pion学习总结：等待传出track的一般流程
Howard Yin 2021-09-02 14:20:03 WebRTC编程框架pion实操
# 先创建设置然后创建PeerConnection：
和传入连接一样的过程，见《pion学习总结：等待传入track的一般流程》
# 创建track
这一步主要是要生成一个TrackLocal子类，在里面指定你要给远端发什么东西以及怎么发。
比如在用实例学习pion - play-from-disk里面用的是pion给的样例track：
// Create a video track
videoTrack, videoTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion")
if videoTrackErr != nil {
panic(videoTrackErr)
}
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再比如在用实例学习pion - rtp-to-webrtc里面用的是pion内置的基础RTP媒体track：
// Create a video track
videoTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion")
if err != nil {
panic(err)
}
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# 将track添加到PeerConnection
就是一个AddTrack函数，不必多讲：
rtpSender, err := peerConnection.AddTrack(videoTrack)
if err != nil {
panic(err)
}
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# 等待连接传入
# 开始往track里写数据
有连接连上了就可以通过track向对方发数据了，也就是往track里写数据。往track里写数据的操作是用户自己定义的，其归根结底就是不断调用TrackLocalContext里的TrackLocalWriter.WriteRTP（这个TrackLocalContext来自于pion里调用用户实现的TrackLocal.Bind，具体情况可参见《pion中的TrackLocal》）。
比如在《用实例学习pion - play-from-disk》里，调用TrackLocalWriter.WriteRTP的过程就封装在WriteSample里，读取视频后这边只要没隔一帧的时间发一次数据即可：
// Keep track of last granule, the difference is the amount of samples in the buffer
var lastGranule uint64
// It is important to use a time.Ticker instead of time.Sleep because
// * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
// * works around latency issues with Sleep (see https://github.com/golang/go/issues/44343)
ticker := time.NewTicker(oggPageDuration)
for ; true; <-ticker.C {
pageData, pageHeader, oggErr := ogg.ParseNextPage()
if oggErr == io.EOF {
fmt.Printf("All audio pages parsed and sent")
os.Exit(0)
}
if oggErr != nil {
panic(oggErr)
}
// The amount of samples is the difference between the last and current timestamp
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
lastGranule = pageHeader.GranulePosition
sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond
if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration}); oggErr != nil {
panic(oggErr)
}
}
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再比如在《用实例学习pion - rtp-to-webrtc》里，track用的是pion内置的TrackLocalStaticRTP：
// Read RTP packets forever and send them to the WebRTC Client
inboundRTPPacket := make([]byte, 1600) // UDP MTU
for {
n, _, err := listener.ReadFrom(inboundRTPPacket)
if err != nil {
panic(fmt.Sprintf("error during read: %s", err))
}
if _, err = videoTrack.Write(inboundRTPPacket[:n]); err != nil {
if errors.Is(err, io.ErrClosedPipe) {
// The peerConnection has been closed.
return
}
panic(err)
}
}
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TrackLocalStaticRTP是一种编解码方式不变的TrackLocal，调用TrackLocalWriter.WriteRTP的过程就在TrackLocalStaticRTP.WriteRTP里。关于这里用的TrackLocalStaticRTP的更多介绍可以参见《pion中的TrackLocal》。