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refactor: 拆分avformat依赖库

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This commit is contained in:
ydajiang
2025-04-08 09:23:18 +08:00
parent 611812da4c
commit a508ef2838
52 changed files with 1374 additions and 1154 deletions
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3
rtsp/rtsp_handler.go
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@@ -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{

3
rtsp/rtsp_server.go
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@@ -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"
)

1
rtsp/rtsp_session.go
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@@ -1,5 +1,6 @@
package rtsp
import (
"bufio"
"bytes"

16
rtsp/rtsp_sink.go
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@@ -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"
@@ -21,8 +21,8 @@ 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
cb func(sdp string) // sdp回调, 响应describe
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
}
func (s *Sink) StartStreaming(transStream stream.TransStream) error {
@@ -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

71
rtsp/rtsp_stream.go
View File

@@ -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

7
rtsp/rtsp_track.go
View File

@@ -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,