHead)
 			// spesic.OnEvent(FLVFrame(copyBuffers(s.Audio.Track.DecoderConfiguration.FLV)))
 		}
 		sendVideoFrame = func(frame *AVFrame) {
@@ -299,10 +308,12 @@ func (s *Subscriber) PlayBlock(subType byte) {
 				case track.READSTATE_INIT:
 					if s.Video != nil {
 						s.AudioReader.FirstTs = s.VideoReader.FirstTs
+
 					}
 				case track.READSTATE_NORMAL:
 					if s.Video != nil {
 						s.AudioReader.SkipTs = s.VideoReader.SkipTs
+						s.AudioReader.SkipRTPTs = s.AudioReader.Track.Ms2RTPTs(s.AudioReader.SkipTs)
 					}
 				}
 				s.AudioReader.Read(ctx, subMode)
--- a/track/aac.go
+++ b/track/aac.go
@@ -40,9 +40,9 @@ func (aac *AAC) WriteADTS(adts []byte) {
 	channel := ((adts[2] & 0x1) << 2) | ((adts[3] & 0xc0) >> 6)
 	config1 := (profile << 3) | ((sampleRate & 0xe) >> 1)
 	config2 := ((sampleRate & 0x1) << 7) | (channel << 3)
+	aac.Media.WriteSequenceHead([]byte{0xAF, 0x00, config1, config2})
 	aac.SampleRate = uint32(codec.SamplingFrequencies[sampleRate])
 	aac.Channels = channel
-	aac.WriteSequenceHead([]byte{0xAF, 0x00, config1, config2})
 	aac.Parse(aac.SequenceHead[2:])
 	aac.Attach()
 }
@@ -93,6 +93,7 @@ func (aac *AAC) WriteSequenceHead(sh []byte) {
 	config1, config2 := aac.SequenceHead[2], aac.SequenceHead[3]
 	aac.Channels = ((config2 >> 3) & 0x0F) //声道
 	aac.SampleRate = uint32(codec.SamplingFrequencies[((config1&0x7)<<1)|(config2>>7)])
+	aac.ClockRate = aac.SampleRate
 	aac.Parse(aac.SequenceHead[2:])
 	aac.Attach()
 }
--- a/track/audio.go
+++ b/track/audio.go
@@ -11,6 +11,7 @@ type Audio struct {
 	CodecID    codec.AudioCodecID
 	Channels   byte
 	SampleSize byte
+	SampleRate uint32
 	AVCCHead   []byte // 音频包在AVCC格式中，AAC会有两个字节，其他的只有一个字节
 	codec.AudioSpecificConfig
 }
@@ -50,7 +51,7 @@ func (av *Audio) WriteRaw(pts uint32, raw []byte) {

 func (av *Audio) WriteAVCC(ts uint32, frame util.BLL) {
 	av.Value.WriteAVCC(ts, frame)
-	av.generateTimestamp(ts * 90)
+	av.generateTimestamp(ts * av.ClockRate / 1000)
 	av.Flush()
 }

--- a/track/base.go
+++ b/track/base.go
@@ -31,15 +31,15 @@ func (p *流速控制) 控制流速(绝对时间戳 uint32) {
 	// 	return
 	// }
 	// 如果收到的帧的时间戳超过实际消耗的时间100ms就休息一下，100ms作为一个弹性区间防止频繁调用sleep
-	if 过快毫秒 := (数据时间差 - 实际时间差) / time.Millisecond; 过快毫秒 > 100 {
-		// println("过快毫秒", 过快毫秒)
-		if 过快毫秒 > p.等待上限 {
-			time.Sleep(time.Millisecond * p.等待上限)
+	if 过快 := (数据时间差 - 实际时间差); 过快 > 100*time.Millisecond {
+		// println("过快毫秒", 过快)
+		if 过快 > p.等待上限 {
+			time.Sleep(p.等待上限)
 		} else {
-			time.Sleep(过快毫秒 * time.Millisecond)
+			time.Sleep(过快)
 		}
-	} else if 过快毫秒 < -100 {
-		// println("过慢毫秒", 过快毫秒)
+	} else if 过快 < -100*time.Millisecond {
+		// println("过慢毫秒", 过快)
 	}
 }

@@ -74,7 +74,7 @@ type Media struct {
 	Base
 	RingBuffer[AVFrame]
 	IDRingList      `json:"-"` //最近的关键帧位置，首屏渲染
-	SampleRate      uint32
+	ClockRate       uint32     //时钟频率,mpeg中均为90000，rtsp中音频根据sample_rate
 	SSRC            uint32
 	PayloadType     byte
 	BytesPool       util.BytesPool `json:"-"`
@@ -88,6 +88,16 @@ type Media struct {
 	流速控制
 }

+// 毫秒转换为RTP时间戳
+func (av *Media) Ms2RTPTs(ms uint32) uint32 {
+	return uint32(uint64(ms) * uint64(av.ClockRate) / 1000)
+}
+
+// RTP时间戳转换为毫秒
+func (av *Media) RTPTs2Ms(rtpts uint32) uint32 {
+	return uint32(uint64(rtpts) * 1000 / uint64(av.ClockRate))
+}
+
 // 为json序列化而计算的数据
 func (av *Media) SnapForJson() {
 	v := av.LastValue
@@ -103,7 +113,7 @@ func (av *Media) SnapForJson() {
 }

 func (av *Media) SetSpeedLimit(value time.Duration) {
-	av.等待上限 = value * time.Millisecond
+	av.等待上限 = value
 }

 func (av *Media) SetStuff(stuff ...any) {
@@ -115,7 +125,7 @@ func (av *Media) SetStuff(stuff ...any) {
 			av.SSRC = uint32(uintptr(unsafe.Pointer(av)))
 			av.等待上限 = config.Global.SpeedLimit
 		case uint32:
-			av.SampleRate = v
+			av.ClockRate = v
 		case byte:
 			av.PayloadType = v
 		case util.BytesPool:
@@ -195,8 +205,9 @@ func (av *Media) Flush() {
 		av.Info("track back online")
 	}
 	if av.deltaTs != 0 {
-		curValue.DTS = uint32(int64(curValue.DTS) + av.deltaTs*90)
-		curValue.PTS = uint32(int64(curValue.PTS) + av.deltaTs*90)
+		rtpts := int64(av.deltaTs) * int64(av.ClockRate) / 1000
+		curValue.DTS = uint32(int64(curValue.DTS) + rtpts)
+		curValue.PTS = uint32(int64(curValue.PTS) + rtpts)
 		curValue.AbsTime = 0
 	}
 	bufferTime := av.Stream.GetPublisherConfig().BufferTime
@@ -213,6 +224,15 @@ func (av *Media) Flush() {
 		// 	av.Stream.Error("sub ring overflow", zap.Int("size", av.AVRing.Size), zap.String("name", av.Name))
 		// }
 	}
+	if av.起始时间.IsZero() {
+		curValue.DeltaTime = 0
+		av.重置(curValue.AbsTime)
+	} else if curValue.AbsTime == 0 {
+		curValue.DeltaTime = (curValue.DTS - preValue.DTS) * 1000 / av.ClockRate
+		curValue.AbsTime = preValue.AbsTime + curValue.DeltaTime
+	} else {
+		curValue.DeltaTime = curValue.AbsTime - preValue.AbsTime
+	}
 	if curValue.AUList.Length > 0 {
 		// 补完RTP
 		if config.Global.EnableRTP && curValue.RTP.Length == 0 {
@@ -223,15 +243,6 @@ func (av *Media) Flush() {
 			av.CompleteAVCC(curValue)
 		}
 	}
-	if av.起始时间.IsZero() {
-		curValue.DeltaTime = 0
-		av.重置(curValue.AbsTime)
-	} else if curValue.AbsTime == 0 {
-		curValue.DeltaTime = (curValue.DTS - preValue.DTS) / 90
-		curValue.AbsTime = preValue.AbsTime + curValue.DeltaTime
-	} else {
-		curValue.DeltaTime = curValue.AbsTime - preValue.AbsTime
-	}
 	av.Base.Flush(&curValue.BaseFrame)
 	if av.等待上限 > 0 {
 		av.控制流速(curValue.AbsTime)
--- a/track/reader-av.go
+++ b/track/reader-av.go
@@ -25,6 +25,7 @@ type AVRingReader struct {
 	FirstSeq   uint32
 	FirstTs    uint32
 	SkipTs     uint32
+	SkipRTPTs  uint32
 	beforeJump uint32
 	ConfSeq    int
 	startTime  time.Time
@@ -97,6 +98,7 @@ func (r *AVRingReader) Read(ctx context.Context, mode int) (err error) {
 			r.FirstTs = r.Frame.AbsTime
 		}
 		r.SkipTs = r.FirstTs
+		r.SkipRTPTs = r.Track.Ms2RTPTs(r.SkipTs)
 		r.FirstSeq = r.Frame.Sequence
 		r.Track.Info("first frame read", zap.Uint32("firstTs", r.FirstTs), zap.Uint32("firstSeq", r.FirstSeq))
 	case READSTATE_FIRST:
@@ -107,6 +109,7 @@ func (r *AVRingReader) Read(ctx context.Context, mode int) (err error) {
 				return
 			}
 			r.SkipTs = frame.AbsTime - r.beforeJump
+			r.SkipRTPTs = r.Track.Ms2RTPTs(r.SkipTs)
 			r.Track.Info("jump", zap.Uint32("skipSeq", r.Track.IDRing.Value.Sequence-r.FirstSeq), zap.Uint32("skipTs", r.SkipTs))
 			r.State = READSTATE_NORMAL
 		} else {
--- a/track/video.go
+++ b/track/video.go
@@ -114,7 +114,8 @@ func (vt *Video) WriteAVCC(ts uint32, frame util.BLL) error {
 	if err != nil {
 		return err
 	}
-	vt.Value.PTS = (ts + cts) * 90
+	vt.Value.PTS = vt.Ms2RTPTs(ts + cts)
+	vt.Value.DTS = vt.Ms2RTPTs(ts)
 	// println(":", vt.Value.Sequence)
 	for nalulen, err := r.ReadBE(vt.nalulenSize); err == nil; nalulen, err = r.ReadBE(vt.nalulenSize) {
 		// var au util.BLL
@@ -148,30 +149,30 @@ func (vt *Video) WriteSliceByte(b ...byte) {
 }

 // 在I帧前面插入sps pps webrtc需要
-func (av *Video) insertDCRtp() {
-	head := av.Value.RTP.Next
+func (vt *Video) insertDCRtp() {
+	head := vt.Value.RTP.Next
 	seq := head.Value.SequenceNumber
-	for _, nalu := range av.ParamaterSets {
+	for _, nalu := range vt.ParamaterSets {
 		var packet RTPFrame
 		packet.Version = 2
-		packet.PayloadType = av.PayloadType
+		packet.PayloadType = vt.PayloadType
 		packet.Payload = nalu
-		packet.SSRC = av.SSRC
-		packet.Timestamp = av.Value.PTS
+		packet.SSRC = vt.SSRC
+		packet.Timestamp = vt.Value.PTS
 		packet.Marker = false
 		head.InsertBeforeValue(packet)
-		av.rtpSequence++
+		vt.rtpSequence++
 	}
-	av.Value.RTP.RangeItem(func(item *util.ListItem[RTPFrame]) bool {
+	vt.Value.RTP.RangeItem(func(item *util.ListItem[RTPFrame]) bool {
 		item.Value.SequenceNumber = seq
 		seq++
 		return true
 	})
 }

-func (av *Video) generateTimestamp(ts uint32) {
-	av.Value.PTS = ts
-	av.Value.DTS = av.dtsEst.Feed(ts)
+func (vt *Video) generateTimestamp(ts uint32) {
+	vt.Value.PTS = ts
+	vt.Value.DTS = vt.dtsEst.Feed(ts)
 }

 func (vt *Video) SetLostFlag() {
@@ -188,7 +189,7 @@ func (vt *Video) CompleteAVCC(rv *AVFrame) {
 	b[1] = 1
 	// println(rv.PTS < rv.DTS, "\t", rv.PTS, "\t", rv.DTS, "\t", rv.PTS-rv.DTS)
 	// 写入CTS
-	util.PutBE(b[2:5], (rv.PTS-rv.DTS)/90)
+	util.PutBE(b[2:5], vt.RTPTs2Ms(rv.PTS-rv.DTS))
 	rv.AVCC.Push(mem)
 	rv.AUList.Range(func(au *util.BLL) bool {
 		mem = vt.BytesPool.Get(4)