refactor: 拆分avformat依赖库

api.go

@@ -43,7 +43,7 @@ func init() {
 
 func filterSourceID(f func(sourceId string, w http.ResponseWriter, req *http.Request), suffix string) func(http.ResponseWriter, *http.Request) {
 	return func(w http.ResponseWriter, req *http.Request) {
-		source, err := stream.Path2SourceId(req.URL.Path, suffix)
+		source, err := stream.Path2SourceID(req.URL.Path, suffix)
 		if err != nil {
 			log.Sugar.Errorf("拉流失败 解析流id发生err: %s path: %s", err.Error(), req.URL.Path)
 			httpResponse(w, http.StatusBadRequest, err.Error())
@@ -398,7 +398,7 @@ func (api *ApiServer) OnSourceList(w http.ResponseWriter, r *http.Request) {
 		var codecs []string
 		tracks := source.OriginTracks()
 		for _, track := range tracks {
-			codecs = append(codecs, track.Stream.CodecId().String())
+			codecs = append(codecs, track.Stream.CodecID.String())
 		}
 
 		details = append(details, SourceDetails{

bridge.go

@@ -28,7 +28,7 @@ func NewStreamEndInfo(source stream.Source) *stream.StreamEndInfo {
 		timestamp[0] = track.Dts + int64(track.FrameDuration)
 		timestamp[1] = track.Pts + int64(track.FrameDuration)
 
-		info.Timestamps[track.Stream.CodecId()] = timestamp
+		info.Timestamps[track.Stream.CodecID] = timestamp
 	}
 
 	for _, transStream := range streams {

flv/flv_block.go

@@ -7,9 +7,8 @@ import (
 
 const (
 	// HttpFlvBlockHeaderSize 在每块http-flv流的头部，预留指定大小的数据, 用于描述flv数据块的长度信息
-	// http-flv是以文件流的形式传输http流, 格式如下: length\r\n|flv data\r\n
+	// http-flv是以文件流形式传输的http流, 格式如下: length\r\n|flv data\r\n
 	// 我们对http-flv-block的封装: |block size[4]|skip count[2]|length\r\n|flv data\r\n
-	// skip count是因为length长度不固定, 需要一个字段说明, 跳过多少字节才是http-flv数据
 	HttpFlvBlockHeaderSize = 20
 )
 

flv/flv_sink.go

@@ -1,8 +1,8 @@
 package flv
 
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 )
 

flv/http_flv.go

@@ -2,8 +2,10 @@ package flv
 
 import (
 	"encoding/binary"
-	"github.com/lkmio/avformat/libflv"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/flv"
+	"github.com/lkmio/flv/amf0"
 	"github.com/lkmio/lkm/rtmp"
 	"github.com/lkmio/lkm/stream"
 )
@@ -11,13 +13,13 @@ import (
 type TransStream struct {
 	stream.TCPTransStream
 
-	Muxer               libflv.Muxer
+	Muxer                  *flv.Muxer
 	flvHeaderBlock         []byte // 单独保存9个字节长的flv头, 只发一次, 后续恢复推流不再发送
 	flvExtraDataBlock      []byte // metadata和sequence header
-	flvExtraDataTagSize int    // 整个flv tag大小
+	flvExtraDataPreTagSize uint32
 }
 
-func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
+func (t *TransStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	t.ClearOutStreamBuffer()
 
 	var flvTagSize int
@@ -26,17 +28,19 @@ func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error
 	var dts int64
 	var pts int64
 	var keyBuffer bool
+	var frameType int
 
 	dts = packet.ConvertDts(1000)
 	pts = packet.ConvertPts(1000)
-	if utils.AVMediaTypeAudio == packet.MediaType() {
-		flvTagSize = 17 + len(packet.Data())
-		data = packet.Data()
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
-		flvTagSize = t.Muxer.ComputeVideoDataSize(uint32(pts-dts)) + libflv.TagHeaderSize + len(packet.AVCCPacketData())
-
-		data = packet.AVCCPacketData()
-		videoKey = packet.KeyFrame()
+	if utils.AVMediaTypeAudio == packet.MediaType {
+		data = packet.Data
+		flvTagSize = flv.TagHeaderSize + t.Muxer.ComputeAudioDataHeaderSize() + len(packet.Data)
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
+		data = avformat.AnnexBPacket2AVCC(packet)
+		flvTagSize = flv.TagHeaderSize + t.Muxer.ComputeVideoDataHeaderSize(uint32(pts-dts)) + len(data)
+		if videoKey = packet.Key; videoKey {
+			frameType = flv.FrameTypeKeyFrame
+		}
 	}
 
 	// 关键帧都放在切片头部，所以遇到关键帧创建新切片, 发送当前切片剩余流
@@ -64,7 +68,7 @@ func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error
 	// 分配block
 	bytes := t.MWBuffer.Allocate(separatorSize+flvTagSize, dts, videoKey)
 	// 写flv tag
-	n += t.Muxer.Input(bytes[n:], packet.MediaType(), len(data), dts, pts, packet.KeyFrame(), false)
+	n += t.Muxer.Input(bytes[n:], packet.MediaType, len(data), dts, pts, false, frameType)
 	copy(bytes[n:], data)
 
 	// 合并写满再发
@@ -82,43 +86,28 @@ func (t *TransStream) AddTrack(track *stream.Track) error {
 		return err
 	}
 
-	if utils.AVMediaTypeAudio == track.Stream.Type() {
-		t.Muxer.AddAudioTrack(track.Stream.CodecId(), 0, 0, 0)
-	} else if utils.AVMediaTypeVideo == track.Stream.Type() {
-		t.Muxer.AddVideoTrack(track.Stream.CodecId())
+	if utils.AVMediaTypeAudio == track.Stream.MediaType {
+		t.Muxer.AddAudioTrack(track.Stream)
+	} else if utils.AVMediaTypeVideo == track.Stream.MediaType {
+		t.Muxer.AddVideoTrack(track.Stream)
 
-		t.Muxer.MetaData().AddNumberProperty("width", float64(track.Stream.CodecParameters().Width()))
-		t.Muxer.MetaData().AddNumberProperty("height", float64(track.Stream.CodecParameters().Height()))
+		t.Muxer.MetaData().AddNumberProperty("width", float64(track.Stream.CodecParameters.Width()))
+		t.Muxer.MetaData().AddNumberProperty("height", float64(track.Stream.CodecParameters.Height()))
 	}
 	return nil
 }
 
 func (t *TransStream) WriteHeader() error {
 	var header [4096]byte
-	var extraDataSize int
 	size := t.Muxer.WriteHeader(header[:])
+	tags := header[9:size]
 	copy(t.flvHeaderBlock[HttpFlvBlockHeaderSize:], header[:9])
-	copy(t.flvExtraDataBlock[HttpFlvBlockHeaderSize:], header[9:size])
+	copy(t.flvExtraDataBlock[HttpFlvBlockHeaderSize:], tags)
 
-	extraDataSize = HttpFlvBlockHeaderSize + (size - 9)
-	for _, track := range t.BaseTransStream.Tracks {
-		var data []byte
-		if utils.AVMediaTypeAudio == track.Stream.Type() {
-			data = track.Stream.Extra()
-		} else if utils.AVMediaTypeVideo == track.Stream.Type() {
-			data = track.Stream.CodecParameters().MP4ExtraData()
-		}
+	t.flvExtraDataPreTagSize = t.Muxer.PrevTagSize()
 
-		n := t.Muxer.Input(t.flvExtraDataBlock[extraDataSize:], track.Stream.Type(), len(data), 0, 0, false, true)
-		extraDataSize += n
-		copy(t.flvExtraDataBlock[extraDataSize:], data)
-		extraDataSize += len(data)
-		t.flvExtraDataTagSize = n - 15 + len(data) + 11
-	}
-
-	// 加上末尾换行符
-	extraDataSize += 2
-	t.flvExtraDataBlock = t.flvExtraDataBlock[:extraDataSize]
+	// +2 加上末尾换行符
+	t.flvExtraDataBlock = t.flvExtraDataBlock[:HttpFlvBlockHeaderSize+size-9+2]
 	writeSeparator(t.flvHeaderBlock)
 	writeSeparator(t.flvExtraDataBlock)
 
@@ -135,9 +124,9 @@ func (t *TransStream) ReadKeyFrameBuffer() ([][]byte, int64, error) {
 
 	// 发送当前内存池已有的合并写切片
 	t.MWBuffer.ReadSegmentsFromKeyFrameIndex(func(bytes []byte) {
-		if t.OutBufferSize < 1 {
 		// 修改第一个flv tag的pre tag size为sequence header tag size
-			binary.BigEndian.PutUint32(bytes[HttpFlvBlockHeaderSize:], uint32(t.flvExtraDataTagSize))
+		if t.OutBufferSize < 1 {
+			binary.BigEndian.PutUint32(bytes[HttpFlvBlockHeaderSize:], t.flvExtraDataPreTagSize)
 		}
 
 		// 遍历发送合并写切片
@@ -173,9 +162,9 @@ func (t *TransStream) flushSegment() ([]byte, bool) {
 	return FormatSegment(segment), key
 }
 
-func NewHttpTransStream(metadata *libflv.AMF0Object, prevTagSize uint32) stream.TransStream {
+func NewHttpTransStream(metadata *amf0.Object, prevTagSize uint32) stream.TransStream {
 	return &TransStream{
-		Muxer:             libflv.NewMuxerWithPrevTagSize(metadata, prevTagSize),
+		Muxer:             flv.NewMuxerWithPrevTagSize(metadata, prevTagSize),
 		flvHeaderBlock:    make([]byte, 31),
 		flvExtraDataBlock: make([]byte, 4096),
 	}
@@ -183,7 +172,7 @@ func NewHttpTransStream(metadata *libflv.AMF0Object, prevTagSize uint32) stream.
 
 func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
 	var prevTagSize uint32
-	var metaData *libflv.AMF0Object
+	var metaData *amf0.Object
 
 	endInfo := source.GetStreamEndInfo()
 	if endInfo != nil {
@@ -191,7 +180,7 @@ func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtoco
 	}
 
 	if stream.SourceTypeRtmp == source.GetType() {
-		metaData = source.(*rtmp.Publisher).Stack.MetaData()
+		metaData = source.(*rtmp.Publisher).Stack.Metadata()
 	}
 
 	return NewHttpTransStream(metaData, prevTagSize), nil

flv/ws_conn.go

@@ -2,7 +2,7 @@ package flv
 
 import (
 	"github.com/gorilla/websocket"
-	"github.com/lkmio/avformat/transport"
+	"github.com/lkmio/transport"
 	"net"
 	"time"
 )

gb28181/forward_sink.go

@@ -1,11 +1,11 @@
 package gb28181
 
 import (
-	"github.com/lkmio/avformat/librtp"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtp"
+	"github.com/lkmio/transport"
 	"net"
 )
 
@@ -17,7 +17,7 @@ const (
 type ForwardSink struct {
 	stream.BaseSink
 	setup  SetupType
-	socket transport.ITransport
+	socket transport.Transport
 	ssrc   uint32
 }
 
@@ -50,7 +50,7 @@ func (f *ForwardSink) Write(index int, data [][]byte, ts int64) error {
 	}
 
 	// 修改为与上级协商的SSRC
-	librtp.ModifySSRC(data[0], f.ssrc)
+	rtp.ModifySSRC(data[0], f.ssrc)
 
 	if SetupUDP == f.setup {
 		f.socket.(*transport.UDPClient).Write(data[0][2:])
@@ -83,14 +83,14 @@ func NewForwardSink(ssrc uint32, serverAddr string, setup SetupType, sinkId stre
 			return nil, 0, err
 		}
 
-		client, err := TransportManger.NewUDPClient(stream.AppConfig.ListenIP, remoteAddr)
+		client, err := TransportManger.NewUDPClient(remoteAddr)
 		if err != nil {
 			return nil, 0, err
 		}
 
 		sink.socket = client
 	} else if SetupActive == setup {
-		server, err := TransportManger.NewTCPServer(stream.AppConfig.ListenIP)
+		server, err := TransportManger.NewTCPServer()
 		if err != nil {
 			return nil, 0, err
 		}

gb28181/forward_test.go

@@ -3,8 +3,8 @@ package gb28181
 import (
 	"encoding/json"
 	"fmt"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 	"net/http"
 	"os"
@@ -82,8 +82,8 @@ func closeForwardSink(source, sink string) {
 	}
 }
 
-func createTransport(setup string) (transport.ITransport, *os.File) {
-	var socket transport.ITransport
+func createTransport(setup string) (transport.Transport, *os.File) {
+	var socket transport.Transport
 	name := fmt.Sprintf("./gb_forward_ps_%s_%d.raw", setup, time.Now().UnixMilli())
 	file, err := os.OpenFile(name, os.O_CREATE|os.O_RDWR, 0666)
 	if err != nil {
@@ -135,7 +135,7 @@ func TestForwardSink(t *testing.T) {
 
 	for {
 		var ids []string
-		var transports []transport.ITransport
+		var transports []transport.Transport
 		var files []*os.File
 
 		// 三种推流方式都测试

gb28181/publish_test.go

@@ -5,50 +5,20 @@ import (
 	"encoding/binary"
 	"encoding/json"
 	"fmt"
-	"github.com/lkmio/avformat/transport"
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/mpeg"
+	"github.com/lkmio/transport"
+	"github.com/pion/rtp"
 	"io"
 	"net"
 	"net/http"
 	"os"
+	"sort"
 	"testing"
 	"time"
 )
 
-// 输入rtp负载的ps流文件路径, 根据ssrc解析, rtp头不要带扩展
-func readRtpRaw(path string, ssrc uint32, tcp bool, cb func([]byte)) {
-	file, err := os.ReadFile(path)
-	if err != nil {
-		panic(err)
-	}
-
-	var offset int
-	tcpRtp := make([]byte, 1500)
-
-	for i := 0; i < len(file)-4; i++ {
-		if ssrc != binary.BigEndian.Uint32(file[i:]) {
-			continue
-		}
-
-		if i-8 != 0 {
-			var err error
-			rtp := file[offset : i-8]
-
-			if tcp {
-				binary.BigEndian.PutUint16(tcpRtp, uint16(len(rtp)))
-				copy(tcpRtp[2:], rtp)
-				cb(tcpRtp[:2+len(rtp)])
-			} else {
-				cb(rtp)
-			}
-
-			if err != nil {
-				panic(err.Error())
-			}
-		}
-		offset = i - 8
-	}
-}
-
 func connectSource(source string, addr string) {
 	v := &struct {
 		Source     string `json:"source"` //GetSourceID
@@ -131,32 +101,137 @@ func createSource(source, setup string, ssrc uint32) (string, uint16) {
 	return connectInfo.Data.IP, connectInfo.Data.Port
 }
 
-func rtp2overTcp(path string, ssrc uint32) {
-	file, err := os.OpenFile("./rtp.raw", os.O_CREATE|os.O_RDWR, 0666)
+// 分割rtp包, 返回rtp over tcp包
+func splitPackets(data []byte, ssrc uint32) ([][]byte, uint32) {
+	tcp := binary.BigEndian.Uint16(data) <= 1500
+	length := len(data)
+	var packets [][]byte
+	if tcp {
+		var offset int
+		for i := 0; i < length; i += 2 {
+			if i > 0 {
+				packets = append(packets, data[offset:i])
+			}
+
+			offset = i
+			i += int(binary.BigEndian.Uint16(data[i:]))
+		}
+
+		if len(packets) > 0 {
+			packet := rtp.Packet{}
+			err := packet.Unmarshal(packets[0][2:])
 			if err != nil {
 				panic(err)
 			}
 
-	readRtpRaw(path, ssrc, true, func(data []byte) {
-		file.Write(data)
-	})
+			return packets, packet.SSRC
+		}
+	} else {
+		// udp包根据ssrc查找
+		var offset int
+		for i := 0; i < length-4; i++ {
+			if ssrc != binary.BigEndian.Uint32(data[i:]) {
+				continue
+			}
 
-	file.Close()
+			if i-8 != 0 {
+				packet := data[offset : i-8]
+				bytes := make([]byte, 2+len(packet))
+				binary.BigEndian.PutUint16(bytes, uint16(len(packet)))
+				copy(bytes[2:], packet)
+				packets = append(packets, bytes)
+			}
+
+			offset = i - 8
+		}
+
+		return packets, ssrc
+	}
+
+	return nil, ssrc
 }
 
-// 使用wireshark直接导出udp流
+var ts int64 = -1
+
+func ctrDelay(data []byte) {
+	packet := rtp.Packet{}
+	err := packet.Unmarshal(data)
+	if err != nil {
+		panic(err)
+	}
+
+	if ts == -1 {
+		ts = int64(packet.Timestamp)
+	}
+
+	if dis := (int64(packet.Timestamp) - ts) / 90; dis > 0 {
+		time.Sleep(time.Duration(dis) * time.Millisecond)
+	}
+
+	ts = int64(packet.Timestamp)
+}
+
+// 使用wireshark直接导出的rtp流
 // 根据ssrc来查找每个rtp包, rtp不要带扩展字段
-func TestUDPRecv(t *testing.T) {
-	path := "D:\\GOProjects\\avformat\\gb28181_h264.rtp"
-	ssrc := 0xBEBC201
+func TestPublish(t *testing.T) {
+	path := "../../source_files/gb28181_h264.rtp"
+	var ssrc uint32 = 0xBEBC201
 	localAddr := "0.0.0.0:20001"
-	setup := "udp" //udp/passive/active
 	id := "hls_mystream"
 
-	rtp2overTcp(path, uint32(ssrc))
-	ip, port := createSource(id, setup, uint32(ssrc))
+	data, err := os.ReadFile(path)
+	if err != nil {
+		panic(err)
+	}
+
+	var packets [][]byte
+	packets, ssrc = splitPackets(data, ssrc)
+	utils.Assert(len(packets) > 0)
+
+	sort.Slice(packets, func(i, j int) bool {
+		packet := rtp.Packet{}
+		if err := packet.Unmarshal(packets[i][2:]); err != nil {
+			panic(err)
+		}
+		packet2 := rtp.Packet{}
+		if err := packet2.Unmarshal(packets[j][2:]); err != nil {
+			panic(err)
+		}
+
+		return packet.SequenceNumber < packet2.SequenceNumber
+	})
+
+	t.Run("demux", func(t *testing.T) {
+		buffer := mpeg.NewProbeBuffer(1024 * 1024 * 2)
+		demuxer := mpeg.NewPSDemuxer(true)
+		demuxer.SetHandler(&avformat.OnUnpackStream2FileHandler{
+			Path: "./ps_demux",
+		})
+
+		file, err := os.OpenFile("./ps_demux.ps", os.O_WRONLY|os.O_CREATE, 132)
+		if err != nil {
+			panic(err)
+		}
+
+		for _, packet := range packets {
+			file.Write(packet[14:])
+			bytes, err := buffer.Input(packet[14:])
+			if err != nil {
+				panic(err)
+			}
+
+			n, err := demuxer.Input(bytes)
+			if err != nil {
+				panic(err)
+			}
+
+			buffer.Reset(n)
+		}
+	})
+
+	t.Run("udp", func(t *testing.T) {
+		ip, port := createSource(id, "udp", ssrc)
 
-	if setup == "udp" {
 		addr, _ := net.ResolveUDPAddr("udp", localAddr)
 		remoteAddr, _ := net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", ip, port))
 
@@ -166,11 +241,15 @@ func TestUDPRecv(t *testing.T) {
 			panic(err)
 		}
 
-		readRtpRaw(path, uint32(ssrc), false, func(data []byte) {
-			client.Write(data)
-			time.Sleep(1 * time.Millisecond)
+		for _, packet := range packets {
+			client.Write(packet[2:])
+			ctrDelay(packet[2:])
+		}
 	})
-	} else if !(setup == "active") {
+
+	t.Run("passive", func(t *testing.T) {
+		ip, port := createSource(id, "passive", ssrc)
+
 		addr, _ := net.ResolveTCPAddr("tcp", localAddr)
 		remoteAddr, _ := net.ResolveTCPAddr("tcp", fmt.Sprintf("%s:%d", ip, port))
 
@@ -181,19 +260,23 @@ func TestUDPRecv(t *testing.T) {
 			panic(err)
 		}
 
-		readRtpRaw(path, uint32(ssrc), true, func(data []byte) {
-			client.Write(data)
-			time.Sleep(1 * time.Millisecond)
+		for _, packet := range packets {
+			client.Write(packet)
+			ctrDelay(packet[2:])
+		}
 	})
-	} else {
+
+	t.Run("active", func(t *testing.T) {
+		ip, port := createSource(id, "active", ssrc)
+
 		addr, _ := net.ResolveTCPAddr("tcp", localAddr)
 		server := transport.TCPServer{}
 
 		server.SetHandler2(func(conn net.Conn) []byte {
-			readRtpRaw(path, uint32(ssrc), true, func(data []byte) {
-				conn.Write(data)
-				time.Sleep(1 * time.Millisecond)
-			})
+			for _, packet := range packets {
+				conn.Write(packet)
+				ctrDelay(packet[2:])
+			}
 
 			return nil
 		}, nil, nil)
@@ -204,7 +287,5 @@ func TestUDPRecv(t *testing.T) {
 		}
 
 		connectSource(id, fmt.Sprintf("%s:%d", ip, port))
-	}
-
-	select {}
+	})
 }

gb28181/source.go

@@ -2,11 +2,12 @@ package gb28181
 
 import (
 	"fmt"
-	"github.com/lkmio/avformat/libmpeg"
-	"github.com/lkmio/avformat/transport"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/mpeg"
+	"github.com/lkmio/transport"
 	"github.com/pion/rtp"
 	"math"
 	"net"
@@ -49,12 +50,10 @@ type GBSource interface {
 type BaseGBSource struct {
 	stream.PublishSource
 
-	deMuxerCtx  *libmpeg.PSDeMuxerContext
-	audioStream utils.AVStream
-	videoStream utils.AVStream
+	probeBuffer *mpeg.PSProbeBuffer
 
 	ssrc      uint32
-	transport transport.ITransport
+	transport transport.Transport
 
 	audioTimestamp         int64
 	videoTimestamp         int64
@@ -64,9 +63,13 @@ type BaseGBSource struct {
 }
 
 func (source *BaseGBSource) Init(receiveQueueSize int) {
-	source.deMuxerCtx = libmpeg.NewPSDeMuxerContext(make([]byte, PsProbeBufferSize))
-	source.deMuxerCtx.SetHandler(source)
+	source.TransDemuxer = mpeg.NewPSDemuxer(false)
+	source.TransDemuxer.SetHandler(source)
+	source.TransDemuxer.SetOnPreprocessPacketHandler(func(packet *avformat.AVPacket) {
+		source.correctTimestamp(packet, packet.Dts, packet.Pts)
+	})
 	source.SetType(stream.SourceType28181)
+	source.probeBuffer = mpeg.NewProbeBuffer(PsProbeBufferSize)
 	source.PublishSource.Init(receiveQueueSize)
 }
 
@@ -85,7 +88,14 @@ func (source *BaseGBSource) Input(data []byte) error {
 
 	packet := rtp.Packet{}
 	_ = packet.Unmarshal(data)
-	err := source.deMuxerCtx.Input(packet.Payload)
+
+	var bytes []byte
+	var n int
+	var err error
+	bytes, err = source.probeBuffer.Input(packet.Payload)
+	if err == nil {
+		n, err = source.TransDemuxer.Input(bytes)
+	}
 
 	// 非解析缓冲区满的错误, 继续解析
 	if err != nil {
@@ -94,94 +104,25 @@ func (source *BaseGBSource) Input(data []byte) error {
 			return err
 		}
 	}
-	return nil
-}
 
-// OnPartPacket 部分es流回调
-func (source *BaseGBSource) OnPartPacket(index int, mediaType utils.AVMediaType, codec utils.AVCodecID, data []byte, first bool) {
-	buffer := source.FindOrCreatePacketBuffer(index, mediaType)
-
-	// 第一个es包, 标记内存起始位置
-	if first {
-		buffer.Mark()
-	}
-
-	buffer.Write(data)
-}
-
-// OnLossPacket 非完整es包丢弃回调, 直接释放内存块
-func (source *BaseGBSource) OnLossPacket(index int, mediaType utils.AVMediaType, codec utils.AVCodecID) {
-	buffer := source.FindOrCreatePacketBuffer(index, mediaType)
-
-	buffer.Fetch()
-	buffer.FreeTail()
-}
-
-// OnCompletePacket 完整帧回调
-func (source *BaseGBSource) OnCompletePacket(index int, mediaType utils.AVMediaType, codec utils.AVCodecID, dts int64, pts int64, key bool) error {
-	buffer := source.FindOrCreatePacketBuffer(index, mediaType)
-	data := buffer.Fetch()
-
-	var packet utils.AVPacket
-	var stream_ utils.AVStream
-	var err error
-
-	defer func() {
-		if packet == nil {
-			buffer.FreeTail()
-		}
-	}()
-
-	if utils.AVMediaTypeAudio == mediaType {
-		stream_, packet, err = stream.ExtractAudioPacket(codec, source.audioStream == nil, data, pts, dts, index, 90000)
-		if err != nil {
-			return err
-		}
-
-		if stream_ != nil {
-			source.audioStream = stream_
-		}
-	} else {
-		if source.videoStream == nil && !key {
-			log.Sugar.Errorf("skip non keyframes conn:%s", source.Conn.RemoteAddr())
-			return nil
-		}
-
-		stream_, packet, err = stream.ExtractVideoPacket(codec, key, source.videoStream == nil, data, pts, dts, index, 90000)
-		if err != nil {
-			return err
-		}
-		if stream_ != nil {
-			source.videoStream = stream_
-		}
-	}
-
-	if stream_ != nil {
-		source.OnDeMuxStream(stream_)
-		if len(source.OriginTracks()) >= source.deMuxerCtx.TrackCount() {
-			source.OnDeMuxStreamDone()
-		}
-	}
-
-	source.correctTimestamp(packet, dts, pts)
-	source.OnDeMuxPacket(packet)
+	source.probeBuffer.Reset(n)
 	return nil
 }
 
 // 纠正国标推流的时间戳
-func (source *BaseGBSource) correctTimestamp(packet utils.AVPacket, dts, pts int64) {
+func (source *BaseGBSource) correctTimestamp(packet *avformat.AVPacket, dts, pts int64) {
 	// dts和pts保持一致
 	pts = int64(math.Max(float64(dts), float64(pts)))
 	dts = pts
-	packet.SetPts(pts)
-	packet.SetDts(dts)
+	packet.Pts = pts
+	packet.Dts = dts
 
 	var lastTimestamp int64
 	var lastCreatedTime int64
-	if utils.AVMediaTypeAudio == packet.MediaType() {
+	if utils.AVMediaTypeAudio == packet.MediaType {
 		lastTimestamp = source.audioTimestamp
 		lastCreatedTime = source.audioPacketCreatedTime
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
 		lastTimestamp = source.videoTimestamp
 		lastCreatedTime = source.videoPacketCreatedTime
 	}
@@ -193,38 +134,38 @@ func (source *BaseGBSource) correctTimestamp(packet utils.AVPacket, dts, pts int
 			duration = 0x1FFFFFFFF - lastTimestamp + pts
 			if duration < 90000 {
 				// 处理正常溢出
-				packet.SetDuration(duration)
+				packet.Duration = duration
 			} else {
 				// 时间戳不正确
-				log.Sugar.Errorf("推流时间戳不正确, 使用系统时钟. ssrc:%d", source.ssrc)
+				log.Sugar.Errorf("推流时间戳不正确, 使用系统时钟. ssrc: %x", source.ssrc)
 				source.isSystemClock = true
 			}
 		} else {
 			duration = pts - lastTimestamp
 		}
 
-		packet.SetDuration(duration)
-		duration = packet.Duration(90000)
+		packet.Duration = duration
+		duration = packet.GetDuration(90000)
 		if duration < 0 || duration < 750 {
-			log.Sugar.Errorf("推流时间戳不正确, 使用系统时钟. ts: %d duration: %d source: %s ssrc: %d", pts, duration, source.ID, source.ssrc)
+			log.Sugar.Errorf("推流时间戳不正确, 使用系统时钟. ts: %d duration: %d source: %s ssrc: %x", pts, duration, source.ID, source.ssrc)
 			source.isSystemClock = true
 		}
 	}
 
 	// 纠正时间戳
 	if source.isSystemClock && lastTimestamp != -1 {
-		duration = (packet.CreatedTime() - lastCreatedTime) * 90
-		packet.SetDts(lastTimestamp + duration)
-		packet.SetPts(lastTimestamp + duration)
-		packet.SetDuration(duration)
+		duration = (packet.CreatedTime - lastCreatedTime) * 90
+		packet.Dts = lastTimestamp + duration
+		packet.Pts = lastTimestamp + duration
+		packet.Duration = duration
 	}
 
-	if utils.AVMediaTypeAudio == packet.MediaType() {
-		source.audioTimestamp = packet.Pts()
-		source.audioPacketCreatedTime = packet.CreatedTime()
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
-		source.videoTimestamp = packet.Pts()
-		source.videoPacketCreatedTime = packet.CreatedTime()
+	if utils.AVMediaTypeAudio == packet.MediaType {
+		source.audioTimestamp = packet.Pts
+		source.audioPacketCreatedTime = packet.CreatedTime
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
+		source.videoTimestamp = packet.Pts
+		source.videoPacketCreatedTime = packet.CreatedTime
 	}
 }
 
@@ -249,11 +190,6 @@ func (source *BaseGBSource) Close() {
 	}
 
 	source.PublishSource.Close()
-
-	if source.deMuxerCtx != nil {
-		source.deMuxerCtx.Close()
-		source.deMuxerCtx = nil
-	}
 }
 
 func (source *BaseGBSource) SetConn(conn net.Conn) {

gb28181/source_active.go

@@ -1,6 +1,7 @@
 package gb28181
 
 import (
+	"github.com/lkmio/transport"
 	"net"
 )
 
@@ -12,8 +13,8 @@ type ActiveSource struct {
 	tcp        *TCPClient
 }
 
-func (a ActiveSource) Connect(remoteAddr *net.TCPAddr) error {
-	client, err := NewTCPClient(a.port, remoteAddr, &a)
+func (a *ActiveSource) Connect(remoteAddr *net.TCPAddr) error {
+	client, err := NewTCPClient(a.port, remoteAddr, a)
 	if err != nil {
 		return err
 	}
@@ -22,7 +23,7 @@ func (a ActiveSource) Connect(remoteAddr *net.TCPAddr) error {
 	return nil
 }
 
-func (a ActiveSource) SetupType() SetupType {
+func (a *ActiveSource) SetupType() SetupType {
 	return SetupActive
 }
 
@@ -33,5 +34,10 @@ func NewActiveSource() (*ActiveSource, int, error) {
 		return nil
 	})
 
-	return &ActiveSource{port: port}, port, nil
+	return &ActiveSource{
+		PassiveSource: PassiveSource{
+			decoder: transport.NewLengthFieldFrameDecoder(0xFFFF, 2),
+		},
+		port: port,
+	}, port, nil
 }

gb28181/source_passive.go

@@ -1,21 +1,24 @@
 package gb28181
 
-import "github.com/lkmio/avformat/transport"
+import "github.com/lkmio/transport"
 
 type PassiveSource struct {
 	BaseGBSource
 	decoder *transport.LengthFieldFrameDecoder
 }
 
-// Input 重写stream.Source的Input函数, 主解析将会把推流数据交给PassiveSource处理
-func (p PassiveSource) Input(data []byte) error {
-	return p.decoder.Input(data)
+// Input 重写stream.Source的Input函数, 主协程把推流数据交给PassiveSource处理
+func (p *PassiveSource) Input(data []byte) error {
+	_, err := DecodeGBRTPOverTCPPacket(data, p, p.decoder, nil, p.Conn)
+	return err
 }
 
-func (p PassiveSource) SetupType() SetupType {
+func (p *PassiveSource) SetupType() SetupType {
 	return SetupPassive
 }
 
 func NewPassiveSource() *PassiveSource {
-	return &PassiveSource{}
+	return &PassiveSource{
+		decoder: transport.NewLengthFieldFrameDecoder(0xFFFF, 2),
+	}
 }

gb28181/tcp_client.go

@@ -1,8 +1,8 @@
 package gb28181
 
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 )
 

gb28181/tcp_server.go

@@ -1,9 +1,10 @@
 package gb28181
 
 import (
-	"github.com/lkmio/avformat/transport"
+	"encoding/hex"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 	"runtime"
 )
@@ -49,11 +50,16 @@ func (T *TCPServer) OnPacket(conn net.Conn, data []byte) []byte {
 
 	// 单端口推流时, 先解析出SSRC找到GBSource. 后序将推流数据交给stream.Source处理
 	if session.source == nil {
-		if err := session.decoder.Input(data); err != nil {
-			log.Sugar.Errorf("解析粘包数据失败 err: %s", err.Error())
-			conn.Close()
+		source, err := DecodeGBRTPOverTCPPacket(data, nil, session.decoder, T.filter, conn)
+		if err != nil {
+			log.Sugar.Errorf("解析rtp失败 err: %s conn: %s data: %s", err.Error(), conn.RemoteAddr().String(), hex.EncodeToString(data))
+			_ = conn.Close()
+			return nil
 		}
 
+		if source != nil {
+			session.Init(source)
+		}
 	} else {
 		// 将流交给Source的主协程处理，主协程最终会调用PassiveSource的Input函数处理
 		if session.source.SetupType() == SetupPassive {
@@ -80,7 +86,7 @@ func NewTCPServer(filter Filter) (*TCPServer, error) {
 	var err error
 	if stream.AppConfig.GB28181.IsMultiPort() {
 		tcp = &transport.TCPServer{}
-		tcp, err = TransportManger.NewTCPServer(stream.AppConfig.ListenIP)
+		tcp, err = TransportManger.NewTCPServer()
 		if err != nil {
 			return nil, err
 		}

gb28181/tcp_session.go

@@ -1,10 +1,9 @@
 package gb28181
 
 import (
-	"encoding/hex"
-	"github.com/lkmio/avformat/transport"
-	"github.com/lkmio/lkm/log"
+	"fmt"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"github.com/pion/rtp"
 	"net"
 )
@@ -13,20 +12,14 @@ import (
 type TCPSession struct {
 	conn          net.Conn
 	source        GBSource
-	decoder       *transport.LengthFieldFrameDecoder
 	receiveBuffer *stream.ReceiveBuffer
+	decoder       *transport.LengthFieldFrameDecoder
 }
 
 func (t *TCPSession) Init(source GBSource) {
 	t.source = source
 	// 创建收流缓冲区
 	t.receiveBuffer = stream.NewTCPReceiveBuffer()
-	// session关联到source后, 后续流数据都由source处理
-	if source.SetupType() == SetupPassive {
-		source.(*PassiveSource).decoder = t.decoder
-	} else {
-		source.(*ActiveSource).decoder = t.decoder
-	}
 }
 
 func (t *TCPSession) Close() {
@@ -35,52 +28,55 @@ func (t *TCPSession) Close() {
 		t.source.Close()
 		t.source = nil
 	}
+}
 
-	if t.decoder != nil {
-		t.decoder.Close()
-		t.decoder = nil
+func DecodeGBRTPOverTCPPacket(data []byte, source GBSource, decoder *transport.LengthFieldFrameDecoder, filter Filter, conn net.Conn) (GBSource, error) {
+	length := len(data)
+	for i := 0; i < length; {
+		n, bytes, err := decoder.Input(data[i:])
+		if err != nil {
+			return source, err
 		}
+
+		i += n
+
+		// 单端口模式,ssrc匹配source
+		if source == nil || stream.SessionStateHandshakeSuccess == source.State() {
+			packet := rtp.Packet{}
+			if err := packet.Unmarshal(bytes); err != nil {
+				return nil, err
+			} else if source == nil {
+				source = filter.FindSource(packet.SSRC)
+			}
+
+			if source == nil {
+				// ssrc 匹配不到Source
+				return nil, fmt.Errorf("gb28181推流失败 ssrc: %x 匹配不到source", packet.SSRC)
+			}
+
+			if stream.SessionStateHandshakeSuccess == source.State() {
+				source.PreparePublish(conn, packet.SSRC, source)
+			}
+		}
+
+		// 如果是单端口推流, 并且刚才与source绑定, 此时正位于网络收流协程, 否则都位于主协程
+		if source.SetupType() == SetupPassive {
+			source.(*PassiveSource).BaseGBSource.Input(bytes)
+		} else {
+			source.(*ActiveSource).BaseGBSource.Input(bytes)
+		}
+	}
+
+	return source, nil
 }
 
 func NewTCPSession(conn net.Conn, filter Filter) *TCPSession {
 	session := &TCPSession{
 		conn: conn,
+		// filter:  filter,
+		decoder: transport.NewLengthFieldFrameDecoder(0xFFFF, 2),
 	}
 
-	// 创建粘包解码器, 并设置解粘包处理回调
-	session.decoder = transport.NewLengthFieldFrameDecoder(0xFFFF, 2, func(bytes []byte) {
-		// 单端口模式,ssrc匹配source
-		if session.source == nil || stream.SessionStateHandshakeSuccess == session.source.State() {
-			packet := rtp.Packet{}
-			if err := packet.Unmarshal(bytes); err != nil {
-				log.Sugar.Errorf("解析rtp失败 err: %s conn: %s data: %s", err.Error(), conn.RemoteAddr().String(), hex.EncodeToString(bytes))
-				conn.Close()
-				return
-			}
-
-			source := filter.FindSource(packet.SSRC)
-			if source == nil {
-				// 匹配不到Source, 直接关闭连接
-				log.Sugar.Errorf("gb28181推流失败 ssrc: %x 匹配不到source conn: %s  data: %s", packet.SSRC, session.conn.RemoteAddr().String(), hex.EncodeToString(bytes))
-				conn.Close()
-				return
-			}
-
-			session.Init(source)
-
-			if stream.SessionStateHandshakeSuccess == session.source.State() {
-				session.source.PreparePublish(session.conn, packet.SSRC, session.source)
-			}
-		}
-
-		// 如果是单端口推流, 并且刚才与source绑定, 此时正位于网络收流协程, 否则都位于主协程
-		if session.source.SetupType() == SetupPassive {
-			session.source.(*PassiveSource).BaseGBSource.Input(bytes)
-		} else {
-			session.source.(*ActiveSource).BaseGBSource.Input(bytes)
-		}
-	})
-
 	// 多端口收流, Source已知, 直接初始化Session
 	if stream.AppConfig.GB28181.IsMultiPort() {
 		session.Init(filter.(*singleFilter).source)

gb28181/udp_server.go

@@ -1,9 +1,9 @@
 package gb28181
 
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"github.com/pion/rtp"
 	"net"
 	"runtime"
@@ -64,7 +64,7 @@ func NewUDPServer(filter Filter) (*UDPServer, error) {
 	var udp *transport.UDPServer
 	var err error
 	if stream.AppConfig.GB28181.IsMultiPort() {
-		udp, err = TransportManger.NewUDPServer(stream.AppConfig.ListenIP)
+		udp, err = TransportManger.NewUDPServer()
 		if err != nil {
 			return nil, err
 		}

go.mod

@@ -1,13 +1,19 @@
 module github.com/lkmio/lkm
 
-require github.com/lkmio/avformat v0.0.0
+require (
+	github.com/lkmio/mpeg v0.0.0
+	github.com/lkmio/flv v0.0.0
+	github.com/lkmio/rtmp v0.0.0
+	github.com/lkmio/transport v0.0.0
+	github.com/lkmio/rtp v0.0.0
+)
 
 require (
 	github.com/gorilla/mux v1.8.1
 	github.com/gorilla/websocket v1.5.1
+	github.com/lkmio/avformat v0.0.0
 	github.com/natefinch/lumberjack v2.0.0+incompatible
-	github.com/pion/rtcp v1.2.14
-	github.com/pion/rtp v1.8.5
+	github.com/pion/interceptor v0.1.25
 	github.com/pion/webrtc/v3 v3.2.29
 	github.com/sirupsen/logrus v1.9.3
 	github.com/x-cray/logrus-prefixed-formatter v0.5.2
@@ -24,10 +30,11 @@ require (
 	github.com/pion/datachannel v1.5.5 // indirect
 	github.com/pion/dtls/v2 v2.2.7 // indirect
 	github.com/pion/ice/v2 v2.3.13 // indirect
-	github.com/pion/interceptor v0.1.25 // indirect
 	github.com/pion/logging v0.2.2 // indirect
 	github.com/pion/mdns v0.0.12 // indirect
 	github.com/pion/randutil v0.1.0 // indirect
+	github.com/pion/rtcp v1.2.12 // indirect
+	github.com/pion/rtp v1.8.5 // indirect
 	github.com/pion/sctp v1.8.12 // indirect
 	github.com/pion/sdp/v3 v3.0.8 // indirect
 	github.com/pion/srtp/v2 v2.0.18 // indirect
@@ -47,4 +54,14 @@ require (
 
 replace github.com/lkmio/avformat => ../avformat
 
+replace github.com/lkmio/mpeg => ../mpeg
+
+replace github.com/lkmio/flv => ../flv
+
+replace github.com/lkmio/rtmp => ../rtmp
+
+replace github.com/lkmio/transport => ../transport
+
+replace github.com/lkmio/rtp => ../rtp
+
 go 1.19

hls/hls_stream.go

@@ -2,32 +2,29 @@ package hls
 
 import (
 	"fmt"
-	"github.com/lkmio/avformat/libhls"
-	"github.com/lkmio/avformat/libmpeg"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/mpeg"
 	"os"
 	"path/filepath"
 	"strconv"
 )
 
-type tsContext struct {
+type TransStream struct {
+	stream.BaseTransStream
+	muxer *mpeg.TSMuxer
+	ctx   struct {
 		segmentSeq      int      // 切片序号
 		writeBuffer     []byte   // ts流的缓冲区, 由TSMuxer使用. 减少用户态和内核态交互，以及磁盘IO频率
 		writeBufferSize int      // 已缓存TS流大小
-
 		url             string   // @See TransStream.tsUrl
 		path            string   // ts切片位于磁盘中的绝对路径
 		file            *os.File // ts切片文件句柄
-}
+	}
 
-type TransStream struct {
-	stream.BaseTransStream
-	muxer   libmpeg.TSMuxer
-	context *tsContext
-
-	M3U8Writer     libhls.M3U8Writer
+	M3U8Writer     stream.M3U8Writer
 	m3u8Name       string   // m3u8文件名
 	m3u8File       *os.File // m3u8文件句柄
 	dir            string   // m3u8文件父目录
@@ -40,16 +37,12 @@ type TransStream struct {
 	PlaylistFormat *string                     // 位于内存中的播放列表，每个sink都引用指针地址.
 }
 
-func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
-	if packet.Index() >= t.muxer.TrackCount() {
-		return nil, -1, false, fmt.Errorf("track not available")
-	}
-
+func (t *TransStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	// 创建一下个切片
 	// 已缓存时长>=指定时长, 如果存在视频, 还需要等遇到关键帧才切片
-	if (!t.ExistVideo || utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame()) && float32(t.muxer.Duration())/90000 >= float32(t.duration) {
+	if (!t.ExistVideo || utils.AVMediaTypeVideo == packet.MediaType && packet.Key) && float32(t.muxer.Duration())/90000 >= float32(t.duration) {
 		// 保存当前切片文件
-		if t.context.file != nil {
+		if t.ctx.file != nil {
 			err := t.flushSegment(false)
 			if err != nil {
 				return nil, -1, false, err
@@ -64,12 +57,23 @@ func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error
 
 	pts := packet.ConvertPts(90000)
 	dts := packet.ConvertDts(90000)
-	if utils.AVMediaTypeVideo == packet.MediaType() {
-		t.muxer.Input(packet.Index(), packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()].Stream), pts, dts, packet.KeyFrame())
-	} else {
-		t.muxer.Input(packet.Index(), packet.Data(), pts, dts, packet.KeyFrame())
+	data := packet.Data
+	if utils.AVMediaTypeVideo == packet.MediaType {
+		data = avformat.AVCCPacket2AnnexB(t.BaseTransStream.Tracks[packet.Index].Stream, packet)
 	}
 
+	length := len(data)
+	capacity := cap(t.ctx.writeBuffer)
+	for i := 0; i < length; {
+		if capacity-t.ctx.writeBufferSize < mpeg.TsPacketSize {
+			_, _ = t.ctx.file.Write(t.ctx.writeBuffer[:t.ctx.writeBufferSize])
+			t.ctx.writeBufferSize = 0
+		}
+
+		bytes := t.ctx.writeBuffer[t.ctx.writeBufferSize : t.ctx.writeBufferSize+mpeg.TsPacketSize]
+		i += t.muxer.Input(bytes, packet.Index, data[i:], length, dts, pts, packet.Key, i == 0)
+		t.ctx.writeBufferSize += mpeg.TsPacketSize
+	}
 	return nil, -1, true, nil
 }
 
@@ -79,42 +83,31 @@ func (t *TransStream) AddTrack(track *stream.Track) error {
 	}
 
 	var err error
-	if utils.AVMediaTypeVideo == track.Stream.Type() {
-		data := track.Stream.CodecParameters().AnnexBExtraData()
-		_, err = t.muxer.AddTrack(track.Stream.Type(), track.Stream.CodecId(), data)
+	if utils.AVMediaTypeVideo == track.Stream.MediaType {
+		data := track.Stream.CodecParameters.AnnexBExtraData()
+		_, err = t.muxer.AddTrack(track.Stream.MediaType, track.Stream.CodecID, data)
 	} else {
-		_, err = t.muxer.AddTrack(track.Stream.Type(), track.Stream.CodecId(), track.Stream.Extra())
+		_, err = t.muxer.AddTrack(track.Stream.MediaType, track.Stream.CodecID, track.Stream.Data)
 	}
 	return err
 }
 
 func (t *TransStream) WriteHeader() error {
+	//if packet.Index >= t.muxer.TrackCount() {
+	//	return nil, -1, false, fmt.Errorf("track not available")
+	//}
 	return t.createSegment()
 }
 
-func (t *TransStream) onTSWrite(data []byte) {
-	t.context.writeBufferSize += len(data)
-}
-
-func (t *TransStream) onTSAlloc(size int) []byte {
-	n := len(t.context.writeBuffer) - t.context.writeBufferSize
-	if n < size {
-		_, _ = t.context.file.Write(t.context.writeBuffer[:t.context.writeBufferSize])
-		t.context.writeBufferSize = 0
-	}
-
-	return t.context.writeBuffer[t.context.writeBufferSize : t.context.writeBufferSize+size]
-}
-
 // 写入新的TS切片，更新M3U8
 func (t *TransStream) flushSegment(end bool) error {
 	// 写入剩余TS包
-	if t.context.writeBufferSize > 0 {
-		_, _ = t.context.file.Write(t.context.writeBuffer[:t.context.writeBufferSize])
-		t.context.writeBufferSize = 0
+	if t.ctx.writeBufferSize > 0 {
+		_, _ = t.ctx.file.Write(t.ctx.writeBuffer[:t.ctx.writeBufferSize])
+		t.ctx.writeBufferSize = 0
 	}
 
-	if err := t.context.file.Close(); err != nil {
+	if err := t.ctx.file.Close(); err != nil {
 		return err
 	}
 
@@ -125,7 +118,7 @@ func (t *TransStream) flushSegment(end bool) error {
 
 	// 更新m3u8列表
 	duration := float32(t.muxer.Duration()) / 90000
-	t.M3U8Writer.AddSegment(duration, t.context.url, t.context.segmentSeq, t.context.path)
+	t.M3U8Writer.AddSegment(duration, t.ctx.url, t.ctx.segmentSeq, t.ctx.path)
 	m3u8Txt := t.M3U8Writer.String()
 
 	//if end {
@@ -163,21 +156,21 @@ func (t *TransStream) createSegment() error {
 	t.muxer.Reset()
 
 	var tsFile *os.File
-	startSeq := t.context.segmentSeq + 1
+	startSeq := t.ctx.segmentSeq + 1
 	for {
 		tsName := fmt.Sprintf(t.tsFormat, startSeq)
 		// ts文件
-		t.context.path = fmt.Sprintf("%s/%s", t.dir, tsName)
+		t.ctx.path = fmt.Sprintf("%s/%s", t.dir, tsName)
 		// m3u8列表中切片的url
-		t.context.url = fmt.Sprintf("%s%s", t.tsUrl, tsName)
+		t.ctx.url = fmt.Sprintf("%s%s", t.tsUrl, tsName)
 
-		file, err := os.OpenFile(t.context.path, os.O_WRONLY|os.O_CREATE|os.O_TRUNC, 0666)
+		file, err := os.OpenFile(t.ctx.path, os.O_WRONLY|os.O_CREATE|os.O_TRUNC, 0666)
 		if err == nil {
 			tsFile = file
 			break
 		}
 
-		log.Sugar.Errorf("创建ts切片文件失败 err:%s path:%s", err.Error(), t.context.path)
+		log.Sugar.Errorf("创建ts切片文件失败 err:%s path:%s", err.Error(), t.ctx.path)
 		if os.IsPermission(err) || os.IsTimeout(err) || os.IsNotExist(err) {
 			return err
 		}
@@ -186,19 +179,24 @@ func (t *TransStream) createSegment() error {
 		startSeq++
 	}
 
-	t.context.segmentSeq = startSeq
-	t.context.file = tsFile
-	_ = t.muxer.WriteHeader()
+	t.ctx.segmentSeq = startSeq
+	t.ctx.file = tsFile
+	n, err := t.muxer.WriteHeader(t.ctx.writeBuffer)
+	if err != nil {
+		return err
+	}
+
+	t.ctx.writeBufferSize = n
 	return nil
 }
 
 func (t *TransStream) Close() ([][]byte, int64, error) {
 	var err error
 
-	if t.context.file != nil {
+	if t.ctx.file != nil {
 		err = t.flushSegment(true)
-		err = t.context.file.Close()
-		t.context.file = nil
+		err = t.ctx.file.Close()
+		t.ctx.file = nil
 	}
 
 	if t.muxer != nil {
@@ -244,7 +242,7 @@ func DeleteOldSegments(id string) {
 // @Params parentDir	保存切片的绝对路径. mu38和ts切片放在同一目录下, 目录地址使用parentDir+urlPrefix
 // @Params segmentDuration 单个切片时长
 // @Params playlistLength 缓存多少个切片
-func NewTransStream(dir, m3u8Name, tsFormat, tsUrl string, segmentDuration, playlistLength int, seq int, playlistFormat *string, writer libhls.M3U8Writer) (stream.TransStream, error) {
+func NewTransStream(dir, m3u8Name, tsFormat, tsUrl string, segmentDuration, playlistLength int, seq int, playlistFormat *string, writer stream.M3U8Writer) (stream.TransStream, error) {
 	// 创建文件夹
 	m3u8Path := fmt.Sprintf("%s/%s", dir, m3u8Name)
 	if err := os.MkdirAll(filepath.Dir(m3u8Path), 0666); err != nil {
@@ -271,7 +269,7 @@ func NewTransStream(dir, m3u8Name, tsFormat, tsUrl string, segmentDuration, play
 	if writer != nil {
 		transStream.M3U8Writer = writer
 	} else {
-		transStream.M3U8Writer = libhls.NewM3U8Writer(playlistLength)
+		transStream.M3U8Writer = stream.NewM3U8Writer(playlistLength)
 	}
 
 	if playlistFormat != nil {
@@ -281,16 +279,11 @@ func NewTransStream(dir, m3u8Name, tsFormat, tsUrl string, segmentDuration, play
 	}
 
 	// 创建TS封装器
-	muxer := libmpeg.NewTSMuxer()
-	muxer.SetWriteHandler(transStream.onTSWrite)
-	muxer.SetAllocHandler(transStream.onTSAlloc)
+	muxer := mpeg.NewTSMuxer()
 
-	// ts封装上下文对象
-	transStream.context = &tsContext{
-		segmentSeq:      seq,
-		writeBuffer:     make([]byte, 1024*1024),
-		writeBufferSize: 0,
-	}
+	// 初始化ts封装上下文对象
+	transStream.ctx.segmentSeq = seq
+	transStream.ctx.writeBuffer = make([]byte, 1024*1024)
 
 	transStream.muxer = muxer
 	transStream.m3u8File = file
@@ -301,7 +294,7 @@ func NewTransStream(dir, m3u8Name, tsFormat, tsUrl string, segmentDuration, play
 func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
 	id := source.GetID()
 
-	var writer libhls.M3U8Writer
+	var writer stream.M3U8Writer
 	var playlistFormat *string
 	startSeq := -1
 

jt1078/jt_demuxer.go

@@ -0,0 +1,97 @@
+package jt1078
+
+import (
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/utils"
+)
+
+type Demuxer struct {
+	avformat.BaseDemuxer
+	prevPacket *Packet
+	sim        string
+	channel    int
+	lastError  string
+}
+
+func (d *Demuxer) ProcessPrevPacket() error {
+	var codec utils.AVCodecID
+	index := d.FindBufferIndex(int(d.prevPacket.pt))
+	bytes, err := d.BaseDemuxer.DataPipeline.Feat(index)
+	if err != nil {
+		return err
+	} else /*if d.prevPacket.packetType > AudioFrameMark {
+		// 透传数据, 丢弃
+		d.DataPipeline.DiscardBackPacket(index)
+		return nil
+	} else*/if err, codec = PT2CodecID(d.prevPacket.pt); err != nil {
+		d.BaseDemuxer.DataPipeline.DiscardBackPacket(index)
+		return err
+	}
+
+	if d.prevPacket.packetType == AudioFrameMark {
+		d.OnAudioPacket(index, codec, bytes, int64(d.prevPacket.ts))
+	} else if d.prevPacket.packetType < AudioFrameMark {
+		// 视频帧
+		d.OnVideoPacket(index, codec, bytes, avformat.IsKeyFrame(codec, bytes), int64(d.prevPacket.ts), int64(d.prevPacket.ts), avformat.PacketTypeAnnexB)
+	}
+
+	if !d.Completed && d.Tracks.Size() > 1 {
+		d.ProbeComplete()
+	}
+	return nil
+}
+
+func (d *Demuxer) Input(data []byte) (int, error) {
+	packet := Packet{}
+	if err := packet.Unmarshal(data); err != nil {
+		return 0, err
+	} else if len(packet.payload) == 0 {
+		// 过滤空数据
+		return 0, nil
+	}
+
+	// 如果时间戳或者负载类型发生变化, 认为是新的音视频帧，处理前一包，创建AVPacket，回调给PublishSource。
+	// 分包标记可能不靠谱
+	if d.prevPacket != nil && (d.prevPacket.ts != packet.ts || d.prevPacket.pt != packet.pt) {
+		err := d.ProcessPrevPacket()
+		if err != nil && err.Error() != d.lastError {
+			println(err.Error())
+			d.lastError = err.Error()
+		}
+	}
+
+	if d.prevPacket == nil {
+		d.prevPacket = &Packet{}
+		d.sim = packet.simNumber
+		d.channel = int(packet.channelNumber)
+	}
+
+	var mediaType utils.AVMediaType
+	if packet.packetType < AudioFrameMark {
+		mediaType = utils.AVMediaTypeVideo
+	} else if packet.packetType == AudioFrameMark {
+		mediaType = utils.AVMediaTypeAudio
+	} else {
+		// 透传数据, 丢弃
+		return len(data), nil
+	}
+
+	index := d.FindBufferIndex(int(packet.pt))
+	_, err := d.DataPipeline.Write(packet.payload, index, mediaType)
+	if err != nil {
+		panic(err)
+	}
+
+	*d.prevPacket = packet
+	return len(data), nil
+}
+
+func NewDemuxer() *Demuxer {
+	return &Demuxer{
+		BaseDemuxer: avformat.BaseDemuxer{
+			DataPipeline: &avformat.StreamsBuffer{},
+			Name:         "jt1078", // vob
+			AutoFree:     false,
+		},
+	}
+}

jt1078/jt_packet.go

@@ -0,0 +1,120 @@
+package jt1078
+
+import (
+	"encoding/binary"
+	"fmt"
+	"github.com/lkmio/avformat/utils"
+)
+
+const (
+	VideoIFrameMark      = 0b000
+	VideoPFrameMark      = 0b001
+	VideoBFrameMark      = 0b010
+	AudioFrameMark       = 0b011
+	TransmissionDataMark = 0b100
+
+	PTVideoH264 = 98
+	PTVideoH265 = 99
+	PTVideoAVS  = 100
+	PTVideoSVAC = 101
+
+	PTAudioG711A  = 6
+	PTAudioG711U  = 7
+	PTAudioG726   = 8
+	PTAudioG729A  = 9
+	PTAudioAAC    = 19
+	PTAudioMP3    = 25
+	PTAudioADPCMA = 26
+)
+
+type Packet struct {
+	pt            byte
+	packetType    byte
+	ts            uint64
+	subMark       byte
+	simNumber     string
+	channelNumber byte
+	payload       []byte
+}
+
+func (p *Packet) Unmarshal(data []byte) error {
+	if len(data) < 12 {
+		return fmt.Errorf("invaild data")
+	}
+
+	packetType := data[11] >> 4 & 0x0F
+	// 忽略透传数据
+	if TransmissionDataMark == packetType {
+		return fmt.Errorf("invaild data")
+	}
+
+	// 忽略低于最低长度的数据包
+	if (AudioFrameMark == packetType && len(data) < 26) || (AudioFrameMark == packetType && len(data) < 22) {
+		return fmt.Errorf("invaild data")
+	}
+
+	// x扩展位,固定为0
+	_ = data[0] >> 4 & 0x1
+	pt := data[1] & 0x7F
+	// seq
+	_ = binary.BigEndian.Uint16(data[2:])
+
+	var simNumber string
+	for i := 4; i < 10; i++ {
+		simNumber += fmt.Sprintf("%02d", data[i])
+	}
+
+	// channel
+	channelNumber := data[10]
+	// subMark
+	subMark := data[11] & 0x0F
+	// 时间戳,单位ms
+	var ts uint64
+	n := 12
+	if TransmissionDataMark != packetType {
+		ts = binary.BigEndian.Uint64(data[n:])
+		n += 8
+	}
+
+	if AudioFrameMark > packetType {
+		// iFrameInterval
+		_ = binary.BigEndian.Uint16(data[n:])
+		n += 2
+		// lastFrameInterval
+		_ = binary.BigEndian.Uint16(data[n:])
+		n += 2
+	}
+
+	// size
+	_ = binary.BigEndian.Uint16(data[n:])
+	n += 2
+
+	p.pt = pt
+	p.packetType = packetType
+	p.ts = ts
+	p.simNumber = simNumber
+	p.channelNumber = channelNumber
+	p.subMark = subMark
+	p.payload = data[n:]
+
+	return nil
+}
+
+func PT2CodecID(pt byte) (error, utils.AVCodecID) {
+	switch pt {
+	case PTVideoH264:
+		return nil, utils.AVCodecIdH264
+	case PTVideoH265:
+		return nil, utils.AVCodecIdH265
+	case PTAudioG711A:
+		return nil, utils.AVCodecIdPCMALAW
+	case PTAudioG711U:
+		return nil, utils.AVCodecIdPCMMULAW
+	case PTAudioAAC:
+		return nil, utils.AVCodecIdAAC
+	//case PTAudioADPCMA:
+	//	return nil, utils.AVCodecIdADPCMAFC
+	default:
+		return fmt.Errorf("the codec %d is not implemented", pt), utils.AVCodecIdNONE
+	}
+}

jt1078/jt_server.go

@@ -1,9 +1,9 @@
 package jt1078
 
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 	"runtime"
 )

jt1078/jt_session.go

@@ -1,86 +1,46 @@
 package jt1078
 
 import (
-	"encoding/binary"
-	"fmt"
-	"github.com/lkmio/avformat/collections"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/transport"
 	"net"
 	"strconv"
 )
 
-const (
-	VideoIFrameMark      = 0b000
-	VideoPFrameMark      = 0b001
-	VideoBFrameMark      = 0b010
-	AudioFrameMark       = 0b011
-	TransmissionDataMark = 0b100
-
-	PTVideoH264 = 98
-	PTVideoH265 = 99
-	PTVideoAVS  = 100
-	PTVideoSVAC = 101
-
-	PTAudioG711A  = 6
-	PTAudioG711U  = 7
-	PTAudioG726   = 8
-	PTAudioG729A  = 9
-	PTAudioAAC    = 19
-	PTAudioMP3    = 25
-	PTAudioADPCMA = 26
-)
-
 type Session struct {
 	stream.PublishSource
-
-	phone   string
 	decoder       *transport.DelimiterFrameDecoder
-
-	audioIndex    int
-	videoIndex    int
-	audioStream   utils.AVStream
-	videoStream   utils.AVStream
-	audioBuffer   collections.MemoryPool
-	videoBuffer   collections.MemoryPool
-	rtpPacket     *RtpPacket
 	receiveBuffer *stream.ReceiveBuffer
 }
 
-type RtpPacket struct {
-	pt            byte
-	packetType    byte
-	ts            uint64
-	subMark       byte
-	simNumber     string
-	channelNumber byte
-
-	payload []byte
-}
-
-func (s *Session) OnJtPTPPacket(data []byte) {
-	packet, err := read1078RTPPacket(data)
+func (s *Session) Input(data []byte) error {
+	var n int
+	for length := len(data); n < length; {
+		i, bytes, err := s.decoder.Input(data[n:])
 		if err != nil {
-		return
+			return err
+		} else if len(bytes) < 1 {
+			break
 		}
 
-	// 过滤空数据
-	if len(packet.payload) == 0 {
-		return
+		n += i
+		demuxer := s.TransDemuxer.(*Demuxer)
+		firstOfPacket := demuxer.prevPacket == nil
+		_, err = demuxer.Input(bytes)
+		if err != nil {
+			return err
 		}
 
 		// 首包处理, hook通知
-	if s.rtpPacket == nil {
-		s.SetID(packet.simNumber + "/" + strconv.Itoa(int(packet.channelNumber)))
-		s.rtpPacket = &RtpPacket{}
-		*s.rtpPacket = packet
+		if firstOfPacket && demuxer.prevPacket != nil {
+			s.SetID(demuxer.sim + "/" + strconv.Itoa(demuxer.channel))
 
 			go func() {
 				_, state := stream.PreparePublishSource(s, true)
 				if utils.HookStateOK != state {
-				log.Sugar.Errorf("1078推流失败 source: %s", s.phone)
+					log.Sugar.Errorf("1078推流失败 source: %s", demuxer.sim)
 
 					if s.Conn != nil {
 						s.Conn.Close()
@@ -88,59 +48,9 @@ func (s *Session) OnJtPTPPacket(data []byte) {
 				}
 			}()
 		}
-
-	// 如果时间戳或者负载类型发生变化, 认为是新的音视频帧，处理前一包，创建AVPacket，回调给PublishSource。
-	// 分包标记可能不靠谱
-	if s.rtpPacket.ts != packet.ts || s.rtpPacket.pt != packet.pt {
-		if s.rtpPacket.packetType == AudioFrameMark && s.audioBuffer != nil {
-			if err := s.processAudioPacket(s.rtpPacket.pt, s.rtpPacket.packetType, s.rtpPacket.ts, s.audioBuffer.Fetch(), s.audioIndex); err != nil {
-				log.Sugar.Errorf("处理音频包失败 phone:%s err:%s", s.phone, err.Error())
-				s.audioBuffer.FreeTail()
 	}
 
-			*s.rtpPacket = packet
-		} else if s.rtpPacket.packetType < AudioFrameMark && s.videoBuffer != nil {
-			if err := s.processVideoPacket(s.rtpPacket.pt, s.rtpPacket.packetType, s.rtpPacket.ts, s.videoBuffer.Fetch(), s.videoIndex); err != nil {
-				log.Sugar.Errorf("处理视频包失败 phone:%s err:%s", s.phone, err.Error())
-				s.videoBuffer.FreeTail()
-			}
-
-			*s.rtpPacket = packet
-		}
-	}
-
-	// 部分音视频帧
-	if packet.packetType == AudioFrameMark {
-		if s.audioBuffer == nil {
-			if s.videoIndex == 0 && s.audioIndex == 0 {
-				s.videoIndex = 1
-			}
-
-			// 创建音频的AVPacket缓冲区
-			s.audioBuffer = s.FindOrCreatePacketBuffer(s.audioIndex, utils.AVMediaTypeAudio)
-		}
-
-		// 将部分音频帧写入缓冲区
-		s.audioBuffer.TryMark()
-		s.audioBuffer.Write(packet.payload)
-	} else {
-		if s.videoBuffer == nil {
-			if s.videoIndex == 0 && s.audioIndex == 0 {
-				s.audioIndex = 1
-			}
-
-			// 创建视频的AVPacket缓冲区
-			s.videoBuffer = s.FindOrCreatePacketBuffer(s.videoIndex, utils.AVMediaTypeVideo)
-		}
-
-		// 将部分视频帧写入缓冲区
-		s.videoBuffer.TryMark()
-		s.videoBuffer.Write(packet.payload)
-	}
-}
-
-func (s *Session) Input(data []byte) error {
-	return s.decoder.Input(data)
+	return nil
 }
 
 func (s *Session) Close() {
@@ -151,149 +61,23 @@ func (s *Session) Close() {
 		s.Conn = nil
 	}
 
-	if s.decoder != nil {
-		s.decoder.Close()
-		s.decoder = nil
-	}
-
 	s.PublishSource.Close()
 }
 
-// 从视频帧中提取AVPacket和AVStream, 回调给PublishSource
-func (s *Session) processVideoPacket(pt byte, pktType byte, ts uint64, data []byte, index int) error {
-	var codecId utils.AVCodecID
-
-	if PTVideoH264 == pt {
-		if s.videoStream == nil && VideoIFrameMark != pktType {
-			log.Sugar.Errorf("skip non keyframes conn:%s", s.Conn.RemoteAddr())
-			return nil
-		}
-		codecId = utils.AVCodecIdH264
-	} else if PTVideoH265 == pt {
-		if s.videoStream == nil && VideoIFrameMark != pktType {
-			log.Sugar.Errorf("skip non keyframes conn:%s", s.Conn.RemoteAddr())
-			return nil
-		}
-		codecId = utils.AVCodecIdH265
-	} else {
-		return fmt.Errorf("the codec %d is not implemented", pt)
-	}
-
-	videoStream, videoPacket, err := stream.ExtractVideoPacket(codecId, VideoIFrameMark == pktType, s.videoStream == nil, data, int64(ts), int64(ts), index, 1000)
-	if err != nil {
-		return err
-	}
-
-	if videoStream != nil {
-		s.videoStream = videoStream
-		s.OnDeMuxStream(videoStream)
-		if s.videoStream != nil && s.audioStream != nil {
-			s.OnDeMuxStreamDone()
-		}
-	}
-
-	s.OnDeMuxPacket(videoPacket)
-	return nil
-}
-
-// 从音频帧中提取AVPacket和AVStream, 回调给PublishSource
-func (s *Session) processAudioPacket(pt byte, pktType byte, ts uint64, data []byte, index int) error {
-	var codecId utils.AVCodecID
-
-	if PTAudioG711A == pt {
-		codecId = utils.AVCodecIdPCMALAW
-	} else if PTAudioG711U == pt {
-		codecId = utils.AVCodecIdPCMMULAW
-	} else if PTAudioAAC == pt {
-		codecId = utils.AVCodecIdAAC
-	} else {
-		return fmt.Errorf("the codec %d is not implemented", pt)
-	}
-
-	audioStream, audioPacket, err := stream.ExtractAudioPacket(codecId, s.audioStream == nil, data, int64(ts), int64(ts), index, 1000)
-	if err != nil {
-		return err
-	}
-
-	if audioStream != nil {
-		s.audioStream = audioStream
-		s.OnDeMuxStream(audioStream)
-		if s.videoStream != nil && s.audioStream != nil {
-			s.OnDeMuxStreamDone()
-		}
-	}
-
-	s.OnDeMuxPacket(audioPacket)
-	return nil
-}
-
-// 读取1078的rtp包, 返回数据类型, 负载类型、时间戳、负载数据
-func read1078RTPPacket(data []byte) (RtpPacket, error) {
-	if len(data) < 12 {
-		return RtpPacket{}, fmt.Errorf("invaild data")
-	}
-
-	packetType := data[11] >> 4 & 0x0F
-	// 忽略透传数据
-	if TransmissionDataMark == packetType {
-		return RtpPacket{}, fmt.Errorf("invaild data")
-	}
-
-	// 忽略低于最低长度的数据包
-	if (AudioFrameMark == packetType && len(data) < 26) || (AudioFrameMark == packetType && len(data) < 22) {
-		return RtpPacket{}, fmt.Errorf("invaild data")
-	}
-
-	// x扩展位,固定为0
-	_ = data[0] >> 4 & 0x1
-	pt := data[1] & 0x7F
-	// seq
-	_ = binary.BigEndian.Uint16(data[2:])
-
-	var simNumber string
-	for i := 4; i < 10; i++ {
-		simNumber += fmt.Sprintf("%02d", data[i])
-	}
-
-	// channel
-	channelNumber := data[10]
-	// subMark
-	subMark := data[11] & 0x0F
-	// 时间戳,单位ms
-	var ts uint64
-	n := 12
-	if TransmissionDataMark != packetType {
-		ts = binary.BigEndian.Uint64(data[n:])
-		n += 8
-	}
-
-	if AudioFrameMark > packetType {
-		// iFrameInterval
-		_ = binary.BigEndian.Uint16(data[n:])
-		n += 2
-		// lastFrameInterval
-		_ = binary.BigEndian.Uint16(data[n:])
-		n += 2
-	}
-
-	// size
-	_ = binary.BigEndian.Uint16(data[n:])
-	n += 2
-
-	return RtpPacket{pt: pt, packetType: packetType, ts: ts, simNumber: simNumber, channelNumber: channelNumber, subMark: subMark, payload: data[n:]}, nil
-}
-
 func NewSession(conn net.Conn) *Session {
+	delimiter := [4]byte{0x30, 0x31, 0x63, 0x64}
 	session := Session{
 		PublishSource: stream.PublishSource{
 			Conn:         conn,
 			Type:         stream.SourceType1078,
+			TransDemuxer: NewDemuxer(),
 		},
-	}
-	delimiter := [4]byte{0x30, 0x31, 0x63, 0x64}
-	session.decoder = transport.NewDelimiterFrameDecoder(1024*1024*2, delimiter[:], session.OnJtPTPPacket)
-	session.receiveBuffer = stream.NewTCPReceiveBuffer()
 
+		decoder:       transport.NewDelimiterFrameDecoder(1024*1024*2, delimiter[:]),
+		receiveBuffer: stream.NewTCPReceiveBuffer(),
+	}
+
+	session.TransDemuxer.SetHandler(&session)
 	session.Init(stream.ReceiveBufferTCPBlockCount)
 	go stream.LoopEvent(&session)
 	return &session

jt1078/jt_test.go

@@ -1,8 +1,10 @@
 package jt1078
 
 import (
-	"github.com/lkmio/avformat/libbufio"
-	"github.com/lkmio/avformat/transport"
+	"encoding/binary"
+	"fmt"
+	"github.com/lkmio/avformat/bufio"
+	"github.com/lkmio/transport"
 	"net"
 	"os"
 	"testing"
@@ -10,7 +12,52 @@ import (
 )
 
 func TestPublish(t *testing.T) {
-	path := "D:\\GOProjects\\avformat\\10352264314-2.bin"
+
+	t.Run("decode_1078_data", func(t *testing.T) {
+		data, err := os.ReadFile("../dump/jt1078-127.0.0.1.50659")
+		if err != nil {
+			panic(err)
+		}
+
+		delimiter := [4]byte{0x30, 0x31, 0x63, 0x64}
+		decoder := transport.NewDelimiterFrameDecoder(1024*1024*2, delimiter[:])
+
+		length := len(data)
+		for j := 0; j < length; j += 4 {
+			size := int(binary.BigEndian.Uint32(data[j:]))
+			if 4+size > length-j {
+				break
+			}
+
+			rtp := data[j+4 : j+4+size]
+			var n int
+			for length := len(rtp); n < length; {
+				i, bytes, err := decoder.Input(rtp[n:])
+				if err != nil {
+					panic(err)
+				} else if len(bytes) < 1 {
+					break
+				}
+
+				n += i
+				packet := Packet{}
+				err = packet.Unmarshal(bytes)
+				if err != nil {
+					panic(err)
+				}
+
+				fmt.Printf("1078 packet ts: %d\r\n", packet.ts)
+			}
+
+			j += size
+		}
+
+	})
+
+	t.Run("publish", func(t *testing.T) {
+
+		//path := "../../source_files/10352264314-2.bin"
+		path := "../../source_files/013800138000-1.bin"
 
 		client := transport.TCPClient{}
 		addr, err := net.ResolveTCPAddr("tcp", "127.0.0.1:1078")
@@ -29,11 +76,10 @@ func TestPublish(t *testing.T) {
 
 		index := 0
 		for index < len(file) {
-		n := libbufio.MinInt(len(file)-index, 1500)
+			n := bufio.MinInt(len(file)-index, 1500)
 			client.Write(file[index : index+n])
 			index += n
 			time.Sleep(1 * time.Millisecond)
 		}
-
-	println("end")
+	})
 }

main.go

@@ -2,16 +2,18 @@ package main
 
 import (
 	"encoding/json"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/flv"
-	"github.com/lkmio/lkm/gb28181"
 	"github.com/lkmio/lkm/hls"
 	"github.com/lkmio/lkm/jt1078"
-	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/record"
+	"github.com/lkmio/lkm/rtsp"
+	"github.com/lkmio/transport"
+	"os"
+
+	"github.com/lkmio/lkm/gb28181"
+	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/rtc"
 	"github.com/lkmio/lkm/rtmp"
-	"github.com/lkmio/lkm/rtsp"
 	"github.com/lkmio/lkm/stream"
 	"go.uber.org/zap/zapcore"
 	"net"
@@ -67,11 +69,18 @@ func init() {
 	log.InitLogger(config.Log.FileLogging, zapcore.Level(stream.AppConfig.Log.Level), stream.AppConfig.Log.Name, stream.AppConfig.Log.MaxSize, stream.AppConfig.Log.MaxBackup, stream.AppConfig.Log.MaxAge, stream.AppConfig.Log.Compress)
 
 	if stream.AppConfig.GB28181.Enable && stream.AppConfig.GB28181.IsMultiPort() {
-		gb28181.TransportManger = transport.NewTransportManager(uint16(stream.AppConfig.GB28181.Port[0]), uint16(stream.AppConfig.GB28181.Port[1]))
+		gb28181.TransportManger = transport.NewTransportManager(config.ListenIP, uint16(stream.AppConfig.GB28181.Port[0]), uint16(stream.AppConfig.GB28181.Port[1]))
 	}
 
 	if stream.AppConfig.Rtsp.Enable && stream.AppConfig.Rtsp.IsMultiPort() {
-		rtsp.TransportManger = transport.NewTransportManager(uint16(stream.AppConfig.Rtsp.Port[1]), uint16(stream.AppConfig.Rtsp.Port[2]))
+		rtsp.TransportManger = transport.NewTransportManager(config.ListenIP, uint16(stream.AppConfig.Rtsp.Port[1]), uint16(stream.AppConfig.Rtsp.Port[2]))
+	}
+
+	// 创建dump目录
+	if stream.AppConfig.Debug {
+		if err := os.MkdirAll("dump", 0666); err != nil {
+			panic(err)
+		}
 	}
 
 	// 打印配置信息

record/record_flv.go

@@ -14,12 +14,13 @@ type FLVFileSink struct {
 }
 
 // Input 输入http-flv数据
-func (f *FLVFileSink) Input(data []byte) error {
+func (f *FLVFileSink) Write(index int, blocks [][]byte, ts int64) error {
 	if f.fail {
 		return nil
 	}
 
-	//去掉不需要的换行符
+	for _, data := range blocks {
+		// 去掉不需要的换行符
 		var offset int
 		for i := 2; i < len(data); i++ {
 			if data[i-2] == 0x0D && data[i-1] == 0x0A {
@@ -30,11 +31,13 @@ func (f *FLVFileSink) Input(data []byte) error {
 
 		_, err := f.file.Write(data[offset : len(data)-2])
 		if err != nil {
-		//只要写入失败一次，后续不再允许写入, 不影响拉流
+			// 只要写入失败一次，后续不再允许写入, 不影响拉流
 			f.fail = true
+			return err
+		}
 	}
 
-	return err
+	return nil
 }
 
 func (f *FLVFileSink) Close() {
@@ -50,20 +53,20 @@ func NewFLVFileSink(sourceId string) (stream.Sink, string, error) {
 	now := time.Now().Format("2006-01-02/15-04-05")
 	path := filepath.Join(stream.AppConfig.Record.Dir, sourceId, now+".flv")
 
-	//创建目录
+	// 创建目录
 	dir := filepath.Dir(path)
 	if err := os.MkdirAll(dir, 0666); err != nil {
 		return nil, "", err
 	}
 
-	//创建flv文件
+	// 创建flv文件
 	file, err := os.OpenFile(path, os.O_CREATE|os.O_RDWR, 0666)
 	if err != nil {
 		return nil, "", err
 	}
 
 	return &FLVFileSink{
-		BaseSink: stream.BaseSink{ID: "record-sink-flv", SourceID: sourceId, Protocol: stream.TransStreamFlv},
+		BaseSink: stream.BaseSink{ID: "record-sink-flv", SourceID: sourceId, Protocol: stream.TransStreamFlv, Ready: true},
 		file:     file,
 	}, path, nil
 }

rtc/rtc_sink.go

@@ -42,7 +42,7 @@ func (s *Sink) StartStreaming(transStream stream.TransStream) error {
 	for index, track := range tracks {
 		var mimeType string
 		var id string
-		codecId := track.Stream.CodecId()
+		codecId := track.Stream.CodecID
 		if utils.AVCodecIdH264 == codecId {
 			mimeType = webrtc.MimeTypeH264
 		} else if utils.AVCodecIdH265 == codecId {
@@ -64,7 +64,7 @@ func (s *Sink) StartStreaming(transStream stream.TransStream) error {
 			continue
 		}
 
-		if utils.AVMediaTypeAudio == track.Stream.Type() {
+		if utils.AVMediaTypeAudio == track.Stream.MediaType {
 			id = "audio"
 		} else {
 			id = "video"

rtc/rtc_stream.go

@@ -1,6 +1,7 @@
 package rtc
 
 import (
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/stream"
 	"github.com/pion/interceptor"
@@ -16,21 +17,23 @@ type transStream struct {
 	stream.BaseTransStream
 }
 
-func (t *transStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
+func (t *transStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	t.ClearOutStreamBuffer()
 
-	if utils.AVMediaTypeAudio == packet.MediaType() {
-		t.AppendOutStreamBuffer(packet.Data())
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
-		if packet.KeyFrame() {
-			extra := t.BaseTransStream.Tracks[packet.Index()].Stream.CodecParameters().AnnexBExtraData()
+	if utils.AVMediaTypeAudio == packet.MediaType {
+		t.AppendOutStreamBuffer(packet.Data)
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
+		avStream := t.BaseTransStream.Tracks[packet.Index].Stream
+		if packet.Key {
+			extra := avStream.CodecParameters.AnnexBExtraData()
 			t.AppendOutStreamBuffer(extra)
 		}
 
-		t.AppendOutStreamBuffer(packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()].Stream))
+		data := avformat.AVCCPacket2AnnexB(avStream, packet)
+		t.AppendOutStreamBuffer(data)
 	}
 
-	return t.OutBuffer[:t.OutBufferSize], int64(uint32(packet.Duration(1000))), utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame(), nil
+	return t.OutBuffer[:t.OutBufferSize], int64(uint32(packet.GetDuration(1000))), utils.AVMediaTypeVideo == packet.MediaType && packet.Key, nil
 }
 
 func (t *transStream) WriteHeader() error {

rtmp/publish_test.go

@@ -2,7 +2,7 @@ package rtmp
 
 import (
 	"encoding/binary"
-	"github.com/lkmio/avformat/transport"
+	"github.com/lkmio/transport"
 	"net"
 	"os"
 	"testing"
@@ -19,7 +19,7 @@ func TestName(t *testing.T) {
 	}
 
 	client := transport.TCPClient{}
-	if err := client.Connect(nil, addr); err != nil {
+	if _, err := client.Connect(nil, addr); err != nil {
 		panic(err)
 	}
 

rtmp/rtmp_publisher.go

@@ -1,50 +1,19 @@
 package rtmp
 
 import (
-	"github.com/lkmio/avformat/libflv"
-	"github.com/lkmio/avformat/librtmp"
-	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtmp"
 	"net"
 )
 
 // Publisher RTMP推流Source
 type Publisher struct {
 	stream.PublishSource
-
-	Stack *librtmp.Stack
+	Stack *rtmp.ServerStack
 }
 
 func (p *Publisher) Input(data []byte) error {
-	return p.Stack.Input(data)
-}
-
-func (p *Publisher) OnDeMuxStream(stream utils.AVStream) {
-	// AVStream的ExtraData已经拷贝, 释放掉内存池中最新分配的内存
-	p.FindOrCreatePacketBuffer(stream.Index(), stream.Type()).FreeTail()
-	p.PublishSource.OnDeMuxStream(stream)
-}
-
-// OnVideo 解析出来的完整视频包
-func (p *Publisher) OnVideo(index int, data []byte, ts uint32) {
-	data = p.FindOrCreatePacketBuffer(index, utils.AVMediaTypeVideo).Fetch()
-	// 交给flv解复用器, 解析出AVPacket
-	p.PublishSource.TransDeMuxer.(libflv.DeMuxer).InputVideo(data, ts)
-}
-
-func (p *Publisher) OnAudio(index int, data []byte, ts uint32) {
-	data = p.FindOrCreatePacketBuffer(index, utils.AVMediaTypeAudio).Fetch()
-	p.PublishSource.TransDeMuxer.(libflv.DeMuxer).InputAudio(data, ts)
-}
-
-// OnPartPacket AVPacket的部分数据包
-func (p *Publisher) OnPartPacket(index int, mediaType utils.AVMediaType, data []byte, first bool) {
-	buffer := p.FindOrCreatePacketBuffer(index, mediaType)
-	if first {
-		buffer.Mark()
-	}
-
-	buffer.Write(data)
+	return p.Stack.Input(p.Conn, data)
 }
 
 func (p *Publisher) Close() {
@@ -52,10 +21,11 @@ func (p *Publisher) Close() {
 	p.Stack = nil
 }
 
-func NewPublisher(source string, stack *librtmp.Stack, conn net.Conn) *Publisher {
-	deMuxer := libflv.NewDeMuxer()
-	publisher := &Publisher{PublishSource: stream.PublishSource{ID: source, Type: stream.SourceTypeRtmp, TransDeMuxer: deMuxer, Conn: conn}, Stack: stack}
+func NewPublisher(source string, stack *rtmp.ServerStack, conn net.Conn) *Publisher {
+	demuxer := stack.FLV
+	publisher := &Publisher{PublishSource: stream.PublishSource{ID: source, Type: stream.SourceTypeRtmp, TransDemuxer: demuxer, Conn: conn}, Stack: stack}
 	// 设置回调, 接受从DeMuxer解析出来的音视频包
-	deMuxer.SetHandler(publisher)
+	demuxer.SetHandler(publisher)
+	demuxer.AutoFree = false
 	return publisher
 }

rtmp/rtmp_server.go

@@ -6,8 +6,8 @@ import (
 	"net"
 	"runtime"
 
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/transport"
 )
 
 type Server interface {

rtmp/rtmp_server_test.go

@@ -1,16 +1,16 @@
 package rtmp
 
 import (
-	"github.com/lkmio/avformat/librtmp"
-	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtmp"
 	"net"
 	"testing"
 )
 
-func CreateTransStream(source stream.Source, protocol stream.TransStreamProtocol, streams []utils.AVStream) stream.TransStream {
+func CreateTransStream(source stream.Source, protocol stream.TransStreamProtocol, streams []*avformat.AVStream) stream.TransStream {
 	if stream.TransStreamRtmp == protocol {
-		return NewTransStream(librtmp.ChunkSize)
+		return NewTransStream(rtmp.MaxChunkSize, nil)
 	}
 
 	return nil

rtmp/rtmp_session.go

@@ -1,20 +1,20 @@
 package rtmp
 
 import (
-	"github.com/lkmio/avformat/librtmp"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtmp"
 	"net"
 )
 
 // Session RTMP会话, 解析处理Message
 type Session struct {
-	stack       *librtmp.Stack // rtmp协议栈, 解析message
-	handle      interface{}    // 持有具体会话句柄(推流端/拉流端)， 在@see OnPublish @see OnPlay回调中赋值
-	isPublisher bool           // 是否时推流会话
-
 	conn        net.Conn
+	stack       *rtmp.ServerStack // rtmp协议栈, 解析message
+	handle      interface{}       // 持有具体会话句柄(推流端/拉流端)， 在@see OnPublish @see OnPlay回调中赋值
+	isPublisher bool              // 是否是推流会话
+
 	receiveBuffer *stream.ReceiveBuffer // 推流源收流队列
 }
 
@@ -29,14 +29,12 @@ func (s *Session) generateSourceID(app, stream string) string {
 }
 
 func (s *Session) OnPublish(app, stream_ string) utils.HookState {
-	log.Sugar.Infof("rtmp onpublish app:%s stream:%s conn:%s", app, stream_, s.conn.RemoteAddr().String())
+	log.Sugar.Infof("rtmp onpublish app: %s stream: %s conn: %s", app, stream_, s.conn.RemoteAddr().String())
 
 	streamName, values := stream.ParseUrl(stream_)
 
 	sourceId := s.generateSourceID(app, streamName)
 	source := NewPublisher(sourceId, s.stack, s.conn)
-	// 设置推流的音视频回调
-	s.stack.SetOnPublishHandler(source)
 
 	// 初始化放在add source前面, 以防add后再init, 空窗期拉流队列空指针.
 	source.Init(stream.ReceiveBufferTCPBlockCount)
@@ -45,7 +43,7 @@ func (s *Session) OnPublish(app, stream_ string) utils.HookState {
 	// 统一处理source推流事件, source是否已经存在, hook回调....
 	_, state := stream.PreparePublishSource(source, true)
 	if utils.HookStateOK != state {
-		log.Sugar.Errorf("rtmp推流失败 source:%s", sourceId)
+		log.Sugar.Errorf("rtmp推流失败 source: %s", sourceId)
 	} else {
 		s.handle = source
 		s.isPublisher = true
@@ -81,12 +79,12 @@ func (s *Session) Input(data []byte) error {
 	if s.isPublisher {
 		return s.handle.(*Publisher).PublishSource.Input(data)
 	} else {
-		return s.stack.Input(data)
+		return s.stack.Input(s.conn, data)
 	}
 }
 
 func (s *Session) Close() {
-	// session/conn/stack相互引用, go释放不了...手动赋值为nil
+	// session/conn/stack相互引用, gc回收不了...手动赋值为nil
 	s.conn = nil
 
 	defer func() {
@@ -120,8 +118,9 @@ func (s *Session) Close() {
 func NewSession(conn net.Conn) *Session {
 	session := &Session{}
 
-	stack := librtmp.NewStack(conn, session)
-	session.stack = stack
+	stackServer := rtmp.NewStackServer(false)
+	stackServer.SetOnStreamHandler(session)
+	session.stack = stackServer
 	session.conn = conn
 	return session
 }

rtmp/rtmp_sink.go

@@ -1,23 +1,23 @@
 package rtmp
 
 import (
-	"github.com/lkmio/avformat/librtmp"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtmp"
 	"net"
 )
 
 type Sink struct {
 	stream.BaseSink
-	stack *librtmp.Stack
+	stack *rtmp.ServerStack
 }
 
 func (s *Sink) StartStreaming(_ stream.TransStream) error {
-	return s.stack.SendStreamBeginChunk()
+	return s.stack.SendStreamBeginChunk(s.Conn)
 }
 
 func (s *Sink) StopStreaming(stream stream.TransStream) {
-	_ = s.stack.SendStreamEOFChunk()
+	_ = s.stack.SendStreamEOFChunk(s.Conn)
 	s.BaseSink.StopStreaming(stream)
 }
 
@@ -26,7 +26,7 @@ func (s *Sink) Close() {
 	s.stack = nil
 }
 
-func NewSink(id stream.SinkID, sourceId string, conn net.Conn, stack *librtmp.Stack) stream.Sink {
+func NewSink(id stream.SinkID, sourceId string, conn net.Conn, stack *rtmp.ServerStack) stream.Sink {
 	return &Sink{
 		BaseSink: stream.BaseSink{ID: id, SourceID: sourceId, State: stream.SessionStateCreated, Protocol: stream.TransStreamRtmp, Conn: conn, DesiredAudioCodecId_: utils.AVCodecIdNONE, DesiredVideoCodecId_: utils.AVCodecIdNONE, TCPStreaming: true},
 		stack:    stack,

rtmp/rtmp_stream.go

@@ -1,59 +1,63 @@
 package rtmp
 
 import (
-	"github.com/lkmio/avformat/libflv"
-	"github.com/lkmio/avformat/librtmp"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/flv"
+	"github.com/lkmio/flv/amf0"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtmp"
 )
 
 type transStream struct {
 	stream.TCPTransStream
+	sequenceHeader []byte // sequence sequenceHeader
 
+	muxer      *flv.Muxer
+	audioChunk rtmp.Chunk
+	videoChunk rtmp.Chunk
 	chunkSize  int
-
-	header     []byte //sequence header
-	headerSize int
-
-	muxer      libflv.Muxer
-	audioChunk librtmp.Chunk
-	videoChunk librtmp.Chunk
-	metaData   *libflv.AMF0Object // 推流方携带的元数据
+	metaData   *amf0.Object // 推流方携带的元数据
 }
 
-func (t *transStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
+func (t *transStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	t.ClearOutStreamBuffer()
 
 	var data []byte
-	var chunk *librtmp.Chunk
+	var chunk *rtmp.Chunk
 	var videoPkt bool
 	var videoKey bool
-	// rtmp chunk消息体的数据大小
+	// chunk负载消息体的数据大小
 	var payloadSize int
-	// 先向rtmp buffer写的flv头,再按照chunk size分割,所以第一个chunk要跳过flv头大小
-	var chunkPayloadOffset int
+	// flv data header
+	var dataHeaderSize int
 	var dts int64
 	var pts int64
 	var keyBuffer bool
+	var frameType int
 
 	dts = packet.ConvertDts(1000)
 	pts = packet.ConvertPts(1000)
 	ct := pts - dts
 
-	if utils.AVMediaTypeAudio == packet.MediaType() {
-		data = packet.Data()
+	// chunk = header+payload(audio data / video data)
+	if utils.AVMediaTypeAudio == packet.MediaType {
+		data = packet.Data
 		chunk = &t.audioChunk
-		chunkPayloadOffset = 2
-		payloadSize += chunkPayloadOffset + len(data)
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
+		dataHeaderSize = t.muxer.ComputeAudioDataHeaderSize()
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
 		videoPkt = true
-		videoKey = packet.KeyFrame()
-		data = packet.AVCCPacketData()
-		chunk = &t.videoChunk
-		chunkPayloadOffset = t.muxer.ComputeVideoDataSize(uint32(ct))
-		payloadSize += chunkPayloadOffset + len(data)
+		if videoKey = packet.Key; videoKey {
+			frameType = flv.FrameTypeKeyFrame
 		}
 
+		data = avformat.AnnexBPacket2AVCC(packet)
+		chunk = &t.videoChunk
+		dataHeaderSize = t.muxer.ComputeVideoDataHeaderSize(uint32(ct))
+	}
+
+	payloadSize += dataHeaderSize + len(data)
+
 	// 遇到视频关键帧, 发送剩余的流, 创建新切片
 	if videoKey {
 		if segment, key := t.MWBuffer.FlushSegment(); len(segment) > 0 {
@@ -66,30 +70,31 @@ func (t *transStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error
 	chunkHeaderSize := 12
 	// type为3的chunk数量
 	numChunks := (payloadSize - 1) / t.chunkSize
-	rtmpMsgSize := chunkHeaderSize + payloadSize + numChunks
+	// 整个chunk大小
+	totalSize := chunkHeaderSize + payloadSize + numChunks
 	// 如果时间戳超过3字节, 每个chunk都需要多4字节的扩展时间戳
-	if dts >= 0xFFFFFF && dts <= 0xFFFFFFFF {
-		rtmpMsgSize += (1 + numChunks) * 4
+	if dts >= 0xFFFFFF && dts >= 0xFFFFFFFF {
+		totalSize += (1 + numChunks) * 4
 	}
 
 	// 分配指定大小的内存
-	allocate := t.MWBuffer.Allocate(rtmpMsgSize, dts, videoKey)
+	bytes := t.MWBuffer.Allocate(totalSize, dts, videoKey)
 
-	// 写第一个type为0的chunk header
+	// 写第一个type为0的chunk sequenceHeader
 	chunk.Length = payloadSize
 	chunk.Timestamp = uint32(dts)
-	n := chunk.MarshalHeader(allocate)
+	n := chunk.MarshalHeader(bytes)
 
 	// 封装成flv
 	if videoPkt {
-		n += t.muxer.WriteVideoData(allocate[n:], uint32(ct), packet.KeyFrame(), false)
+		n += t.muxer.VideoData.Marshal(bytes[n:], uint32(ct), frameType, false)
 	} else {
-		n += t.muxer.WriteAudioData(allocate[n:], false)
+		n += t.muxer.AudioData.Marshal(bytes[n:], false)
 	}
 
 	// 将flv data写入chunk body
-	n += chunk.WriteBody(allocate[n:], data, t.chunkSize, chunkPayloadOffset)
-	utils.Assert(len(allocate) == n)
+	n += chunk.WriteBody(bytes[n:], data, t.chunkSize, dataHeaderSize)
+	utils.Assert(len(bytes) == n)
 
 	// 合并写满了再发
 	if segment, key := t.MWBuffer.PeekCompletedSegment(); len(segment) > 0 {
@@ -101,10 +106,10 @@ func (t *transStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error
 }
 
 func (t *transStream) ReadExtraData(_ int64) ([][]byte, int64, error) {
-	utils.Assert(t.headerSize > 0)
+	utils.Assert(len(t.sequenceHeader) > 0)
 
-	// 发送sequence header
-	return [][]byte{t.header[:t.headerSize]}, 0, nil
+	// 发送sequence sequenceHeader
+	return [][]byte{t.sequenceHeader}, 0, nil
 }
 
 func (t *transStream) ReadKeyFrameBuffer() ([][]byte, int64, error) {
@@ -122,20 +127,20 @@ func (t *transStream) WriteHeader() error {
 	utils.Assert(t.Tracks != nil)
 	utils.Assert(!t.BaseTransStream.Completed)
 
-	var audioStream utils.AVStream
-	var videoStream utils.AVStream
+	var audioStream *avformat.AVStream
+	var videoStream *avformat.AVStream
 	var audioCodecId utils.AVCodecID
 	var videoCodecId utils.AVCodecID
 
 	for _, track := range t.Tracks {
-		if utils.AVMediaTypeAudio == track.Stream.Type() {
+		if utils.AVMediaTypeAudio == track.Stream.MediaType {
 			audioStream = track.Stream
-			audioCodecId = audioStream.CodecId()
-			t.audioChunk = librtmp.NewAudioChunk()
-		} else if utils.AVMediaTypeVideo == track.Stream.Type() {
+			audioCodecId = audioStream.CodecID
+			t.audioChunk = rtmp.NewAudioChunk()
+		} else if utils.AVMediaTypeVideo == track.Stream.MediaType {
 			videoStream = track.Stream
-			videoCodecId = videoStream.CodecId()
-			t.videoChunk = librtmp.NewVideoChunk()
+			videoCodecId = videoStream.CodecID
+			t.videoChunk = rtmp.NewVideoChunk()
 		}
 	}
 
@@ -143,67 +148,67 @@ func (t *transStream) WriteHeader() error {
 
 	// 初始化
 	t.BaseTransStream.Completed = true
-	t.muxer = libflv.NewMuxer(t.metaData)
+	t.muxer = flv.NewMuxer(t.metaData)
 	if utils.AVCodecIdNONE != audioCodecId {
-		t.muxer.AddAudioTrack(audioCodecId, 0, 0, 0)
+		t.muxer.AddAudioTrack(audioStream)
 	}
 
 	if utils.AVCodecIdNONE != videoCodecId {
-		t.muxer.AddVideoTrack(videoCodecId)
+		t.muxer.AddVideoTrack(videoStream)
 	}
 
 	var n int
-	t.header = make([]byte, 4096)
+	t.sequenceHeader = make([]byte, 4096)
 
-	// 生成推流的数据头(chunk+sequence header)
+	// 生成数据头
 	if audioStream != nil {
-		n += t.muxer.WriteAudioData(t.header[12:], true)
-		extra := audioStream.Extra()
-		copy(t.header[n+12:], extra)
+		n += t.muxer.AudioData.Marshal(t.sequenceHeader[12:], true)
+		extra := audioStream.Data
+		copy(t.sequenceHeader[n+12:], extra)
 		n += len(extra)
 
 		t.audioChunk.Length = n
-		t.audioChunk.MarshalHeader(t.header)
+		t.audioChunk.MarshalHeader(t.sequenceHeader)
 		n += 12
 	}
 
 	if videoStream != nil {
 		tmp := n
-		n += t.muxer.WriteVideoData(t.header[n+12:], 0, false, true)
-		extra := videoStream.CodecParameters().MP4ExtraData()
-		copy(t.header[n+12:], extra)
+		n += t.muxer.VideoData.Marshal(t.sequenceHeader[n+12:], 0, 0, true)
+		extra := videoStream.CodecParameters.MP4ExtraData()
+		copy(t.sequenceHeader[n+12:], extra)
 		n += len(extra)
 
 		t.videoChunk.Length = 5 + len(extra)
-		t.videoChunk.MarshalHeader(t.header[tmp:])
+		t.videoChunk.MarshalHeader(t.sequenceHeader[tmp:])
 		n += 12
 	}
 
 	// 创建元数据chunk
 	var body [1024]byte
-	amf0 := libflv.AMF0{}
-	amf0.AddString("onMetaData")
-	amf0.Add(t.muxer.MetaData())
-	length, _ := amf0.Marshal(body[:])
+	metadata := amf0.Data{}
+	metadata.AddString("onMetaData")
+	metadata.Add(t.muxer.MetaData())
+	length, _ := metadata.Marshal(body[:])
 
-	metaData := librtmp.Chunk{
-		Type:           librtmp.ChunkType0,
-		ChunkStreamID_: 5,
+	metaDataChunk := rtmp.Chunk{
+		Type:          rtmp.ChunkType0,
+		ChunkStreamID: 5,
 		Timestamp:     0,
-		TypeID:         librtmp.MessageTypeIDDataAMF0,
+		TypeID:        rtmp.MessageTypeIDDataAMF0,
 		StreamID:      1,
 		Body:          body[:length],
 		Length:        length,
 	}
 
 	var tmp [1600]byte
-	size := metaData.Marshal(tmp[:], librtmp.ChunkSize)
-	// metadata 放在sequence之前
-	copy(t.header[size:], t.header[:n])
-	copy(t.header, tmp[:][:size])
+	size := metaDataChunk.Marshal(tmp[:], rtmp.MaxChunkSize)
+	// metadata放在sequence之前
+	copy(t.sequenceHeader[size:], t.sequenceHeader[:n])
+	copy(t.sequenceHeader, tmp[:][:size])
 
 	n += size
-	t.headerSize = n
+	t.sequenceHeader = t.sequenceHeader[:n]
 	t.MWBuffer = stream.NewMergeWritingBuffer(t.ExistVideo)
 	return nil
 }
@@ -219,15 +224,15 @@ func (t *transStream) Close() ([][]byte, int64, error) {
 	return t.OutBuffer[:t.OutBufferSize], 0, nil
 }
 
-func NewTransStream(chunkSize int, metaData *libflv.AMF0Object) stream.TransStream {
+func NewTransStream(chunkSize int, metaData *amf0.Object) stream.TransStream {
 	return &transStream{chunkSize: chunkSize, metaData: metaData}
 }
 
 func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
 	// 获取推流的元数据
-	var metaData *libflv.AMF0Object
+	var metaData *amf0.Object
 	if stream.SourceTypeRtmp == source.GetType() {
-		metaData = source.(*Publisher).Stack.MetaData()
+		metaData = source.(*Publisher).Stack.Metadata()
 	}
-	return NewTransStream(librtmp.ChunkSize, metaData), nil
+	return NewTransStream(rtmp.MaxChunkSize, metaData), nil
 }

rtsp/rtsp_handler.go

@@ -1,5 +1,6 @@
 package rtsp
 
+
 import (
 	"fmt"
 	"github.com/lkmio/avformat/utils"
@@ -70,7 +71,7 @@ func (h handler) Process(session *session, method string, url_ *url.URL, headers
 		return fmt.Errorf("please establish a session first")
 	}
 
-	source, _ := stream.Path2SourceId(url_.Path, "")
+	source, _ := stream.Path2SourceID(url_.Path, "")
 
 	//反射调用各个处理函数
 	results := m.Call([]reflect.Value{

rtsp/rtsp_server.go

@@ -1,9 +1,10 @@
 package rtsp
 
+
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
+	"github.com/lkmio/transport"
 	"net"
 	"runtime"
 )

rtsp/rtsp_session.go

@@ -1,5 +1,6 @@
 package rtsp
 
+
 import (
 	"bufio"
 	"bytes"

rtsp/rtsp_sink.go

@@ -1,11 +1,11 @@
 package rtsp
 
 import (
-	"github.com/lkmio/avformat/librtp"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtp"
+	"github.com/lkmio/transport"
 	"github.com/pion/rtcp"
 	"net"
 	"time"
@@ -21,7 +21,7 @@ var (
 type Sink struct {
 	stream.BaseSink
 
-	senders []*librtp.RtpSender // 一个rtsp源, 可能存在多个流, 每个流都需要拉取
+	senders []*rtp.RtpSender // 一个rtsp源, 可能存在多个流, 每个流都需要拉取
 	cb      func(sdp string) // sdp回调, 响应describe
 }
 
@@ -31,7 +31,7 @@ func (s *Sink) StartStreaming(transStream stream.TransStream) error {
 		return nil
 	}
 
-	s.senders = make([]*librtp.RtpSender, transStream.TrackSize())
+	s.senders = make([]*rtp.RtpSender, transStream.TrackSize())
 	// sdp回调给sink, sink应答给describe请求
 	if s.cb != nil {
 		s.cb(transStream.(*TransStream).sdp)
@@ -47,19 +47,19 @@ func (s *Sink) AddSender(index int, tcp bool, ssrc uint32) (uint16, uint16, erro
 	var rtpPort uint16
 	var rtcpPort uint16
 
-	sender := librtp.RtpSender{
+	sender := rtp.RtpSender{
 		SSRC: ssrc,
 	}
 
 	if tcp {
 		s.TCPStreaming = true
 	} else {
-		sender.Rtp, err = TransportManger.NewUDPServer("0.0.0.0")
+		sender.Rtp, err = TransportManger.NewUDPServer()
 		if err != nil {
 			return 0, 0, err
 		}
 
-		sender.Rtcp, err = TransportManger.NewUDPServer("0.0.0.0")
+		sender.Rtcp, err = TransportManger.NewUDPServer()
 		if err != nil {
 			sender.Rtp.Close()
 			sender.Rtp = nil

rtsp/rtsp_stream.go

@@ -3,11 +3,12 @@ package rtsp
 import (
 	"encoding/binary"
 	"fmt"
-	"github.com/lkmio/avformat/libavc"
-	"github.com/lkmio/avformat/librtp"
-	"github.com/lkmio/avformat/librtsp/sdp"
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/avc"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtp"
+	"github.com/pion/sdp/v3"
 	"net"
 	"strconv"
 )
@@ -38,21 +39,22 @@ func (t *TransStream) OverTCP(data []byte, channel int) {
 	binary.BigEndian.PutUint16(data[2:], uint16(len(data)-4))
 }
 
-func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
+func (t *TransStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	t.ClearOutStreamBuffer()
 
 	var ts uint32
-	track := t.RtspTracks[packet.Index()]
-	if utils.AVMediaTypeAudio == packet.MediaType() {
+	track := t.RtspTracks[packet.Index]
+	if utils.AVMediaTypeAudio == packet.MediaType {
 		ts = uint32(packet.ConvertPts(track.Rate))
-		t.PackRtpPayload(track, packet.Index(), packet.Data(), ts)
-	} else if utils.AVMediaTypeVideo == packet.MediaType() {
+		t.PackRtpPayload(track, packet.Index, packet.Data, ts)
+	} else if utils.AVMediaTypeVideo == packet.MediaType {
 		ts = uint32(packet.ConvertPts(track.Rate))
-		data := libavc.RemoveStartCode(packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()].Stream))
-		t.PackRtpPayload(track, packet.Index(), data, ts)
+		annexBData := avformat.AVCCPacket2AnnexB(t.BaseTransStream.Tracks[packet.Index].Stream, packet)
+		data := avc.RemoveStartCode(annexBData)
+		t.PackRtpPayload(track, packet.Index, data, ts)
 	}
 
-	return t.OutBuffer[:t.OutBufferSize], int64(ts), utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame(), nil
+	return t.OutBuffer[:t.OutBufferSize], int64(ts), utils.AVMediaTypeVideo == packet.MediaType && packet.Key, nil
 }
 
 func (t *TransStream) ReadExtraData(ts int64) ([][]byte, int64, error) {
@@ -65,7 +67,7 @@ func (t *TransStream) ReadExtraData(ts int64) ([][]byte, int64, error) {
 		// 回滚序号和时间戳
 		index := int(track.StartSeq) - len(track.ExtraDataBuffer)
 		for i, bytes := range track.ExtraDataBuffer {
-			librtp.RollbackSeq(bytes[OverTcpHeaderSize:], index+i+1)
+			rtp.RollbackSeq(bytes[OverTcpHeaderSize:], index+i+1)
 			binary.BigEndian.PutUint32(bytes[OverTcpHeaderSize+4:], uint32(ts))
 		}
 
@@ -98,9 +100,9 @@ func (t *TransStream) AddTrack(track *stream.Track) error {
 		return err
 	}
 
-	payloadType, ok := librtp.CodecIdPayloads[track.Stream.CodecId()]
+	payloadType, ok := rtp.CodecIdPayloads[track.Stream.CodecID]
 	if !ok {
-		return fmt.Errorf("no payload type was found for codecid: %d", track.Stream.CodecId())
+		return fmt.Errorf("no payload type was found for codecid: %d", track.Stream.CodecID)
 	}
 
 	// 恢复上次拉流的序号
@@ -111,35 +113,34 @@ func (t *TransStream) AddTrack(track *stream.Track) error {
 	}
 
 	// 创建RTP封装器
-	var muxer librtp.Muxer
-	if utils.AVCodecIdH264 == track.Stream.CodecId() {
-		muxer = librtp.NewH264Muxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
-	} else if utils.AVCodecIdH265 == track.Stream.CodecId() {
-		muxer = librtp.NewH265Muxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
-	} else if utils.AVCodecIdAAC == track.Stream.CodecId() {
-		muxer = librtp.NewAACMuxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
-	} else if utils.AVCodecIdPCMALAW == track.Stream.CodecId() || utils.AVCodecIdPCMMULAW == track.Stream.CodecId() {
-		muxer = librtp.NewMuxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
+	var muxer rtp.Muxer
+	if utils.AVCodecIdH264 == track.Stream.CodecID {
+		muxer = rtp.NewH264Muxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
+	} else if utils.AVCodecIdH265 == track.Stream.CodecID {
+		muxer = rtp.NewH265Muxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
+	} else if utils.AVCodecIdAAC == track.Stream.CodecID {
+		muxer = rtp.NewAACMuxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
+	} else if utils.AVCodecIdPCMALAW == track.Stream.CodecID || utils.AVCodecIdPCMMULAW == track.Stream.CodecID {
+		muxer = rtp.NewMuxer(payloadType.Pt, int(startSeq), 0xFFFFFFFF)
 	}
 
-	rtspTrack := NewRTSPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, track.Stream.Type())
+	rtspTrack := NewRTSPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, track.Stream.MediaType)
 	t.RtspTracks = append(t.RtspTracks, rtspTrack)
 	index := len(t.RtspTracks) - 1
 
 	// 将sps和pps按照单一模式打包
 	bufferIndex := t.buffer.Index()
-	if utils.AVMediaTypeVideo == track.Stream.Type() {
-		parameters := track.Stream.CodecParameters()
-
-		if utils.AVCodecIdH265 == track.Stream.CodecId() {
-			bytes := parameters.(*utils.HEVCCodecData).VPS()
-			t.PackRtpPayload(rtspTrack, index, libavc.RemoveStartCode(bytes[0]), 0)
+	if utils.AVMediaTypeVideo == track.Stream.MediaType {
+		parameters := track.Stream.CodecParameters
+		if utils.AVCodecIdH265 == track.Stream.CodecID {
+			bytes := parameters.(*avformat.HEVCCodecData).VPS()
+			t.PackRtpPayload(rtspTrack, index, avc.RemoveStartCode(bytes[0]), 0)
 		}
 
 		spsBytes := parameters.SPS()
 		ppsBytes := parameters.PPS()
-		t.PackRtpPayload(rtspTrack, index, libavc.RemoveStartCode(spsBytes[0]), 0)
-		t.PackRtpPayload(rtspTrack, index, libavc.RemoveStartCode(ppsBytes[0]), 0)
+		t.PackRtpPayload(rtspTrack, index, avc.RemoveStartCode(spsBytes[0]), 0)
+		t.PackRtpPayload(rtspTrack, index, avc.RemoveStartCode(ppsBytes[0]), 0)
 
 		// 拷贝扩展数据的rtp包
 		size := t.buffer.Index() - bufferIndex
@@ -191,7 +192,7 @@ func (t *TransStream) WriteHeader() error {
 	}
 
 	for i, track := range t.Tracks {
-		payloadType, _ := librtp.CodecIdPayloads[track.Stream.CodecId()]
+		payloadType, _ := rtp.CodecIdPayloads[track.Stream.CodecID]
 		mediaDescription := sdp.MediaDescription{
 			ConnectionInformation: &sdp.ConnectionInformation{
 				NetworkType: "IN",
@@ -209,10 +210,10 @@ func (t *TransStream) WriteHeader() error {
 		mediaDescription.MediaName.Protos = []string{"RTP", "AVP"}
 		mediaDescription.MediaName.Formats = []string{strconv.Itoa(payloadType.Pt)}
 
-		if utils.AVMediaTypeAudio == track.Stream.Type() {
+		if utils.AVMediaTypeAudio == track.Stream.MediaType {
 			mediaDescription.MediaName.Media = "audio"
 
-			if utils.AVCodecIdAAC == track.Stream.CodecId() {
+			if utils.AVCodecIdAAC == track.Stream.CodecID {
 				//[14496-3], [RFC6416] profile-level-id:
 				//1 : Main Audio Profile Level 1
 				//9 : Speech Audio Profile Level 1

rtsp/rtsp_track.go

@@ -1,8 +1,9 @@
 package rtsp
 
+
 import (
-	"github.com/lkmio/avformat/librtp"
 	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/rtp"
 )
 
 // Track RtspTrack 对rtsp每路输出流的封装
@@ -13,14 +14,14 @@ type Track struct {
 	StartSeq  uint16
 	EndSeq    uint16
 
-	Muxer           librtp.Muxer
+	Muxer           rtp.Muxer
 	ExtraDataBuffer [][]byte // 缓存带有编码信息的rtp包, 对所有sink通用
 }
 
 func (r *Track) Close() {
 }
 
-func NewRTSPTrack(muxer librtp.Muxer, pt byte, rate int, mediaType utils.AVMediaType) *Track {
+func NewRTSPTrack(muxer rtp.Muxer, pt byte, rate int, mediaType utils.AVMediaType) *Track {
 	stream := &Track{
 		PT:        pt,
 		Rate:      rate,

stream/config.go

@@ -232,11 +232,6 @@ func GetStreamPlayUrls(source string) []string {
 
 // DumpStream2File 保存推流到文件, 用4字节帧长分割
 func DumpStream2File(sourceType SourceType, conn net.Conn, data []byte) {
-	if err := os.MkdirAll("dump", 0666); err != nil {
-		log.Sugar.Errorf("创建dump文件夹失败 err:%s", err.Error())
-		return
-	}
-
 	path := fmt.Sprintf("dump/%s-%s", sourceType.String(), conn.RemoteAddr().String())
 	path = strings.ReplaceAll(path, ":", ".")
 

stream/gop_buffer.go

@@ -1,6 +1,7 @@
 package stream
 
 import (
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/collections"
 	"github.com/lkmio/avformat/utils"
 )
@@ -9,14 +10,14 @@ import (
 type GOPBuffer interface {
 
 	// AddPacket Return bool 缓存帧是否成功, 如果首帧非关键帧, 缓存失败
-	AddPacket(packet utils.AVPacket) bool
+	AddPacket(packet *avformat.AVPacket) bool
 
 	// SetDiscardHandler 设置丢弃帧时的回调
-	SetDiscardHandler(handler func(packet utils.AVPacket))
+	SetDiscardHandler(handler func(packet *avformat.AVPacket))
 
-	PeekAll(handler func(packet utils.AVPacket))
+	PeekAll(handler func(packet *avformat.AVPacket))
 
-	Peek(index int) utils.AVPacket
+	Peek(index int) *avformat.AVPacket
 
 	Size() int
 
@@ -26,24 +27,24 @@ type GOPBuffer interface {
 }
 
 type streamBuffer struct {
-	buffer             collections.RingBuffer
+	buffer             collections.RingBuffer[*avformat.AVPacket]
 	existVideoKeyFrame bool
-	discardHandler     func(packet utils.AVPacket)
+	discardHandler     func(packet *avformat.AVPacket)
 }
 
-func (s *streamBuffer) AddPacket(packet utils.AVPacket) bool {
+func (s *streamBuffer) AddPacket(packet *avformat.AVPacket) bool {
 	// 缓存满,清空
 	if s.Size()+1 == s.buffer.Capacity() {
 		s.Clear()
 	}
 
 	// 丢弃首帧视频非关键帧
-	if utils.AVMediaTypeVideo == packet.MediaType() && !s.existVideoKeyFrame && !packet.KeyFrame() {
+	if utils.AVMediaTypeVideo == packet.MediaType && !s.existVideoKeyFrame && !packet.Key {
 		return false
 	}
 
 	// 丢弃前一组GOP
-	videoKeyFrame := utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame()
+	videoKeyFrame := utils.AVMediaTypeVideo == packet.MediaType && packet.Key
 	if videoKeyFrame {
 		if s.existVideoKeyFrame {
 			s.discard()
@@ -56,7 +57,7 @@ func (s *streamBuffer) AddPacket(packet utils.AVPacket) bool {
 	return true
 }
 
-func (s *streamBuffer) SetDiscardHandler(handler func(packet utils.AVPacket)) {
+func (s *streamBuffer) SetDiscardHandler(handler func(packet *avformat.AVPacket)) {
 	s.discardHandler = handler
 }
 
@@ -65,36 +66,36 @@ func (s *streamBuffer) discard() {
 		pkt := s.buffer.Pop()
 
 		if s.discardHandler != nil {
-			s.discardHandler(pkt.(utils.AVPacket))
+			s.discardHandler(pkt)
 		}
 	}
 
 	s.existVideoKeyFrame = false
 }
 
-func (s *streamBuffer) Peek(index int) utils.AVPacket {
+func (s *streamBuffer) Peek(index int) *avformat.AVPacket {
 	utils.Assert(index < s.buffer.Size())
 	head, tail := s.buffer.Data()
 
 	if index < len(head) {
-		return head[index].(utils.AVPacket)
+		return head[index]
 	} else {
-		return tail[index-len(head)].(utils.AVPacket)
+		return tail[index-len(head)]
 	}
 }
 
-func (s *streamBuffer) PeekAll(handler func(packet utils.AVPacket)) {
+func (s *streamBuffer) PeekAll(handler func(packet *avformat.AVPacket)) {
 	head, tail := s.buffer.Data()
 
 	if head != nil {
 		for _, value := range head {
-			handler(value.(utils.AVPacket))
+			handler(value)
 		}
 	}
 
 	if tail != nil {
 		for _, value := range tail {
-			handler(value.(utils.AVPacket))
+			handler(value)
 		}
 	}
 }
@@ -112,5 +113,5 @@ func (s *streamBuffer) Close() {
 }
 
 func NewStreamBuffer() GOPBuffer {
-	return &streamBuffer{buffer: collections.NewRingBuffer(1000), existVideoKeyFrame: false}
+	return &streamBuffer{buffer: collections.NewRingBuffer[*avformat.AVPacket](1000), existVideoKeyFrame: false}
 }

stream/m3u8_writer.go

@@ -0,0 +1,173 @@
+package stream
+
+import (
+	"bytes"
+	"github.com/lkmio/avformat/collections"
+	"math"
+	"strconv"
+)
+
+const (
+	ExtM3u      = "EXTM3U"
+	ExtXVersion = "EXT-X-VERSION" //在文件中唯一
+
+	ExtINF              = "EXTINF"              //<duration>(浮点类型, 版本小于3用整型),[<title>]
+	ExXByteRange        = "EXT-X-BYTERANGE"     //版本4及以上,分片位置
+	ExtXDiscontinuity   = "EXT-X-DISCONTINUITY" //后面的切片不连续, 文件格式、时间戳发生变化
+	ExtXKey             = "EXT-X-KEY"           //加密使用
+	ExtXMap             = "EXT-X-MAP"           //音视频的元数据
+	ExtXProgramDateTime = "EXT-X-PROGRAM-DATE-TIME"
+	ExtXDateRange       = "EXT-X-DATERANGE"
+
+	ExtXTargetDuration        = "EXT-X-TARGETDURATION" //切片最大时长, 整型单位秒
+	ExtXMediaSequence         = "EXT-X-MEDIA-SEQUENCE" //第一个切片序号
+	ExtXDiscontinuitySequence = "EXT-X-DISCONTINUITY-SEQUENCE"
+	ExtXEndList               = "EXT-X-ENDLIST"
+	ExtXPlaylistType          = "EXT-X-PLAYLIST-TYPE"
+	ExtXIFramesOnly           = "EXT-X-I-FRAMES-ONLY"
+
+	ExtXMedia           = "EXT-X-MEDIA"
+	ExtXStreamINF       = "EXT-X-STREAM-INF"
+	ExtXIFrameStreamINF = "EXT-X-I-FRAME-STREAM-INF"
+	ExtXSessionData     = "EXT-X-SESSION-DATA"
+	ExtXSessionKey      = "EXT-X-SESSION-KEY"
+
+	ExtXIndependentSegments = "EXT-X-INDEPENDENT-SEGMENTS"
+	ExtXStart               = "EXT-X-START"
+)
+
+//HttpContent-Type头必须是"application/vnd.apple.mpegurl"或"audio/mpegurl"
+//无BOM
+
+type Segment struct {
+	Duration float32
+	Url      string
+	Sequence int
+	Path     string
+}
+
+type M3U8Writer interface {
+	// AddSegment 添加切片
+	//@Params  duration 切片时长
+	//@Params  url m3u8列表中切片的url
+	//@Params  sequence m3u8列表中的切片序号
+	//@Params  path 切片位于磁盘中的绝对路径
+	AddSegment(duration float32, url string, sequence int, path string)
+
+	String() string
+
+	// Size 返回切片文件数量
+	Size() int
+
+	// Get Head 返回指定索引切片文件
+	Get(index int) Segment
+}
+
+type m3u8Writer struct {
+	stringBuffer *bytes.Buffer
+	segments     *collections.Queue[Segment]
+}
+
+func (m *m3u8Writer) AddSegment(duration float32 /*title string,*/, url string, sequence int, path string) {
+	if m.segments.IsFull() {
+		m.segments.Pop()
+	}
+
+	m.segments.Push(Segment{Duration: duration, Url: url, Sequence: sequence, Path: path})
+}
+
+// 返回切片时长最长的值(秒)
+func (m *m3u8Writer) targetDuration() int {
+	var targetDuration int
+	head, tail := m.segments.Data()
+
+	compute := func(playlist []Segment) {
+		for _, segment := range playlist {
+			// 会影响播放器缓存.
+			round := int(math.Ceil(float64(segment.Duration)))
+			if round > targetDuration {
+				targetDuration = round
+			}
+		}
+	}
+
+	if head != nil {
+		compute(head)
+	}
+
+	if tail != nil {
+		compute(tail)
+	}
+
+	return targetDuration
+}
+
+func (m *m3u8Writer) String() string {
+	// 仅实现简单的播放列表
+	head, tail := m.segments.Data()
+	if head == nil {
+		return ""
+	}
+
+	m.stringBuffer.Reset()
+	m.stringBuffer.WriteString("#EXTM3U\r\n")
+	// 仅实现第三个版本
+	m.stringBuffer.WriteString("#EXT-X-VERSION:3\r\n")
+	m.stringBuffer.WriteString("#EXT-X-TARGETDURATION:")
+	m.stringBuffer.WriteString(strconv.Itoa(m.targetDuration()))
+	m.stringBuffer.WriteString("\r\n")
+	m.stringBuffer.WriteString("#EXT-X-MEDIA-SEQUENCE:")
+	m.stringBuffer.WriteString(strconv.Itoa(head[0].Sequence))
+	m.stringBuffer.WriteString("\r\n")
+
+	appendSegments := func(playlist []Segment) {
+		for _, segment := range playlist {
+			m.stringBuffer.WriteString("#EXTINF:")
+			m.stringBuffer.WriteString(strconv.FormatFloat(float64(segment.Duration), 'f', -1, 32))
+			m.stringBuffer.WriteString(",\r\n")
+			m.stringBuffer.WriteString(segment.Url + "%s") // %s用于替换每个sink的拉流key
+			m.stringBuffer.WriteString("\r\n")
+		}
+	}
+
+	if head != nil {
+		appendSegments(head)
+	}
+
+	if tail != nil {
+		appendSegments(tail)
+	}
+
+	return m.stringBuffer.String()
+}
+
+func (m *m3u8Writer) Size() int {
+	var size int
+	head, tail := m.segments.Data()
+
+	if head != nil {
+		size += len(head)
+	}
+
+	if tail != nil {
+		size += len(tail)
+	}
+
+	return size
+}
+
+func (m *m3u8Writer) Get(index int) Segment {
+	head, tail := m.segments.Data()
+	if index >= len(head) {
+		return tail[index-len(head)]
+	} else {
+		return head[index]
+	}
+}
+
+func NewM3U8Writer(len int) M3U8Writer {
+	return &m3u8Writer{
+		stringBuffer: bytes.NewBuffer(make([]byte, 0, 1024*10)),
+		segments:     collections.NewQueue[Segment](len),
+	}
+}

stream/sink.go

@@ -3,9 +3,9 @@ package stream
 import (
 	"context"
 	"fmt"
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
+	"github.com/lkmio/transport"
 	"net"
 	"net/url"
 	"sync"

stream/source.go

@@ -2,16 +2,16 @@ package stream
 
 import (
 	"fmt"
-	"github.com/lkmio/avformat/collections"
-	"github.com/lkmio/avformat/transport"
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/lkm/log"
+	"github.com/lkmio/transport"
 	"net"
 	"net/url"
+	"strings"
 	"sync"
 	"sync/atomic"
 	"time"
 
-	"github.com/lkmio/avformat/stream"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/transcode"
 )
@@ -64,23 +64,8 @@ type Source interface {
 	// IsCompleted 所有推流track是否解析完毕
 	IsCompleted() bool
 
-	// FindOrCreatePacketBuffer 查找或者创建AVPacket的内存池
-	FindOrCreatePacketBuffer(index int, mediaType utils.AVMediaType) collections.MemoryPool
-
 	// OnDiscardPacket GOP缓存溢出回调, 释放AVPacket
-	OnDiscardPacket(pkt utils.AVPacket)
-
-	// OnDeMuxStream 解析出AVStream回调
-	OnDeMuxStream(stream utils.AVStream)
-
-	// OnDeMuxStreamDone 所有track解析完毕回调, 后续的OnDeMuxStream回调不再处理
-	OnDeMuxStreamDone()
-
-	// OnDeMuxPacket 解析出AvPacket回调
-	OnDeMuxPacket(packet utils.AVPacket)
-
-	// OnDeMuxDone 所有流解析完毕回调
-	OnDeMuxDone()
+	OnDiscardPacket(pkt *avformat.AVPacket)
 
 	Init(receiveQueueSize int)
 
@@ -135,7 +120,7 @@ type PublishSource struct {
 	state SessionState
 	Conn  net.Conn
 
-	TransDeMuxer     stream.DeMuxer            // 负责从推流协议中解析出AVStream和AVPacket
+	TransDemuxer     avformat.Demuxer       // 负责从推流协议中解析出AVStream和AVPacket
 	recordSink       Sink                   // 每个Source的录制流
 	recordFilePath   string                 // 录制流文件路径
 	hlsStream        TransStream            // HLS传输流, 如果开启, 在@see writeHeader 函数中直接创建, 如果等拉流时再创建, 会进一步加大HLS延迟.
@@ -143,7 +128,7 @@ type PublishSource struct {
 	videoTranscoders []transcode.Transcoder // 视频解码器
 	originTracks     TrackManager           // 推流的音视频Streams
 	allStreamTracks  TrackManager           // 推流Streams+转码器获得的Stream
-	pktBuffers       [8]collections.MemoryPool // 推流每路的AVPacket缓存, AVPacket的data从该内存池中分配. 在GOP缓存溢出时,释放池中内存.
+	//pktBuffers       [8]collections.MemoryPool // 推流每路的AVPacket缓存, AVPacket的data从该内存池中分配. 在GOP缓存溢出时,释放池中内存.
 	gopBuffer GOPBuffer // GOP缓存, 音频和视频混合使用, 以视频关键帧为界, 缓存第二个视频关键帧时, 释放前一组gop. 如果不存在视频流, 不缓存音频
 
 	closed     atomic.Bool // source是否已经关闭
@@ -152,7 +137,7 @@ type PublishSource struct {
 
 	probeTimer *time.Timer // track解析超时计时器, 触发时执行@see writeHeader
 
-	TransStreams         map[TransStreamID]TransStream     // 所有的输出流, 持有Sink
+	TransStreams         map[TransStreamID]TransStream     // 所有输出流
 	sinks                map[SinkID]Sink                   // 保存所有Sink
 	slowSinks            map[SinkID]Sink                   // 因推流慢被挂起的sink队列
 	TransStreamSinks     map[TransStreamID]map[SinkID]Sink // 输出流对应的Sink
@@ -169,6 +154,7 @@ type PublishSource struct {
 	urlValues         url.Values         // 推流url携带的参数
 	createTime        time.Time          // source创建时间
 	statistics        *BitrateStatistics // 码流统计
+	streamLogger      avformat.OnUnpackStream2FileHandler
 }
 
 func (s *PublishSource) SetLastPacketTime(time2 time.Time) {
@@ -220,6 +206,17 @@ func (s *PublishSource) Init(receiveQueueSize int) {
 	s.slowSinks = make(map[SinkID]Sink, 12)
 	s.TransStreamSinks = make(map[TransStreamID]map[SinkID]Sink, len(transStreamFactories)+1)
 	s.statistics = NewBitrateStatistics()
+
+	// 此处设置的探测时长, 只是为了保证在probeTimeout触发前, 一直在探测
+	s.TransDemuxer.SetProbeDuration(60000)
+
+	// 启动探测超时计时器
+	s.probeTimer = time.AfterFunc(time.Duration(AppConfig.ProbeTimeout)*time.Millisecond, func() {
+		s.PostEvent(func() {
+			// s.writeHeader()
+			s.TransDemuxer.ProbeComplete()
+		})
+	})
 }
 
 func (s *PublishSource) CreateDefaultOutStreams() {
@@ -255,28 +252,6 @@ func (s *PublishSource) CreateDefaultOutStreams() {
 	}
 }
 
-// FindOrCreatePacketBuffer 查找或者创建AVPacket的内存池
-func (s *PublishSource) FindOrCreatePacketBuffer(index int, mediaType utils.AVMediaType) collections.MemoryPool {
-	if index >= cap(s.pktBuffers) {
-		panic("流路数过多...")
-	}
-
-	if s.pktBuffers[index] == nil {
-		if utils.AVMediaTypeAudio == mediaType {
-			s.pktBuffers[index] = collections.NewRbMemoryPool(48000 * 12)
-		} else if AppConfig.GOPCache {
-			// 开启GOP缓存
-			s.pktBuffers[index] = collections.NewRbMemoryPool(AppConfig.GOPBufferSize)
-		} else {
-			// 未开启GOP缓存
-			// 1M缓存大小, 单帧绰绰有余
-			s.pktBuffers[index] = collections.NewRbMemoryPool(1024 * 1000)
-		}
-	}
-
-	return s.pktBuffers[index]
-}
-
 func (s *PublishSource) Input(data []byte) error {
 	s.streamPipe <- data
 	s.statistics.Input(len(data))
@@ -328,20 +303,20 @@ func (s *PublishSource) CreateTransStream(id TransStreamID, protocol TransStream
 
 func (s *PublishSource) DispatchGOPBuffer(transStream TransStream) {
 	if s.gopBuffer != nil {
-		s.gopBuffer.PeekAll(func(packet utils.AVPacket) {
+		s.gopBuffer.PeekAll(func(packet *avformat.AVPacket) {
 			s.DispatchPacket(transStream, packet)
 		})
 	}
 }
 
 // DispatchPacket 分发AVPacket
-func (s *PublishSource) DispatchPacket(transStream TransStream, packet utils.AVPacket) {
+func (s *PublishSource) DispatchPacket(transStream TransStream, packet *avformat.AVPacket) {
 	data, timestamp, videoKey, err := transStream.Input(packet)
 	if err != nil || len(data) < 1 {
 		return
 	}
 
-	s.DispatchBuffer(transStream, packet.Index(), data, timestamp, videoKey)
+	s.DispatchBuffer(transStream, packet.Index, data, timestamp, videoKey)
 }
 
 // DispatchBuffer 分发传输流
@@ -427,26 +402,26 @@ func (s *PublishSource) doAddSink(sink Sink) bool {
 	}
 
 	if !disableAudio && utils.AVCodecIdNONE == audioCodecId {
-		audioCodecId = audioTrack.Stream.CodecId()
+		audioCodecId = audioTrack.Stream.CodecID
 	}
 	if !disableVideo && utils.AVCodecIdNONE == videoCodecId {
-		videoCodecId = videoTrack.Stream.CodecId()
+		videoCodecId = videoTrack.Stream.CodecID
 	}
 
 	// 创建音频转码器
-	if !disableAudio && audioCodecId != audioTrack.Stream.CodecId() {
+	if !disableAudio && audioCodecId != audioTrack.Stream.CodecID {
 		utils.Assert(false)
 	}
 
 	// 创建视频转码器
-	if !disableVideo && videoCodecId != videoTrack.Stream.CodecId() {
+	if !disableVideo && videoCodecId != videoTrack.Stream.CodecID {
 		utils.Assert(false)
 	}
 
 	// 查找传输流需要的所有track
 	var tracks []*Track
 	for _, track := range s.originTracks.All() {
-		if disableVideo && track.Stream.Type() == utils.AVMediaTypeVideo {
+		if disableVideo && track.Stream.MediaType == utils.AVMediaTypeVideo {
 			continue
 		}
 
@@ -504,7 +479,6 @@ func (s *PublishSource) doAddSink(sink Sink) bool {
 	_, ok := sink.GetConn().(*transport.Conn)
 	if ok && sink.IsTCPStreaming() && transStream.OutStreamBufferCapacity() > 2 {
 		length := transStream.OutStreamBufferCapacity() - 2
-		fmt.Printf("发送缓冲区容量: %d\r\n", length)
 		sink.EnableAsyncWriteMode(length)
 	}
 
@@ -617,17 +591,17 @@ func (s *PublishSource) DoClose() {
 
 	s.closed.Store(true)
 
-	if s.TransDeMuxer != nil {
-		s.TransDeMuxer.Close()
-		s.TransDeMuxer = nil
+	if s.TransDemuxer != nil {
+		s.TransDemuxer.Close()
+		s.TransDemuxer = nil
 	}
 
 	// 清空未写完的buffer
-	for _, buffer := range s.pktBuffers {
-		if buffer != nil {
-			buffer.Reset()
-		}
-	}
+	//for _, buffer := range s.pktBuffers {
+	//	if buffer != nil {
+	//		buffer.Reset()
+	//	}
+	//}
 
 	// 释放GOP缓存
 	if s.gopBuffer != nil {
@@ -735,40 +709,9 @@ func (s *PublishSource) Close() {
 }
 
 // OnDiscardPacket GOP缓存溢出丢弃回调
-func (s *PublishSource) OnDiscardPacket(packet utils.AVPacket) {
-	s.FindOrCreatePacketBuffer(packet.Index(), packet.MediaType()).FreeHead()
-}
-
-func (s *PublishSource) OnDeMuxStream(stream utils.AVStream) {
-	if s.completed {
-		log.Sugar.Warnf("添加%s track失败,已经WriteHeader. source: %s", stream.Type().ToString(), s.ID)
-		return
-	} else if !s.NotTrackAdded(stream.Index()) {
-		log.Sugar.Warnf("添加%s track失败,已经添加索引为%d的track. source: %s", stream.Type().ToString(), stream.Index(), s.ID)
-		return
-	}
-
-	s.originTracks.Add(NewTrack(stream, 0, 0))
-	s.allStreamTracks.Add(NewTrack(stream, 0, 0))
-
-	// 启动track解析超时计时器
-	if len(s.originTracks.All()) == 1 {
-		s.probeTimer = time.AfterFunc(time.Duration(AppConfig.ProbeTimeout)*time.Millisecond, func() {
-			s.PostEvent(func() {
-				s.writeHeader()
-			})
-		})
-	}
-
-	if utils.AVMediaTypeVideo == stream.Type() {
-		s.existVideo = true
-	}
-
-	// 创建GOPBuffer
-	if AppConfig.GOPCache && s.existVideo && s.gopBuffer == nil {
-		s.gopBuffer = NewStreamBuffer()
-		// 设置GOP缓存溢出回调
-		s.gopBuffer.SetDiscardHandler(s.OnDiscardPacket)
+func (s *PublishSource) OnDiscardPacket(packet *avformat.AVPacket) {
+	if s.TransDemuxer != nil {
+		s.TransDemuxer.DiscardHeadPacket(packet.BufferIndex)
 	}
 }
 
@@ -797,7 +740,7 @@ func (s *PublishSource) writeHeader() {
 		// 恢复每路track的时间戳
 		tracks := s.originTracks.All()
 		for _, track := range tracks {
-			timestamps := streamInfo.Timestamps[track.Stream.CodecId()]
+			timestamps := streamInfo.Timestamps[track.Stream.CodecID]
 			track.Dts = timestamps[0]
 			track.Pts = timestamps[1]
 		}
@@ -805,7 +748,7 @@ func (s *PublishSource) writeHeader() {
 
 	// 纠正GOP中的时间戳
 	if s.gopBuffer != nil && s.gopBuffer.Size() != 0 {
-		s.gopBuffer.PeekAll(func(packet utils.AVPacket) {
+		s.gopBuffer.PeekAll(func(packet *avformat.AVPacket) {
 			s.CorrectTimestamp(packet)
 		})
 	}
@@ -833,7 +776,7 @@ func (s *PublishSource) IsCompleted() bool {
 // NotTrackAdded 返回该index对应的track是否没有添加
 func (s *PublishSource) NotTrackAdded(index int) bool {
 	for _, track := range s.originTracks.All() {
-		if track.Stream.Index() == index {
+		if track.Stream.Index == index {
 			return false
 		}
 	}
@@ -841,39 +784,84 @@ func (s *PublishSource) NotTrackAdded(index int) bool {
 	return true
 }
 
-func (s *PublishSource) OnDeMuxStreamDone() {
-	s.writeHeader()
-}
-
-func (s *PublishSource) CorrectTimestamp(packet utils.AVPacket) {
+func (s *PublishSource) CorrectTimestamp(packet *avformat.AVPacket) {
 	// 对比第一包的时间戳和上次推流的最后时间戳。如果小于上次的推流时间戳，则在原来的基础上累加。
 	if s.streamEndInfo != nil && !s.timestampModeDecided {
 		s.timestampModeDecided = true
 
-		timestamps := s.streamEndInfo.Timestamps[packet.CodecId()]
+		timestamps := s.streamEndInfo.Timestamps[packet.CodecID]
 		s.accumulateTimestamps = true
 		log.Sugar.Infof("累加时间戳 上次推流dts: %d, pts: %d", timestamps[0], timestamps[1])
 	}
 
-	track := s.originTracks.Find(packet.CodecId())
-	duration := packet.Duration(packet.Timebase())
+	track := s.originTracks.Find(packet.CodecID)
+	duration := packet.GetDuration(packet.Timebase)
 
 	// 根据duration来累加时间戳
 	if s.accumulateTimestamps {
-		offset := packet.Pts() - packet.Dts()
-		packet.SetDts(track.Dts + duration)
-		packet.SetPts(packet.Dts() + offset)
+		offset := packet.Pts - packet.Dts
+		packet.Dts = track.Dts + duration
+		packet.Pts = packet.Dts + offset
 	}
 
-	track.Dts = packet.Dts()
-	track.Pts = packet.Pts()
+	track.Dts = packet.Dts
+	track.Pts = packet.Pts
 	track.FrameDuration = int(duration)
 }
 
-func (s *PublishSource) OnDeMuxPacket(packet utils.AVPacket) {
+func (s *PublishSource) OnNewTrack(track avformat.Track) {
+	if AppConfig.Debug {
+		s.streamLogger.Path = "dump/" + strings.ReplaceAll(s.ID, "/", "_")
+		s.streamLogger.OnNewTrack(track)
+	}
+
+	stream := track.GetStream()
+
+	if s.completed {
+		log.Sugar.Warnf("添加%s track失败,已经WriteHeader. source: %s", stream.MediaType, s.ID)
+		return
+	} else if !s.NotTrackAdded(stream.Index) {
+		log.Sugar.Warnf("添加%s track失败,已经添加索引为%d的track. source: %s", stream.MediaType, stream.Index, s.ID)
+		return
+	}
+
+	s.originTracks.Add(NewTrack(stream, 0, 0))
+	s.allStreamTracks.Add(NewTrack(stream, 0, 0))
+
+	if utils.AVMediaTypeVideo == stream.MediaType {
+		s.existVideo = true
+	}
+
+	// 创建GOPBuffer
+	if AppConfig.GOPCache && s.existVideo && s.gopBuffer == nil {
+		s.gopBuffer = NewStreamBuffer()
+		// 设置GOP缓存溢出回调
+		s.gopBuffer.SetDiscardHandler(s.OnDiscardPacket)
+	}
+}
+
+func (s *PublishSource) OnTrackComplete() {
+	if AppConfig.Debug {
+		s.streamLogger.OnTrackComplete()
+	}
+
+	s.writeHeader()
+}
+
+func (s *PublishSource) OnTrackNotFind() {
+	if AppConfig.Debug {
+		s.streamLogger.OnTrackNotFind()
+	}
+}
+
+func (s *PublishSource) OnPacket(packet *avformat.AVPacket) {
+	if AppConfig.Debug {
+		s.streamLogger.OnPacket(packet)
+	}
+
 	// track超时，忽略推流数据
-	if s.NotTrackAdded(packet.Index()) {
-		s.FindOrCreatePacketBuffer(packet.Index(), packet.MediaType()).FreeTail()
+	if s.NotTrackAdded(packet.Index) {
+		s.TransDemuxer.DiscardBackPacket(packet.BufferIndex)
 		return
 	}
 
@@ -883,7 +871,7 @@ func (s *PublishSource) OnDeMuxPacket(packet utils.AVPacket) {
 	}
 
 	// 遇到关键帧, 恢复推流
-	if utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame() && len(s.slowSinks) > 0 {
+	if utils.AVMediaTypeVideo == packet.MediaType && packet.Key && len(s.slowSinks) > 0 {
 		s.ResumeStreaming()
 	}
 
@@ -899,14 +887,10 @@ func (s *PublishSource) OnDeMuxPacket(packet utils.AVPacket) {
 
 	// 未开启GOP缓存或只存在音频流, 释放掉内存
 	if !AppConfig.GOPCache || !s.existVideo {
-		s.FindOrCreatePacketBuffer(packet.Index(), packet.MediaType()).FreeTail()
+		s.TransDemuxer.DiscardBackPacket(packet.BufferIndex)
 	}
 }
 
-func (s *PublishSource) OnDeMuxDone() {
-
-}
-
 func (s *PublishSource) GetType() SourceType {
 	return s.Type
 }

stream/source_utils.go

@@ -1,10 +1,7 @@
 package stream
 
 import (
-	"encoding/hex"
 	"fmt"
-	"github.com/lkmio/avformat/libavc"
-	"github.com/lkmio/avformat/libhevc"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/log"
 	"net/url"
@@ -94,7 +91,7 @@ func (s SessionState) String() string {
 	panic(fmt.Sprintf("unknown session state %d", s))
 }
 
-func Path2SourceId(path string, suffix string) (string, error) {
+func Path2SourceID(path string, suffix string) (string, error) {
 	source := strings.TrimSpace(path)
 	if strings.HasPrefix(source, "/") {
 		source = source[1:]
@@ -129,95 +126,96 @@ func ParseUrl(name string) (string, url.Values) {
 	return name, nil
 }
 
-func ExtractVideoPacket(codec utils.AVCodecID, key, extractStream bool, data []byte, pts, dts int64, index, timebase int) (utils.AVStream, utils.AVPacket, error) {
-	var stream utils.AVStream
-
-	if utils.AVCodecIdH264 == codec {
-		//从关键帧中解析出sps和pps
-		if key && extractStream {
-			sps, pps, err := libavc.ParseExtraDataFromKeyNALU(data)
-			if err != nil {
-				log.Sugar.Errorf("从关键帧中解析sps pps失败 data:%s", hex.EncodeToString(data))
-				return nil, nil, err
-			}
-
-			codecData, err := utils.NewAVCCodecData(sps, pps)
-			if err != nil {
-				log.Sugar.Errorf("解析sps pps失败 data:%s sps:%s, pps:%s", hex.EncodeToString(data), hex.EncodeToString(sps), hex.EncodeToString(pps))
-				return nil, nil, err
-			}
-
-			stream = utils.NewAVStream(utils.AVMediaTypeVideo, 0, codec, codecData.AnnexBExtraData(), codecData)
-		}
-
-	} else if utils.AVCodecIdH265 == codec {
-		if key && extractStream {
-			vps, sps, pps, err := libhevc.ParseExtraDataFromKeyNALU(data)
-			if err != nil {
-				log.Sugar.Errorf("从关键帧中解析vps sps pps失败  data:%s", hex.EncodeToString(data))
-				return nil, nil, err
-			}
-
-			codecData, err := utils.NewHEVCCodecData(vps, sps, pps)
-			if err != nil {
-				log.Sugar.Errorf("解析sps pps失败 data:%s vps:%s sps:%s, pps:%s", hex.EncodeToString(data), hex.EncodeToString(vps), hex.EncodeToString(sps), hex.EncodeToString(pps))
-				return nil, nil, err
-			}
-
-			stream = utils.NewAVStream(utils.AVMediaTypeVideo, 0, codec, codecData.AnnexBExtraData(), codecData)
-		}
-
-	}
-
-	packet := utils.NewVideoPacket(data, dts, pts, key, utils.PacketTypeAnnexB, codec, index, timebase)
-	return stream, packet, nil
-}
-
-func ExtractAudioPacket(codec utils.AVCodecID, extractStream bool, data []byte, pts, dts int64, index, timebase int) (utils.AVStream, utils.AVPacket, error) {
-	var stream utils.AVStream
-	var packet utils.AVPacket
-	if utils.AVCodecIdAAC == codec {
-		//必须包含ADTSHeader
-		if len(data) < 7 {
-			return nil, nil, fmt.Errorf("need more data")
-		}
-
-		var skip int
-		header, err := utils.ReadADtsFixedHeader(data)
-		if err != nil {
-			log.Sugar.Errorf("读取ADTSHeader失败 data:%s", hex.EncodeToString(data[:7]))
-			return nil, nil, err
-		} else {
-			skip = 7
-			//跳过ADtsHeader长度
-			if header.ProtectionAbsent() == 0 {
-				skip += 2
-			}
-		}
-
-		if extractStream {
-			configData, err := utils.ADtsHeader2MpegAudioConfigData(header)
-			config, err := utils.ParseMpeg4AudioConfig(configData)
-			println(config)
-			if err != nil {
-				log.Sugar.Errorf("adt头转m4ac失败 data:%s", hex.EncodeToString(data[:7]))
-				return nil, nil, err
-			}
-
-			stream = utils.NewAVStream(utils.AVMediaTypeAudio, index, codec, configData, nil)
-		}
-
-		packet = utils.NewAudioPacket(data[skip:], dts, pts, codec, index, timebase)
-	} else if utils.AVCodecIdPCMALAW == codec || utils.AVCodecIdPCMMULAW == codec {
-		if extractStream {
-			stream = utils.NewAVStream(utils.AVMediaTypeAudio, index, codec, nil, nil)
-		}
-
-		packet = utils.NewAudioPacket(data, dts, pts, codec, index, timebase)
-	}
-
-	return stream, packet, nil
-}
+//
+//func ExtractVideoPacket(codec utils.AVCodecID, key, extractStream bool, data []byte, pts, dts int64, index, timebase int) (*avformat.AVStream, *avformat.AVPacket, error) {
+//	var stream *avformat.AVStream
+//
+//	if utils.AVCodecIdH264 == codec {
+//		//从关键帧中解析出sps和pps
+//		if key && extractStream {
+//			sps, pps, err := avc.ParseExtraDataFromKeyNALU(data)
+//			if err != nil {
+//				log.Sugar.Errorf("从关键帧中解析sps pps失败 data:%s", hex.EncodeToString(data))
+//				return nil, nil, err
+//			}
+//
+//			codecData, err := utils.NewAVCCodecData(sps, pps)
+//			if err != nil {
+//				log.Sugar.Errorf("解析sps pps失败 data:%s sps:%s, pps:%s", hex.EncodeToString(data), hex.EncodeToString(sps), hex.EncodeToString(pps))
+//				return nil, nil, err
+//			}
+//
+//			stream = avformat.NewAVStream(utils.AVMediaTypeVideo, 0, codec, codecData.AnnexBExtraData(), codecData)
+//		}
+//
+//	} else if utils.AVCodecIdH265 == codec {
+//		if key && extractStream {
+//			vps, sps, pps, err := hevc.ParseExtraDataFromKeyNALU(data)
+//			if err != nil {
+//				log.Sugar.Errorf("从关键帧中解析vps sps pps失败  data:%s", hex.EncodeToString(data))
+//				return nil, nil, err
+//			}
+//
+//			codecData, err := utils.NewHEVCCodecData(vps, sps, pps)
+//			if err != nil {
+//				log.Sugar.Errorf("解析sps pps失败 data:%s vps:%s sps:%s, pps:%s", hex.EncodeToString(data), hex.EncodeToString(vps), hex.EncodeToString(sps), hex.EncodeToString(pps))
+//				return nil, nil, err
+//			}
+//
+//			stream = avformat.NewAVStream(utils.AVMediaTypeVideo, 0, codec, codecData.AnnexBExtraData(), codecData)
+//		}
+//
+//	}
+//
+//	packet := avformat.NewVideoPacket(data, dts, pts, key, utils.PacketTypeAnnexB, codec, index, timebase)
+//	return stream, packet, nil
+//}
+//
+//func ExtractAudioPacket(codec utils.AVCodecID, extractStream bool, data []byte, pts, dts int64, index, timebase int) (*avformat.AVStream, *avformat.AVPacket, error) {
+//	var stream *avformat.AVStream
+//	var packet *avformat.AVPacket
+//	if utils.AVCodecIdAAC == codec {
+//		//必须包含ADTSHeader
+//		if len(data) < 7 {
+//			return nil, nil, fmt.Errorf("need more data")
+//		}
+//
+//		var skip int
+//		header, err := utils.ReadADtsFixedHeader(data)
+//		if err != nil {
+//			log.Sugar.Errorf("读取ADTSHeader失败 data:%s", hex.EncodeToString(data[:7]))
+//			return nil, nil, err
+//		} else {
+//			skip = 7
+//			//跳过ADtsHeader长度
+//			if header.ProtectionAbsent() == 0 {
+//				skip += 2
+//			}
+//		}
+//
+//		if extractStream {
+//			configData, err := utils.ADtsHeader2MpegAudioConfigData(header)
+//			config, err := utils.ParseMpeg4AudioConfig(configData)
+//			println(config)
+//			if err != nil {
+//				log.Sugar.Errorf("adt头转m4ac失败 data:%s", hex.EncodeToString(data[:7]))
+//				return nil, nil, err
+//			}
+//
+//			stream = avformat.NewAVStream(utils.AVMediaTypeAudio, index, codec, configData, nil)
+//		}
+//
+//		packet = utils.NewAudioPacket(data[skip:], dts, pts, codec, index, timebase)
+//	} else if utils.AVCodecIdPCMALAW == codec || utils.AVCodecIdPCMMULAW == codec {
+//		if extractStream {
+//			stream = avformat.NewAVStream(utils.AVMediaTypeAudio, index, codec, nil, nil)
+//		}
+//
+//		packet = utils.NewAudioPacket(data, dts, pts, codec, index, timebase)
+//	}
+//
+//	return stream, packet, nil
+//}
 
 // StartReceiveDataTimer 启动收流超时计时器
 func StartReceiveDataTimer(source Source) *time.Timer {
@@ -308,7 +306,7 @@ func LoopEvent(source Source) {
 			}
 
 			if err := source.Input(data); err != nil {
-				log.Sugar.Errorf("处理输入流失败 释放source:%s err:%s", source.GetID(), err.Error())
+				log.Sugar.Errorf("解析推流数据发生err: %s 释放source: %s", err.Error(), source.GetID())
 				source.DoClose()
 				return
 			}

stream/stream_endinfo.go

@@ -1,7 +1,6 @@
 package stream
 
 import (
-	"github.com/lkmio/avformat/libhls"
 	"github.com/lkmio/avformat/utils"
 	"sync"
 )
@@ -14,13 +13,13 @@ func init() {
 	streamEndInfoManager = &StreamEndInfoManager{sources: make(map[string]*StreamEndInfo, 32)}
 }
 
-// StreamEndInfo 保留结束推流Source的推流信息
-// 在结束推流时，如果还有拉流端没有断开，则保留一些推流信息(目前有时间戳、ts切片序号等等)。在下次推流时，使用该时间戳作为新传输流的起始时间戳，保证拉流端在拉流时不会重现pts和dts错误.
-// 如果在此之前，陆续有拉流端断开，直至sink计数为0，则会不再保留该信息。
+// StreamEndInfo 保存Source结束推流时的推流信息
+// 在结束推流时，如果还有拉流端没有断开，则保留一些推流信息(时间戳、ts切片序号等等)。在下次推流时，使用该时间戳作为新传输流的起始时间戳，保证拉流端在拉流时不会出现pts和dts错误.
+// 如果重新推流之前，陆续有拉流端断开，直至sink计数为0，删除保存的推流信息。
 type StreamEndInfo struct {
 	ID             string
 	Timestamps     map[utils.AVCodecID][2]int64 // 每路track结束时间戳
-	M3U8Writer     libhls.M3U8Writer            // 保存M3U8生成器
+	M3U8Writer     M3U8Writer                   // 保存M3U8生成器
 	PlaylistFormat *string                      // M3U8播放列表
 	RtspTracks     map[byte]uint16              // rtsp每路track的结束序号
 	FLVPrevTagSize uint32                       // flv的最后一个tag大小, 下次生成flv时作为prev tag size
@@ -32,7 +31,7 @@ func EqualsTracks(info *StreamEndInfo, tracks []*Track) bool {
 	}
 
 	for _, track := range tracks {
-		if _, ok := info.Timestamps[track.Stream.CodecId()]; !ok {
+		if _, ok := info.Timestamps[track.Stream.CodecID]; !ok {
 			return false
 		}
 	}

stream/stream_server.go

@@ -1,8 +1,8 @@
 package stream
 
 import (
-	"github.com/lkmio/avformat/transport"
 	"github.com/lkmio/lkm/log"
+	"github.com/lkmio/transport"
 	"net"
 )
 

stream/track.go

@@ -1,14 +1,16 @@
 package stream
 
-import "github.com/lkmio/avformat/utils"
+import (
+	"github.com/lkmio/avformat"
+)
 
 type Track struct {
-	Stream        utils.AVStream
+	Stream        *avformat.AVStream
 	Pts           int64 // 最新的PTS
 	Dts           int64 // 最新的DTS
 	FrameDuration int   // 单帧时长, timebase和推流一致
 }
 
-func NewTrack(stream utils.AVStream, dts, pts int64) *Track {
+func NewTrack(stream *avformat.AVStream, dts, pts int64) *Track {
 	return &Track{stream, dts, pts, 0}
 }

stream/track_manager.go

@@ -10,8 +10,8 @@ type TrackManager struct {
 
 func (s *TrackManager) Add(track *Track) {
 	for _, t := range s.tracks {
-		utils.Assert(t.Stream.Type() != track.Stream.Type())
-		utils.Assert(t.Stream.CodecId() != track.Stream.CodecId())
+		utils.Assert(t.Stream.MediaType != track.Stream.MediaType)
+		utils.Assert(t.Stream.CodecID != track.Stream.CodecID)
 	}
 
 	s.tracks = append(s.tracks, track)
@@ -19,7 +19,7 @@ func (s *TrackManager) Add(track *Track) {
 
 func (s *TrackManager) Find(id utils.AVCodecID) *Track {
 	for _, track := range s.tracks {
-		if track.Stream.CodecId() == id {
+		if track.Stream.CodecID == id {
 			return track
 		}
 	}
@@ -29,7 +29,7 @@ func (s *TrackManager) Find(id utils.AVCodecID) *Track {
 
 func (s *TrackManager) FindWithType(mediaType utils.AVMediaType) *Track {
 	for _, track := range s.tracks {
-		if track.Stream.Type() == mediaType {
+		if track.Stream.MediaType == mediaType {
 			return track
 		}
 	}
@@ -40,7 +40,7 @@ func (s *TrackManager) FindWithType(mediaType utils.AVMediaType) *Track {
 func (s *TrackManager) FindTracks(id utils.AVCodecID) []*Track {
 	var tracks []*Track
 	for _, track := range s.tracks {
-		if track.Stream.CodecId() == id {
+		if track.Stream.CodecID == id {
 			tracks = append(tracks, track)
 		}
 	}
@@ -51,7 +51,7 @@ func (s *TrackManager) FindTracks(id utils.AVCodecID) []*Track {
 func (s *TrackManager) FindTracksWithType(mediaType utils.AVMediaType) []*Track {
 	var tracks []*Track
 	for _, track := range s.tracks {
-		if track.Stream.Type() == mediaType {
+		if track.Stream.MediaType == mediaType {
 			tracks = append(tracks, track)
 		}
 	}

stream/trans_stream.go

@@ -1,6 +1,7 @@
 package stream
 
 import (
+	"github.com/lkmio/avformat"
 	"github.com/lkmio/avformat/utils"
 )
 
@@ -11,7 +12,7 @@ type TransStream interface {
 	SetID(id TransStreamID)
 
 	// Input 封装传输流, 返回合并写块、时间戳、合并写块是否包含视频关键帧
-	Input(packet utils.AVPacket) ([][]byte, int64, bool, error)
+	Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error)
 
 	AddTrack(track *Track) error
 
@@ -67,13 +68,13 @@ func (t *BaseTransStream) SetID(id TransStreamID) {
 	t.ID = id
 }
 
-func (t *BaseTransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
+func (t *BaseTransStream) Input(packet *avformat.AVPacket) ([][]byte, int64, bool, error) {
 	return nil, -1, false, nil
 }
 
 func (t *BaseTransStream) AddTrack(track *Track) error {
 	t.Tracks = append(t.Tracks, track)
-	if utils.AVMediaTypeVideo == track.Stream.Type() {
+	if utils.AVMediaTypeVideo == track.Stream.MediaType {
 		t.ExistVideo = true
 	}
 	return nil

stream/trans_utils.go

@@ -57,7 +57,7 @@ func GenerateTransStreamID(protocol TransStreamProtocol, tracks ...*Track) Trans
 	streamId = uint64(protocol) << 56
 
 	for i, track := range tracks {
-		id, ok := narrowCodecIds[int(track.Stream.CodecId())]
+		id, ok := narrowCodecIds[int(track.Stream.CodecID)]
 		utils.Assert(ok)
 
 		streamId |= uint64(id) << (48 - i*8)