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GStreamer videotestsrc 到 RTP

转载 作者:行者123 更新时间:2023-12-04 18:44:41 27 4
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我只是想让 RTP 示例正常工作,但由于缺少插件或引脚不正确,我看到的每个示例都无法执行。

这似乎是最有希望的,但是,尽管服务器和客户端似乎正确启动并进入“播放”状态,但没有任何 react :

服务器:

gst-launch -v videotestsrc ! \
video/x-raw-rgb, format=\(fourcc\)RGB, width=4, height=4, frame-rate=1/1 ! rtpvrawpay !
udpsink host=127.0.0.1

服务器输出:
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)RGB, depth=(string)8, width=(string)4, height=(string)4, colorimetry=(string)SMPTE240M, payload=(int)96, ssrc=(uint)3779397700, clock-base=(uint)1161131286, seqnum-base=(uint)43390
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...

客户:
gst-launch-0.10 -v udpsrc caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)RGB, depth=(string)8, width=(string)4, height=(string)4, colorimetry=(string)SMPTE240M, payload=(int)96, ssrc=(uint)3779397700, clock-base=(uint)1161131286, seqnum-base=(uint)43390" ! rtpvrawdepay ! xvimagesink 

客户端输出:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:src: caps = video/x-raw-rgb, width=(int)4, height=(int)4, format=(fourcc)0x00000000, framerate=(fraction)0/1, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, bpp=(int)24, depth=(int)24
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)RGB, depth=(string)8, width=(string)4, height=(string)4, colorimetry=(string)SMPTE240M, ssrc=(uint)3779397700, clock-base=(uint)1161131286, seqnum-base=(uint)43390

最佳答案

这些工作:

服务器:

gst-launch-0.10 -v \
gstrtpbin name=rtpbin1 \
videotestsrc ! x264enc ! rtph264pay ! rtpbin1.send_rtp_sink_0 \
rtpbin1.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5011 \
rtpbin1.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5012 \
udpsrc port=5013 ! rtpbin1.recv_rtcp_sink_0

客户:
gst-launch-0.10 -v \
videomixer name=mix ! ffmpegcolorspace ! autovideosink sync=false async=false \
gstrtpbin name=rtpbin1 \
udpsrc port=5011 caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z0LAFdkBQfsBEAAAAwAXc1lAAPFi5IAA\\,aMuMsg\\=\\=\", ssrc=(uint)595281375, payload=(int)96, clock-base=(uint)3105254905, seqnum-base=(uint)59233" ! rtpbin1.recv_rtp_sink_0 rtpbin1. ! rtph264depay ! queue ! ffdec_h264 ! videobox border-alpha=0 top=0 left=0 ! mix. \
udpsrc port=5012 ! rtpbin1.recv_rtcp_sink_0 \
rtpbin1.send_rtcp_src_0 ! udpsink port=5013 host=127.0.0.1

关于GStreamer videotestsrc 到 RTP,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/17127965/

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