2024年pion实现录制WebRTC流_pion webrtc 哪有教程(1)，2024年最新最新整理

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savePath      string
ctx           context.Context
cancel        context.CancelFunc
pc            \*webrtc.PeerConnection
hasAudioTrack bool
hasVideoTrack bool
videoFinish   chan struct{}
audioFinish   chan struct{}

}

func (self *Stream) onTrack(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) error {
// Send a PLI on an interval so that the publisher is pushing a keyframe
codec := track.Codec()

trackDesc := fmt.Sprintf("channels=%v", codec.Channels)
if track.Kind() == webrtc.RTPCodecTypeVideo {
	trackDesc = fmt.Sprintf("fmtp=%v", codec.SDPFmtpLine)
}
logrus.Infof("Got track %v, pt=%v tbn=%v, %v", codec.MimeType, codec.PayloadType, codec.ClockRate, trackDesc)
var err error
if codec.MimeType == "audio/opus" {
	var da media.Writer
	defer func() {
		if da != nil {
			da.Close()
		}
	}()
	audiopath := self.savePath + self.Display + "\_audio.ogg"
	if da, err = oggwriter.New(audiopath, codec.ClockRate, codec.Channels); err != nil {
		return errors.Wrapf(err, "创建"+audiopath+"失败")
	}
	self.hasAudioTrack = true
	logrus.Infof("Open ogg writer file=%v , tbn=%v, channels=%v", audiopath, codec.ClockRate, codec.Channels)
	if err = self.writeTrackToDisk(da, track); err != nil {
		return err
	}
	self.audioFinish <- struct{}{}
} else if codec.MimeType == "video/H264" {
	var dv_h264 media.Writer
	videopath := self.savePath + self.Display + "\_video.h264"

	if dv_h264, err = h264writer.New(videopath); err != nil {
		return err
	}
	logrus.Infof("Open h264 writer file=%v", videopath)
	self.hasVideoTrack = true
	if err = self.writeTrackToDisk(dv_h264, track); err != nil {
		return err
	}
	self.audioFinish <- struct{}{}
} else {
	logrus.Warnf("Ignore track %v pt=%v", codec.MimeType, codec.PayloadType)
}

return nil

}

func (self *Stream) writeTrackToDisk(w media.Writer, track *webrtc.TrackRemote) error {
for self.ctx.Err() == nil {
pkt, _, err := track.ReadRTP()
//fmt.Println(filename, pkt.Timestamp)
if err != nil {
if self.ctx.Err() != nil {
return nil
}
return err
}

	if w == nil {
		continue
	}

	if err := w.WriteRTP(pkt); err != nil {
		if len(pkt.Payload) <= 2 {
			continue
		}
		logrus.Warnf("Ignore write RTP %vB err %+v\n", len(pkt.Payload), err)
	}
}

return self.ctx.Err()

}

func (self *Stream) Stop() bool {
self.cancel()
if self.hasAudioTrack {
<-self.audioFinish
}
if self.hasVideoTrack {
<-self.videoFinish
}

if self.hasVideoTrack && self.hasAudioTrack {
	audiopath := self.savePath + self.Display + "\_audio.ogg"
	videopath := self.savePath + self.Display + "\_video.h264"

	cmd := exec.Command("ffmpeg",
		"-i",
		audiopath,
		"-i",
		videopath,
		self.savePath+self.Display+".ts",
		"-y")
	if err := cmd.Run(); err != nil {
		logrus.Errorf("拼接音频和视频失败:%v", err)
		return false
	}
	return true
}
return false

}

func NewStream(host, room, display, savePath string) (*Stream, error) {
var err error
stream := &Stream{
Host: host,
Room: room,
Display: display,
rtcUrl: “webrtc://” + host + “/” + room + “/” + display,
savePath: savePath,
hasAudioTrack: false,
hasVideoTrack: false,
videoFinish: make(chan struct{}, 1),
audioFinish: make(chan struct{}, 1),
}
stream.ctx, stream.cancel = context.WithCancel(context.Background())

//创建PeerConncetion
stream.pc, err = newPeerConnection(webrtc.Configuration{})
if err != nil {
	return nil, errors.Wrapf(err, "创建PeerConnection失败")
}

//设置方向
stream.pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, webrtc.RTPTransceiverInit{
	Direction: webrtc.RTPTransceiverDirectionRecvonly,
})
stream.pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RTPTransceiverInit{
	Direction: webrtc.RTPTransceiverDirectionRecvonly,
})

//创建offer
offer, err := stream.pc.CreateOffer(nil)
if err != nil {
	return nil, errors.Wrap(err, "创建Local offer失败")
}

// 设置本地sdp
if err = stream.pc.SetLocalDescription(offer); err != nil {
	return nil, errors.Wrap(err, "设置Local SDP失败")
}

// 设置远端SDP
answer, err := apiRtcRequest(stream.ctx, "/rtc/v1/play", stream.rtcUrl, offer.SDP)
if err != nil {
	return nil, errors.Wrap(err, "SDP协商失败")
}

if err = stream.pc.SetRemoteDescription(webrtc.SessionDescription{
	Type: webrtc.SDPTypeAnswer, SDP: answer,
}); err != nil {
	return nil, errors.Wrap(err, "设置Remote SDP失败")
}

stream.pc.OnTrack(func(track \*webrtc.TrackRemote, receiver \*webrtc.RTPReceiver) {
	fmt.Println("------------------")
	err = stream.onTrack(track, receiver)
	if err != nil {
		codec := track.Codec()
		logrus.Errorf("Handle track %v, pt=%v\nerr %v", codec.MimeType, codec.PayloadType, err)
		stream.cancel()
	}
	stream.pc.Close()
})

stream.pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
	logrus.Infof("ICE state %v", state)

	if state == webrtc.ICEConnectionStateFailed || state == webrtc.ICEConnectionStateClosed {
		if stream.ctx.Err() != nil {
			return
		}

		logrus.Warnf("Close for ICE state %v", state)
		stream.cancel()
		stream.pc.Close()
	}
})
key := "webrtc://" + host + "/" + room
Streams.Store(key, stream)
return stream, nil

}

func newPeerConnection(configuration webrtc.Configuration) (*webrtc.PeerConnection, error) {
m := &webrtc.MediaEngine{}
if err := m.RegisterDefaultCodecs(); err != nil {
return nil, err
}

for _, extension := range []string{sdp.SDESMidURI, sdp.SDESRTPStreamIDURI, sdp.TransportCCURI} {
	if extension == sdp.TransportCCURI {
		continue
	}
	if err := m.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{URI: extension}, webrtc.RTPCodecTypeVideo); err != nil {
		return nil, err
	}
}

// https://github.com/pion/ion/issues/130
// https://github.com/pion/ion-sfu/pull/373/files#diff-6f42c5ac6f8192dd03e5a17e9d109e90cb76b1a4a7973be6ce44a89ffd1b5d18R73
for _, extension := range []string{sdp.SDESMidURI, sdp.SDESRTPStreamIDURI, sdp.AudioLevelURI} {
	if extension == sdp.AudioLevelURI {
		continue
	}
	if err := m.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{URI: extension}, webrtc.RTPCodecTypeAudio); err != nil {
		return nil, err
	}
}

i := &interceptor.Registry{}
if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
	return nil, err
}

api := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i))
return api.NewPeerConnection(configuration)

}

api.go

package stream

import (
“context”
“encoding/json”
“fmt”
“github.com/pkg/errors”
“github.com/sirupsen/logrus”
“io/ioutil”
“net/http”
“net/url”
“strings”
)

// Request SRS RTC API, the apiPath like “/rtc/v1/play”, the r is WebRTC url like
// “webrtc://localhost/live/livestream”, and the offer is SDP in string.
//
// Return the response of answer SDP in string.
func apiRtcRequest(ctx context.Context, apiPath, rtcurl, offer string) (string, error) {
u, err := url.Parse(rtcurl)
if err != nil {
return “”, errors.Wrapf(err, “Parse url %v”, rtcurl)
}

// Build api url.
host := u.Host
if !strings.Contains(host, ":") {
	host += ":1985"
}

api := fmt.Sprintf("http://%v", host)
if !strings.HasPrefix(apiPath, "/") {
	api += "/"
}
api += apiPath

if !strings.HasSuffix(apiPath, "/") {
	api += "/"

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[外链图片转存中…(img-AIXVAkYF-1715656625603)]
[外链图片转存中…(img-pPIzA0vT-1715656625603)]

既有适合小白学习的零基础资料，也有适合3年以上经验的小伙伴深入学习提升的进阶课程，涵盖了95%以上Go语言开发知识点，真正体系化！

由于文件比较多，这里只是将部分目录截图出来，全套包含大厂面经、学习笔记、源码讲义、实战项目、大纲路线、讲解视频，并且后续会持续更新

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