我正在玩gstreamer将各种来源流式传输到RTMP. 它适用于RTSP和MP4源，但是当我尝试asf源时它不起作用。
我正在使用的管道是：gst-launch-1.0 -v filesrc location="C:\\Users\\user1\\Download
s\\Video\\test2.asf" ! asfparse ! asfdemux ! flvmux ! rtmpsink location='rtmp://127.0.0.1:1935/live/test'
我得到的输出是：
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstAsfParse:asfparse0.GstPad:src: caps = video/x-ms-asf, parsed=(boolean)true
/GstPipeline:pipeline0/GstASFDemux:asfdemux0.GstPad:sink: caps = video/x-ms-asf, parsed=(boolean)true
/GstPipeline:pipeline0/GstFlvMux:flvmux0: streamable = true
/GstPipeline:pipeline0/GstFlvMux:flvmux0.GstPad:src: caps = video/x-flv, streamheader=(buffer)< 464c..., 1200.... >
/GstPipeline:pipeline0/GstRTMPSink:rtmpsink0.GstPad:sink: caps = video/x-flv, streamheader=(buffer)< 464c.., 1200... >
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Got EOS from element "pipeline0". <<<-------- why?
Execution ended after 0:00:00.001711787
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
该文件使用playbin.
流信息：
如果我更改管道以重新编码视频，则流媒体端没有错误，但两者都VLC无法gstreamer playbin播放，他们似乎只是永远尝试。
用于重新编码和流式传输的管道：
gst-launch-1.0 -v filesrc location="C:\\Users\\user1\\Downloads\\Video\
\test2.asf" ! asfparse ! asfdemux ! decodebin ! x264enc ! flvmux ! rtmpsink location='rtmp://127.0.0.1:1935/live/test2'