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支持恢复上次推流的时间戳和序号

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This commit is contained in:
yangjiechina
2024-11-30 17:48:39 +08:00
parent 62542c3e4f
commit 92bd005fc4
39 changed files with 674 additions and 1228 deletions
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3
rtsp/rtsp_sink.go
View File

@@ -32,7 +32,7 @@ func (s *Sink) StartStreaming(transStream stream.TransStream) error {
// sdp回调给sink, sink应答给describe请求
if s.sdpCb != nil {
s.sdpCb(transStream.(*TranStream).sdp)
s.sdpCb(transStream.(*TransStream).sdp)
s.sdpCb = nil
}
@@ -126,6 +126,7 @@ func (s *Sink) Write(index int, data [][]byte, rtpTime int64) error {
func (s *Sink) isConnected(index int) bool {
return s.TCPStreaming || (s.senders[index] != nil && s.senders[index].RtpConn != nil)
}
func (s *Sink) Close() {
s.BaseSink.Close()

126
rtsp/rtsp_stream.go
View File

@@ -17,115 +17,129 @@ const (
OverTcpMagic = 0x24
)
// TranStream rtsp传输流封装
// TransStream rtsp传输流封装
// 低延迟是rtsp特性, 所以不考虑实现GOP缓存
type TranStream struct {
type TransStream struct {
stream.BaseTransStream
addr net.IPAddr
addrType string
urlFormat string
rtpTracks []*Track
RtspTracks []*Track
//oldTracks []*Track
oldTracks map[byte]uint16
sdp string
buffer *stream.ReceiveBuffer
buffer *stream.ReceiveBuffer // 保存封装后的rtp包
}
func (t *TranStream) OverTCP(data []byte, channel int) {
func (t *TransStream) OverTCP(data []byte, channel int) {
data[0] = OverTcpMagic
data[1] = byte(channel)
binary.BigEndian.PutUint16(data[2:], uint16(len(data)-4))
}
func (t *TranStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
func (t *TransStream) Input(packet utils.AVPacket) ([][]byte, int64, bool, error) {
t.ClearOutStreamBuffer()
var ts uint32
track := t.rtpTracks[packet.Index()]
track.seq = track.muxer.GetHeader().Seq
track := t.RtspTracks[packet.Index()]
if utils.AVMediaTypeAudio == packet.MediaType() {
ts = uint32(packet.ConvertPts(track.rate))
t.PackRtpPayload(track.muxer, packet.Index(), packet.Data(), ts)
ts = uint32(packet.ConvertPts(track.Rate))
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()]))
t.PackRtpPayload(track.muxer, packet.Index(), data, ts)
ts = uint32(packet.ConvertPts(track.Rate))
data := libavc.RemoveStartCode(packet.AnnexBPacketData(t.BaseTransStream.Tracks[packet.Index()].Stream))
t.PackRtpPayload(track, packet.Index(), data, ts)
}
return t.OutBuffer[:t.OutBufferSize], int64(ts), utils.AVMediaTypeVideo == packet.MediaType() && packet.KeyFrame(), nil
}
func (t *TranStream) ReadExtraData(ts int64) ([][]byte, int64, error) {
func (t *TransStream) ReadExtraData(ts int64) ([][]byte, int64, error) {
// 返回视频编码数据的rtp包
for _, track := range t.rtpTracks {
if utils.AVMediaTypeVideo != track.mediaType {
for _, track := range t.RtspTracks {
if utils.AVMediaTypeVideo != track.MediaType {
continue
}
// 回滚序号和时间戳
index := int(track.seq) - len(track.extraDataBuffer)
for i, bytes := range track.extraDataBuffer {
index := int(track.StartSeq) - len(track.ExtraDataBuffer)
for i, bytes := range track.ExtraDataBuffer {
librtp.RollbackSeq(bytes[OverTcpHeaderSize:], index+i+1)
binary.BigEndian.PutUint32(bytes[OverTcpHeaderSize+4:], uint32(ts))
}
return track.extraDataBuffer, ts, nil
return track.ExtraDataBuffer, ts, nil
}
return nil, ts, nil
}
func (t *TranStream) PackRtpPayload(muxer librtp.Muxer, channel int, data []byte, timestamp uint32) {
func (t *TransStream) PackRtpPayload(track *Track, channel int, data []byte, timestamp uint32) {
var index int
muxer.Input(data, timestamp, func() []byte {
// 保存开始序号
track.StartSeq = track.Muxer.GetHeader().Seq
track.Muxer.Input(data, timestamp, func() []byte {
index = t.buffer.Index()
block := t.buffer.GetBlock()
return block[OverTcpHeaderSize:]
}, func(bytes []byte) {
track.EndSeq = track.Muxer.GetHeader().Seq
packet := t.buffer.Get(index)[:OverTcpHeaderSize+len(bytes)]
t.OverTCP(packet, channel)
t.AppendOutStreamBuffer(packet)
})
}
func (t *TranStream) AddTrack(stream utils.AVStream) error {
if err := t.BaseTransStream.AddTrack(stream); err != nil {
func (t *TransStream) AddTrack(track *stream.Track) error {
if err := t.BaseTransStream.AddTrack(track); err != nil {
return err
}
payloadType, ok := librtp.CodecIdPayloads[stream.CodecId()]
payloadType, ok := librtp.CodecIdPayloads[track.Stream.CodecId()]
if !ok {
return fmt.Errorf("no payload type was found for codecid: %d", 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
var startSeq uint16
if t.oldTracks != nil {
startSeq, ok = t.oldTracks[byte(payloadType.Pt)]
utils.Assert(ok)
}
// 创建RTP封装器
var muxer librtp.Muxer
if utils.AVCodecIdH264 == stream.CodecId() {
muxer = librtp.NewH264Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdH265 == stream.CodecId() {
muxer = librtp.NewH265Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdAAC == stream.CodecId() {
muxer = librtp.NewAACMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdPCMALAW == stream.CodecId() || utils.AVCodecIdPCMMULAW == stream.CodecId() {
muxer = librtp.NewMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
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)
}
t.rtpTracks = append(t.rtpTracks, NewRTSPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, stream.Type()))
index := len(t.rtpTracks) - 1
rtspTrack := NewRTSPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, track.Stream.Type())
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 == stream.Type() {
parameters := stream.CodecParameters()
if utils.AVMediaTypeVideo == track.Stream.Type() {
parameters := track.Stream.CodecParameters()
if utils.AVCodecIdH265 == stream.CodecId() {
if utils.AVCodecIdH265 == track.Stream.CodecId() {
bytes := parameters.(*utils.HEVCCodecData).VPS()
t.PackRtpPayload(muxer, index, bytes[0], 0)
t.PackRtpPayload(rtspTrack, index, bytes[0], 0)
}
spsBytes := parameters.SPS()
ppsBytes := parameters.PPS()
t.PackRtpPayload(muxer, index, spsBytes[0], 0)
t.PackRtpPayload(muxer, index, ppsBytes[0], 0)
t.PackRtpPayload(rtspTrack, index, spsBytes[0], 0)
t.PackRtpPayload(rtspTrack, index, ppsBytes[0], 0)
// 拷贝扩展数据的rtp包
size := t.buffer.Index() - bufferIndex
@@ -137,14 +151,14 @@ func (t *TranStream) AddTrack(stream utils.AVStream) error {
extraRtpBuffer[i] = dst[:OverTcpHeaderSize+binary.BigEndian.Uint16(dst[2:])]
}
t.rtpTracks[index].extraDataBuffer = extraRtpBuffer
t.RtspTracks[index].ExtraDataBuffer = extraRtpBuffer
}
return nil
}
func (t *TranStream) Close() ([][]byte, int64, error) {
for _, track := range t.rtpTracks {
func (t *TransStream) Close() ([][]byte, int64, error) {
for _, track := range t.RtspTracks {
if track != nil {
track.Close()
}
@@ -153,7 +167,7 @@ func (t *TranStream) Close() ([][]byte, int64, error) {
return nil, 0, nil
}
func (t *TranStream) WriteHeader() error {
func (t *TransStream) WriteHeader() error {
description := sdp.SessionDescription{
Version: 0,
Origin: sdp.Origin{
@@ -177,7 +191,7 @@ func (t *TranStream) WriteHeader() error {
}
for i, track := range t.Tracks {
payloadType, _ := librtp.CodecIdPayloads[track.CodecId()]
payloadType, _ := librtp.CodecIdPayloads[track.Stream.CodecId()]
mediaDescription := sdp.MediaDescription{
ConnectionInformation: &sdp.ConnectionInformation{
NetworkType: "IN",
@@ -195,10 +209,10 @@ func (t *TranStream) WriteHeader() error {
mediaDescription.MediaName.Protos = []string{"RTP", "AVP"}
mediaDescription.MediaName.Formats = []string{strconv.Itoa(payloadType.Pt)}
if utils.AVMediaTypeAudio == track.Type() {
if utils.AVMediaTypeAudio == track.Stream.Type() {
mediaDescription.MediaName.Media = "audio"
if utils.AVCodecIdAAC == track.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
@@ -237,12 +251,13 @@ func (t *TranStream) WriteHeader() error {
return nil
}
func NewTransStream(addr net.IPAddr, urlFormat string) stream.TransStream {
t := &TranStream{
func NewTransStream(addr net.IPAddr, urlFormat string, oldTracks map[byte]uint16) stream.TransStream {
t := &TransStream{
addr: addr,
urlFormat: urlFormat,
// 在将AVPacket打包rtp时, 会使用多个buffer块, 回环覆盖多个rtp块, 如果是TCP拉流并且网络不好, 推流的数据会错乱.
buffer: stream.NewReceiveBuffer(1500, 1024),
buffer: stream.NewReceiveBuffer(1500, 1024),
oldTracks: oldTracks,
}
if addr.IP.To4() != nil {
@@ -254,10 +269,15 @@ func NewTransStream(addr net.IPAddr, urlFormat string) stream.TransStream {
return t
}
func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, streams []utils.AVStream) (stream.TransStream, error) {
func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
trackFormat := "?track=%d"
var oldTracks map[byte]uint16
if endInfo := source.GetStreamEndInfo(); endInfo != nil {
oldTracks = endInfo.RtspTracks
}
return NewTransStream(net.IPAddr{
IP: net.ParseIP(stream.AppConfig.PublicIP),
Zone: "",
}, trackFormat), nil
}, trackFormat, oldTracks), nil
}

21
rtsp/rtsp_track.go
View File

@@ -7,13 +7,14 @@ import (
// Track RtspTrack 对rtsp每路输出流的封装
type Track struct {
pt byte
rate int
mediaType utils.AVMediaType
seq uint16
PT byte
Rate int
MediaType utils.AVMediaType
StartSeq uint16
EndSeq uint16
muxer librtp.Muxer
extraDataBuffer [][]byte // 缓存带有编码信息的rtp包, 对所有sink通用
Muxer librtp.Muxer
ExtraDataBuffer [][]byte // 缓存带有编码信息的rtp包, 对所有sink通用
}
func (r *Track) Close() {
@@ -21,10 +22,10 @@ func (r *Track) Close() {
func NewRTSPTrack(muxer librtp.Muxer, pt byte, rate int, mediaType utils.AVMediaType) *Track {
stream := &Track{
pt: pt,
rate: rate,
muxer: muxer,
mediaType: mediaType,
PT: pt,
Rate: rate,
Muxer: muxer,
MediaType: mediaType,
}
return stream