在Gstreamer中使用SRT协议 使用以下两个命令,我可以通过SRT传输videotestsrc源

在Gstreamer中使用SRT协议 使用以下两个命令,我可以通过SRT传输videotestsrc源,gstreamer,srt,Gstreamer,Srt,并以这种方式播放: gst-play-1.0 srt://127.0.0.1:8888 现在,我想流式传输一个rtsp源,我通过以下方式获得它: 但是,当我在播放时出现以下错误: gst-play-1.0 srt://127.0.0.1:8888 Press 'k' to see a list of keyboard shortcuts. Now playing srt://127.0.0.1:8888 Pipeline is live. ERROR Could not determine

并以这种方式播放:

gst-play-1.0 srt://127.0.0.1:8888
  • 现在,我想流式传输一个rtsp源,我通过以下方式获得它:
  • 但是,当我在播放时出现以下错误:

    gst-play-1.0 srt://127.0.0.1:8888
    
    Press 'k' to see a list of keyboard shortcuts.
    Now playing srt://127.0.0.1:8888
    Pipeline is live.
    ERROR Could not determine type of stream. for srt://127.0.0.1:8888
    ERROR debug information: ../subprojects/gstreamer/plugins/elements/gsttypefindelement.c(999): gst_type_find_element_chain_do_typefinding (): /GstPlayBin:playbin/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind
    Reached end of play list.
    
    我怎样才能解决它

    gst-launch-1.0 rtspsrc location=rtsp://localhost:8554/main latency=100 ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! video/x-raw,width=640,height=480 ! srtsink uri="srt://:8888" sync=false async=false
    
    gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! identity silent=false ! fakesink async=false
    
    gst-play-1.0 srt://127.0.0.1:8888
    
    Press 'k' to see a list of keyboard shortcuts.
    Now playing srt://127.0.0.1:8888
    Pipeline is live.
    ERROR Could not determine type of stream. for srt://127.0.0.1:8888
    ERROR debug information: ../subprojects/gstreamer/plugins/elements/gsttypefindelement.c(999): gst_type_find_element_chain_do_typefinding (): /GstPlayBin:playbin/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind
    Reached end of play list.