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我正在开发一个 GStreamer 应用程序，并且在为传入的 RTP 流实现播放器方面遇到了一些困难。我正在尝试围绕 gstrtpbin 元素构建管道。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 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

可以用来播放这样的流：

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

为了完整性：我正在使用 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}'

有人能帮我理解为什么 gst-launch 脚本会失败吗？错误“原因未链接”让我认为 gstrtpbin 和 rtph264depay 元素之间的链接已损坏。但我不知道如何解决它。

编辑
按照 RAOF 的建议，我修复了命令中的一些错误。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 ...

我仍在试图弄清楚如何解决这个问题。如果您可以提供帮助，请随时提供帮助。

Edit2
我使用 SDP 解决方案再次测试，观察到“不存在的 PPS”错误也会出现，但视频确实可以播放。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”一定是由我的管道中的错误引起的，或者可能是某些 Windows 插件中的错误。我会做一些进一步的研究...

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3 回答 3

12

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 ! ...你需要拥有... ! .recv_rtp_sink_0 gstrtpbin ! ....

其次，vlc 正在发送一个 MPEG2 传输流——你已经mux=ts在 rtp 流输出描述符中——但你试图卸载一个原始的 h264 流。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

为我工作，使用 TS demuxer (mpegtsdemux) 和 h264 解码器 (fluh264dec)。

于 2009-12-22T03:21:38.177 回答
1 
gst-launch-0.10 -vvvv rtspsrc location=rtsp://192.168.250.100:554 latency=100 ! \
application/x-rtp,media="video",payload=99,clock-rate=90000,encoding-name="H264"  ! \
rtph264depay !  ffdec_h264 ! ffmpegcolorspace ! xvimagesink

这对我来说适用于“Grandtec Electronic MegaPixel WIFI CAM”

于 2012-05-04T17:59:43.747 回答
0

你也可以试试这个。

gst-launch-0.10 -v rtspsrc location="rtsp://10.107.2.217/StreamingSetting?version=1.0&action=getRTSPStream&ChannelID=1&ChannelName=Channel1" user-id=admin user-pw=admin123 caps=" application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264,payload=(int)96,ssrc=(uint)237526004,clock-base=(uint)1584170994,seqnum-base=(uint)42626" port=554 ! rtph264depay queue-delay=0 ! h264parse ! decodebin2 ! queue leaky=1 ! autovideosink

当您在网络上的流媒体是安全的时，它也可以工作，并且它可以工作

rtsp://10.107.2.217

RTSP Port : 554

Video Codec : H.264

希望，对大家有帮助。

于 2014-03-26T08:29:05.867 回答