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使用GStreamer播放传入的RTP流

如何解决《使用GStreamer播放传入的RTP流》经验,为你挑选了1个好方法。

我正在开发一个GStreamer应用程序,并为实现传入RTP流的播放器而苦苦挣扎.我正在尝试围绕gstrtpbin元素构建一个管道.我正在尝试使用gst-launch构造对管道进行建模:

VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"

gst-launch -v udpsrc caps=$VIDEO_CAPS port=4444 \
              ! gstrtpbin .recv_rtp_sink_0 \
              ! rtph264depay ! ffdec_h264 ! xvimagesink

当我启动脚本时,GStreamer报告这些错误:

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0: ntp-ns-base = 3469468914024449000
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0.GstPad:recv_rtp_sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_sink_0: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_sink_0.GstProxyPad:proxypad0: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0.GstPad:recv_rtp_src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpPtDemux:rtpptdemux0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264
/GstPipeline:pipeline0/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_0_960476599_33.GstProxyPad:proxypad1: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)33
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2378): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)
Execution ended after 209381685 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
/GstPipeline:pipeline0/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_0_960476599_33: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpPtDemux:rtpptdemux0.GstPad:src_33: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpPtDemux:rtpptdemux0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:src_960476599: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0.GstPad:recv_rtp_src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0.GstPad:recv_rtp_sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_sink_0: caps = NULL
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL
Setting pipeline to NULL ...
Freeing pipeline ...

我应该提到它适用于playbin和SDP文件.例如这个文件:

v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 4444 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000

可以像这样播放流:

gst-launch -vvv playbin uri=file://`pwd`/stream.sdp

为了完整性:我使用VLC发送数据.这是命令:

vlc -I rc /usr/local/movies/sample.mp4 \
    --screen-fps=10 :screen-caching=100 \
    --sout='#transcode{vcodec=h264,venc=x264{bframes=0,keyint=40},vb=512}:\
                   rtp{mux=ts,dst=127.0.0.1,port=4444}'

有人会帮我理解gst-launch脚本失败的原因吗?错误"原因未链接"让我认为gstrtpbin和rtph264depay元素之间的链接被破坏了.但我不知道如何解决它.

编辑
遵循RAOF的建议我在我的命令中修复了一些错误.但是我使用的是ffdec_h264和autovideosink,因为在我的Windows系统上我没有安装fluh264dec和xvimage sink插件:

gst-launch-0.10 udpsrc port=4444 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" ! .recv_rtp_sink_0 gstrtpbin ! rtpmp2tdepay ! mpegtsdemux ! ffdec_h264 ! autovideosink 

这会导致新的错误:

0:00:00.743000000   516   024070A8 ERROR                 ffmpeg .:0:: non-existing PPS referenced
0:00:00.744000000   516   024070A8 ERROR                 ffmpeg .:0:: non-existing PPS referenced
0:00:00.745000000   516   024070A8 ERROR                 ffmpeg .:0:: decode_slice_header error
0:00:00.745000000   516   024070A8 ERROR                 ffmpeg .:0:: no frame!
0:00:00.812000000   516   024070A8 ERROR                 ffmpeg .:0:: non-existing PPS referenced
0:00:00.813000000   516   024070A8 ERROR                 ffmpeg .:0:: non-existi
...
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow
 error.
Additional debug info:
..\Source\gstreamer\libs\gst\base\gstbasesrc.c(2378): gst_base_src_loop (): /Gst
Pipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-negotiated (-4)
Execution ended after 4790000000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

我还在试图弄清楚如何解决这个问题.如果您能提供帮助,请随时这样做.

Edit2
我使用SDP解决方案再次测试并观察到"不存在的PPS"错误也会发生,但视频确实会播放.另一方面,致命的"内部数据流错误"仅在使用自定义管道解决方案时显示.我怀疑"不存在的PPS"错误是由x264编码器引起的."内部数据流错误"必须由我的管道中的错误引起,或者某些Windows插件中的错误.我会进一步研究......



1> RAOF..:

据我所知,你有两个问题:

首先,似乎接收器规范的顺序很重要:而不是... ! gstrtpbin .recv_rtp_sink_0 ! ...你需要的... ! .recv_rtp_sink_0 gstrtpbin ! ....

其次,vlc正在发送一个MPEG2传输流 - 你已经获得mux=ts了rtp流输出描述符 - 但是你正试图降低原始h264流的负载.您需要对ts流进行depayload,然后对其进行解复用以获取h264流数据.

所以,最后,管道

gst-launch-0.10 -v udpsrc port=4444 \
caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" \
! .recv_rtp_sink_0 gstrtpbin ! rtpmp2tdepay \
! mpegtsdemux ! fluh264dec ! xvimagesink

适用于我,使用TS demuxer(mpegtsdemux)和h264解码器(fluh264dec).

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