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rtc支持单端口

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This commit is contained in:
yangjiechina
2024-07-13 13:53:26 +08:00
parent 0114a36249
commit 27da874258
7 changed files with 114 additions and 40 deletions
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1
main.go
View File

@@ -35,6 +35,7 @@ func init() {
stream.SetDefaultConfig(config)
stream.AppConfig = *config
stream.InitHookUrl()
rtc.InitConfig()
//初始化日志
log.InitLogger(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)

2
rtc.html
View File

@@ -32,7 +32,7 @@
<body>
<div style="margin-top: 10px; margin-left: 10px;">
<input style="width: 100px;" id="url" type="text" value="live/rtc/_mystream"/>
<input style="width: 100px;" id="url" type="text" value="hls/mystream.rtc"/>
<button onclick="play()"> 播放</button>
</div>

121
rtc/rtc_stream.go
View File

@@ -1,35 +1,33 @@
package rtc
import (
"fmt"
"github.com/lkmio/avformat/utils"
"github.com/lkmio/lkm/log"
"github.com/lkmio/lkm/stream"
"github.com/pion/interceptor"
"github.com/pion/webrtc/v3"
"net"
)
var (
webrtcApi *webrtc.API
)
type transStream struct {
stream.BaseTransStream
}
func NewTransStream() stream.TransStream {
t := &transStream{}
return t
}
func TransStreamFactory(source stream.Source, protocol stream.Protocol, streams []utils.AVStream) (stream.TransStream, error) {
return NewTransStream(), nil
}
func (t *transStream) Input(packet utils.AVPacket) error {
if utils.AVMediaTypeAudio == packet.MediaType() {
} else if utils.AVMediaTypeVideo == packet.MediaType() {
for _, iSink := range t.Sinks {
sink_ := iSink.(*sink)
if sink_.state < webrtc.ICEConnectionStateConnected {
continue
}
for _, iSink := range t.Sinks {
sink_ := iSink.(*sink)
if sink_.state < webrtc.ICEConnectionStateConnected {
continue
}
if utils.AVMediaTypeAudio == packet.MediaType() {
sink_.input(packet.Index(), packet.Data(), uint32(packet.Duration(1000)))
} else if utils.AVMediaTypeVideo == packet.MediaType() {
if packet.KeyFrame() {
extra := t.BaseTransStream.Tracks[packet.Index()].CodecParameters().AnnexBExtraData()
sink_.input(packet.Index(), extra, 0)
@@ -47,34 +45,56 @@ func (t *transStream) AddSink(sink_ stream.Sink) error {
var videoTrack *webrtc.TrackLocalStaticSample
rtcSink := sink_.(*sink)
rtcSink.setTrackCount(len(t.Tracks))
connection, err := webrtc.NewPeerConnection(webrtc.Configuration{})
connection, err := webrtcApi.NewPeerConnection(webrtc.Configuration{})
connection.OnICECandidate(func(candidate *webrtc.ICECandidate) {
})
for index, track := range t.Tracks {
if utils.AVCodecIdH264 != track.CodecId() {
var mimeType string
var id string
if utils.AVCodecIdH264 == track.CodecId() {
mimeType = webrtc.MimeTypeH264
} else if utils.AVCodecIdH265 == track.CodecId() {
mimeType = webrtc.MimeTypeH265
} else if utils.AVCodecIdAV1 == track.CodecId() {
mimeType = webrtc.MimeTypeAV1
} else if utils.AVCodecIdVP8 == track.CodecId() {
mimeType = webrtc.MimeTypeVP8
} else if utils.AVCodecIdVP9 == track.CodecId() {
mimeType = webrtc.MimeTypeVP9
} else if utils.AVCodecIdOPUS == track.CodecId() {
mimeType = webrtc.MimeTypeOpus
} else if utils.AVCodecIdPCMALAW == track.CodecId() {
mimeType = webrtc.MimeTypePCMA
} else if utils.AVCodecIdPCMMULAW == track.CodecId() {
mimeType = webrtc.MimeTypePCMU
} else {
log.Sugar.Errorf("codec %d not compatible with webrtc", track.CodecId())
continue
}
videoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, "video", "pion")
if utils.AVMediaTypeAudio == track.Type() {
id = "audio"
} else {
id = "video"
}
videoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: mimeType}, id, "pion")
if err != nil {
panic(err)
}
if _, err := connection.AddTransceiverFromTrack(videoTrack, webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly}); err != nil {
} else if _, err := connection.AddTransceiverFromTrack(videoTrack, webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly}); err != nil {
return err
}
if _, err = connection.AddTrack(videoTrack); err != nil {
} else if _, err = connection.AddTrack(videoTrack); err != nil {
return err
}
rtcSink.addTrack(index, videoTrack)
}
if err = connection.SetRemoteDescription(webrtc.SessionDescription{Type: webrtc.SDPTypeOffer, SDP: rtcSink.offer}); err != nil {
if len(connection.GetTransceivers()) == 0 {
return fmt.Errorf("no track added")
} else if err = connection.SetRemoteDescription(webrtc.SessionDescription{Type: webrtc.SDPTypeOffer, SDP: rtcSink.offer}); err != nil {
return err
}
@@ -85,8 +105,8 @@ func (t *transStream) AddSink(sink_ stream.Sink) error {
} else if err = connection.SetLocalDescription(answer); err != nil {
return err
}
<-complete
<-complete
connection.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
rtcSink.state = state
if webrtc.ICEConnectionStateDisconnected > state {
@@ -102,3 +122,44 @@ func (t *transStream) AddSink(sink_ stream.Sink) error {
func (t *transStream) WriteHeader() error {
return nil
}
func NewTransStream() stream.TransStream {
t := &transStream{}
return t
}
func InitConfig() {
setting := webrtc.SettingEngine{}
var ips []string
ips = append(ips, stream.AppConfig.PublicIP)
udpListener, err := net.ListenUDP("udp", &net.UDPAddr{
IP: net.ParseIP(stream.AppConfig.ListenIP),
Port: stream.AppConfig.WebRtc.Port,
})
if err != nil {
panic(err)
}
//设置公网ip和监听端口
setting.SetICEUDPMux(webrtc.NewICEUDPMux(nil, udpListener))
setting.SetNAT1To1IPs(ips, webrtc.ICECandidateTypeHost)
//注册音视频编码器
m := &webrtc.MediaEngine{}
if err := m.RegisterDefaultCodecs(); err != nil {
panic(err)
}
i := &interceptor.Registry{}
if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
panic(err)
}
webrtcApi = webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i), webrtc.WithSettingEngine(setting))
}
func TransStreamFactory(source stream.Source, protocol stream.Protocol, streams []utils.AVStream) (stream.TransStream, error) {
return NewTransStream(), nil
}

6
stream/config.go
View File

@@ -69,6 +69,11 @@ type GB28181Config struct {
Port []int `json:"port"`
}
type WebRtcConfig struct {
TransportConfig
Port int `json:"port"`
}
func (g TransportConfig) IsEnableTCP() bool {
return strings.Contains(g.Transport, "TCP")
}
@@ -244,6 +249,7 @@ type AppConfig_ struct {
JT1078 JT1078Config
Rtsp RtspConfig
GB28181 GB28181Config
WebRtc WebRtcConfig
Hook HookConfig
Record RecordConfig

2
stream/hook_sink.go
View File

@@ -28,7 +28,7 @@ func PreparePlaySink(sink Sink) (*http.Response, utils.HookState) {
sink.Lock()
defer sink.UnLock()
if SessionStateClose == sink.State() {
if SessionStateClosed == sink.State() {
log.Sugar.Warnf("添加到sink到等待队列失败, sink已经断开链接 %s", sink.Id())
return response, utils.HookStateFailure
} else {

14
stream/sink.go
View File

@@ -72,7 +72,11 @@ type Sink interface {
func GenerateSinkId(addr net.Addr) SinkId {
network := addr.Network()
if "tcp" == network {
id := uint64(binary.BigEndian.Uint32(addr.(*net.TCPAddr).IP.To4()))
to4 := addr.(*net.TCPAddr).IP.To4()
if to4 == nil {
to4 = make([]byte, 4)
}
id := uint64(binary.BigEndian.Uint32(to4))
id <<= 32
id |= uint64(addr.(*net.TCPAddr).Port << 16)
@@ -186,7 +190,9 @@ func (s *BaseSink) DesiredVideoCodecId() utils.AVCodecID {
// 拉流断开连接,不需要考虑线程安全
// 踢流走source管道删除,并且关闭Conn
func (s *BaseSink) Close() {
utils.Assert(SessionStateClose != s.State_)
if SessionStateClosed != s.State_ {
return
}
if s.Conn != nil {
s.Conn.Close()
@@ -202,12 +208,12 @@ func (s *BaseSink) Close() {
{
s.Lock()
defer s.UnLock()
if s.State_ == SessionStateClose {
if s.State_ == SessionStateClosed {
return
}
state = s.State_
s.State_ = SessionStateClose
s.State_ = SessionStateClosed
}
if state == SessionStateTransferring {

8
stream/source.go
View File

@@ -50,7 +50,7 @@ const (
SessionStateHandshakeDone = SessionState(4) //握手完成
SessionStateWait = SessionState(5) //位于等待队列中
SessionStateTransferring = SessionState(6) //推拉流中
SessionStateClose = SessionState(7) //关闭状态
SessionStateClosed = SessionState(7) //关闭状态
)
// Source 父类Source负责, 除解析流以外的所有事情
@@ -407,7 +407,7 @@ func (s *PublishSource) AddSink(sink Sink) bool {
sink.Lock()
defer sink.UnLock()
if SessionStateClose == sink.State() {
if SessionStateClosed == sink.State() {
log.Sugar.Warnf("AddSink失败, sink已经断开链接 %s", sink.PrintInfo())
} else {
transStream.AddSink(sink)
@@ -519,7 +519,7 @@ func (s *PublishSource) doClose() {
sink.Lock()
defer sink.UnLock()
if SessionStateClose == sink.State() {
if SessionStateClosed == sink.State() {
log.Sugar.Warnf("添加到sink到等待队列失败, sink已经断开链接 %s", sink.PrintInfo())
} else {
sink.SetState(SessionStateWait)
@@ -527,7 +527,7 @@ func (s *PublishSource) doClose() {
}
}
if SessionStateClose != sink.State() {
if SessionStateClosed != sink.State() {
sink.Flush()
}
})