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client.go
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client.go
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package rtsp
import (
"context"
"net"
"github.com/bluenviron/gortsplib/v4"
"github.com/bluenviron/gortsplib/v4/pkg/base"
"github.com/bluenviron/gortsplib/v4/pkg/url"
"go.uber.org/zap"
"m7s.live/engine/v4"
)
type RTSPClient struct {
*gortsplib.Client `json:"-" yaml:"-"`
gortsplib.Transport
DialContext func(ctx context.Context, network, address string) (net.Conn, error) `json:"-" yaml:"-"`
}
type RTSPPuller struct {
RTSPPublisher
engine.Puller
RTSPClient
}
func (p *RTSPClient) Close() error {
if p.Client != nil {
p.Client.Close()
}
return nil
}
func (p *RTSPClient) Disconnect() {
if p.Client != nil {
p.Client.Close()
}
}
func (p *RTSPPuller) Connect() error {
client := &gortsplib.Client{
DialContext: p.DialContext,
AnyPortEnable: true,
}
p.Transport = gortsplib.TransportTCP
client.Transport = &p.Transport
// parse URL
u, err := url.Parse(p.RemoteURL)
if err != nil {
return err
}
// connect to the server
if err = client.Start(u.Scheme, u.Host); err != nil {
return err
}
p.Client = client
p.SetIO(p)
return nil
}
func (p *RTSPPuller) Pull() (err error) {
u, _ := url.Parse(p.RemoteURL)
var res *base.Response
if rtspConfig.SendOptions {
if res, err = p.Options(u); err != nil {
p.Error("Options", zap.Error(err))
return
}
}
p.Debug("Options", zap.Any("res", res))
// find published tracks
session, res, err := p.Describe(u)
if err != nil {
p.Error("Describe", zap.Error(err))
return err
}
p.Debug("Describe", zap.Any("res", res))
p.session = session
err = p.SetTracks()
if err != nil {
p.Error("SetTracks", zap.Error(err))
return err
}
if err = p.SetupAll(session.BaseURL, session.Medias); err != nil {
p.Error("SetupAndPlay", zap.Error(err))
return err
}
p.OnPacketRTPAny(p.OnPacket)
res, err = p.Play(nil)
p.Debug("Play", zap.Any("res", res))
if err != nil {
p.Error("Play", zap.Error(err))
return err
}
return p.Wait()
}
type RTSPPusher struct {
RTSPSubscriber
engine.Pusher
RTSPClient
}
func (p *RTSPPusher) OnEvent(event any) {
switch v := event.(type) {
case engine.VideoRTP:
p.Client.WritePacketRTP(p.videoTrack, v.Packet)
case engine.AudioRTP:
p.Client.WritePacketRTP(p.audioTrack, v.Packet)
default:
p.RTSPSubscriber.OnEvent(event)
}
}
func (p *RTSPPusher) Connect() error {
p.Client = &gortsplib.Client{
DialContext: p.DialContext,
WriteQueueSize: rtspConfig.WriteBufferCount,
}
p.Transport = gortsplib.TransportTCP
p.Client.Transport = &p.Transport
// parse URL
u, err := url.Parse(p.RemoteURL)
if err != nil {
p.Error("url.Parse", zap.Error(err))
return err
}
// connect to the server
if err = p.Client.Start(u.Scheme, u.Host); err != nil {
p.Error("Client.Start", zap.Error(err))
return err
}
p.SetIO(p)
if rtspConfig.SendOptions {
_, err = p.Client.Options(u)
}
return err
}
func (p *RTSPPusher) Push() (err error) {
var u *url.URL
u, err = url.Parse(p.RemoteURL)
// startTime := time.Now()
// for len(p.tracks) < 2 {
// if time.Sleep(time.Second); time.Since(startTime) > time.Second*10 {
// return fmt.Errorf("timeout")
// }
// }
var res *base.Response
if res, err = p.Announce(u, p.session); err != nil {
p.Error("Announce", zap.Error(err))
return
} else {
p.Debug("Announce", zap.Any("res", res))
}
err = p.SetupAll(u, p.session.Medias)
if err != nil {
p.Error("Setup", zap.Error(err))
return
}
if res, err = p.Record(); err != nil {
p.Error("Record", zap.Error(err))
return
} else {
p.Debug("Record", zap.Any("res", res))
}
p.PlayRTP()
return
}