随着互联网技术的不断发展和网络应用的普及,视频流媒体技术逐渐受到广泛关注。在视频流媒体技术中,RTSP是一种常用的流媒体传输协议,它可以使用户在网络上实现音视频数据的传输和播放。但是,由于RTSP协议在传输过程中采用的是TCP协议,这样就会导致传输速度比较慢,而且占用带宽比较高,这对于用户的视频观看体验会带来一定影响。为了解决这个问题,我们可以采用将RTSP流转成HTTP流的方法,这样会更加节约带宽,提高传输速度。本文将介绍如何使用golang来实现rtsp转http的过程。
一、使用golang实现RTSP转HTTP
在进行RTSP转HTTP的实现之前,我们需要先了解一下RTSP与HTTP之间的区别。RTSP协议需要实现流媒体传输协议的控制部分,但HTTP协议则是实现静态的页面传输和数据传递。因此,如果想要将RTSP流转成HTTP流,必须在其中增加一个中间层,来实现控制部分与HTTP静态页面的传输。
Golang是一种高效、简单、稳定且高度并发的编程语言,非常适合用于处理实时音视频流数据。因此,我们选择用golang来实现RTSP转HTTP的过程。
首先,我们需要使用golang中的库来实现RTSP协议的解析和处理。在golang中,有一个第三方库叫做"gosip",它实现了RTSP协议的解析和处理。我们可以使用该库来实现RTSP转HTTP的过程。另外,我们还需要用到golang中的HTTP库来实现HTTP协议的传输。
具体实现步骤如下:
import ( "github.com/gorilla/mux" "github.com/nareix/srtcp" "github.com/nareix/udp" "github.com/nareix/webrtc" "github.com/nareix/xtcp" "github.com/nareix/joy4/format" "github.com/nareix/joy4/format/ts" "github.com/nareix/joy4/av" "github.com/nareix/joy4/container/rtp" "github.com/nareix/joy4/av/pubsub" "github.com/nareix/joy4/cgo/ffmpeg" "github.com/nareix/joy4/av/pktque" "net/http" "io" "fmt" "bytes" "strconv" ) ... // 使用gosip库解析RTSP请求 func processRTSP(rtspRequest io.ReadWriteCloser, pubsub1 *pubsub.PubSub, aacWrite io.WriteCloser, h264Write io.WriteCloser) { sessionHandle := func(s *rtsp.Session) { p, err := s.Streams() checkError(err) var vtrack av.CodecData var atracks []av.CodecData for _, vi := range p { switch vi.Type().(type) { case av.H264CodecData: vtrack = vi.(av.H264CodecData) break case av.AACCodecData: atracks = append(atracks, vi.(av.AACCodecData)) break } } var streamDialers []av.MuxCloser var streamWriters []av.MuxCloser // 创建H264的PubSub并添加到H264 Pub集 H264Pub := pubsub.NewSimpleMuxer(100) streamDialers = append(streamDialers, H264Pub) go func() { H264Out := <-H264Pub.Out H264Outs, err := rtp.Encode(H264Out, vtrack.(av.VideoCodecData).Sdp()) checkError(err) defer H264Outs.Close() n, err := s.WriteInterleaved(H264Outs) checkError(err) fmt.Println("Sent", n, "bytes. H264") }() // 创建AAC的PubSub并添加到AAC Pub集 AACPubs := make([]*pubsub.PubSub, len(atracks)) for i, atrack := range atracks { AACPubs[i] = pubsub.NewSimpleMuxer(100) streamDialers = append(streamDialers, AACPubs[i]) go func(atrack av.CodecData, AACPubs1 *pubsub.PubSub) { out := <-AACPubs1.Out aacOut, _ := atrack.NewMuxer(out) defer aacOut.Close() outs, err := rtp.Encode(aacOut, atrack.(av.AudioCodecData).Sdp()) checkError(err) defer outs.Close() n, err := s.WriteInterleaved(outs) checkError(err) fmt.Println("Sent", n, "bytes. Audio") }(atrack, AACPubs[i]) } // 打开相应的转换器 if aacWrite != nil { streamWriters = append(streamWriters, aacWrite) pubAACOut := make(chan []byte) AACPubs[0].Out <- pubAACOut go func() { // 把音频包推送到channel,再写到文件 for { samples := <-pubAACOut _, err := aacWrite.Write(samples) checkError(err) } }() } if h264Write != nil { streamWriters = append(streamWriters, h264Write) H264Pub.Out <- h264Write } // 等待停止 <-s.Done() for _, dialer := range streamDialers { fmt.Println("Closing dialer") dialer.Close() } for _, writer := range streamWriters { fmt.Println("Closing writer") writer.Close() } } for { req, err := rtsp.NewRequest() checkError(err) s, err := rtsp.NewSession(req, rtspRequest) if err != nil { fmt.Println(err) break } sessionHandle(s) } }
... // 使用HTTP协议请求推送的HLS流 func processHTTP(w http.ResponseWriter, r *http.Request) { ctx := &av.Context{Logger: logger} fmt.Println("New connection") defer fmt.Println("Closing write") v := mux.Vars(r) streamName := v["name"] if r.Method == "GET" { fmt.Println("HTTP GET request received...") segSeq := 0 for { writer := NewHTTPStreamer(streamName, segSeq, w) segSeq++ pubsub1 := pubsub.NewSimpleMuxer(100) // 创建http请求推送流着音视频流 go func() { defer writer.Close() fmt.Println("Connected HTTP Writer. Waiting for output.") for { Out := <-pubsub1.Out fmt.Println("Received output") ctx := &av.Context{Logger: logger, Write: writer} fmt.Println(ctx) defer ctx.Flush() stream := Out[0] fmt.Println(stream) _ = avutil.CopyPackets(ctx, stream) } }() aacWrite, h264Write := getHLS(path.Join(videoTempDir, streamName), segSeq) processRTSP(NewRTSPReader(streamName), pubsub1, aacWrite, h264Write) } } } // 实现HTTP音视频流 func NewHTTPStreamer(base string, sn int, w http.ResponseWriter) *HTTPStreamer { str := fmt.Sprintf("%v/%v-%d.ts", videoTempDir, base, sn) f, _ := os.Create(str) return &HTTPStreamer{Writer: f, ResponseWriter: w} } ...
... // 连接RTSP请求和HTTP响应 func streamInvoke(w http.ResponseWriter, r *http.Request) { fmt.Println(r.URL.Path) if strings.HasPrefix(r.URL.Path, "/stream/") { processHTTP(w, r) } else if strings.HasPrefix(r.URL.Path, "/hls/") { processHLS(w, r) } else { fmt.Println(r.URL.Path) w.WriteHeader(404) } } ...
以上就是通过golang来实现RTSP转HTTP的过程的具体实现方式。通过将RTSP流转化为HTTP流,可以使视频媒体的传输更加快速高效,提高用户的观看体验。
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