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使用 Gstreamer 從網絡攝像頭播放 rtsp stream

[英]Play rtsp stream from webcam using Gstreamer

我想從 IP 相機TS-WPTCAM中獲取 stream 視頻。 我可以使用rtsp://192.168.100.50:19112/ipcam_h264.sdp直接 stream vlc 中的視頻但是當我嘗試使用 Gstreamer 時,它不播放視頻。 下面是output。

Lnx-Workstation:~$ gst-launch-1.0 -v rtspsrc location="rtsp://192.168.100.50:19112/ipcam_h264.sdp" name=demux demux. ! queue max-size-buffers=2 ! rtph264depay ! autovideosink sync=false

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Got context from element 'autovideosink0': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayGBM\)\ gldisplaygbm0";
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://192.168.100.50:19112/ipcam_h264.sdp
0:00:20.130738709 13686 0x5632fcf9d2d0 ERROR                default gstrtspconnection.c:1004:gst_rtsp_connection_connect_with_response: failed to connect: Socket I/O timed out
0:00:20.130840128 13686 0x5632fcf9d2d0 ERROR                rtspsrc gstrtspsrc.c:4702:gst_rtsp_conninfo_connect:<demux> Could not connect to server. (Generic error)
0:00:20.130850670 13686 0x5632fcf9d2d0 WARN                 rtspsrc gstrtspsrc.c:7469:gst_rtspsrc_retrieve_sdp:<demux> error: Failed to connect. (Generic error)
0:00:20.130893392 13686 0x5632fcf9d2d0 WARN                 rtspsrc gstrtspsrc.c:7548:gst_rtspsrc_open:<demux> can't get sdp
0:00:20.130917551 13686 0x5632fcf9d2d0 WARN                 rtspsrc gstrtspsrc.c:5628:gst_rtspsrc_loop:<demux> we are not connected
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:demux: Could not open resource for reading and writing.
Additional debug info:
gstrtspsrc.c(7469): gst_rtspsrc_retrieve_sdp (): /GstPipeline:pipeline0/GstRTSPSrc:demux:
Failed to connect. (Generic error)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

我還嘗試像這樣使用playbin播放視頻:

Lnx-Workstation:~$ gst-launch-1.0 -v playbin uri=rtsp://192.168.100.50:19112/ipcam_h264.sdp uridecodebin0::source::latency=100

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: ring-buffer-max-size = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = rtsp://192.168.100.50:19112/ipcam_h264.sdp
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: latency = 100
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = "\(GstRTSPSrc\)\ source"
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://192.168.100.50:19112/ipcam_h264.sdp
0:00:20.040912220 13549 0x55e2654b5e80 ERROR                default gstrtspconnection.c:1004:gst_rtsp_connection_connect_with_response: failed to connect: Socket I/O timed out
0:00:20.041032034 13549 0x55e2654b5e80 ERROR                rtspsrc gstrtspsrc.c:4702:gst_rtsp_conninfo_connect:<source> Could not connect to server. (Generic error)
0:00:20.041058980 13549 0x55e2654b5e80 WARN                 rtspsrc gstrtspsrc.c:7469:gst_rtspsrc_retrieve_sdp:<source> error: Failed to connect. (Generic error)
0:00:20.041160200 13549 0x55e2654b5e80 WARN                 rtspsrc gstrtspsrc.c:7548:gst_rtspsrc_open:<source> can't get sdp
0:00:20.041185827 13549 0x55e2654b5e80 WARN                 rtspsrc gstrtspsrc.c:5628:gst_rtspsrc_loop:<source> we are not connected
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: Could not open resource for reading and writing.
Additional debug info:
gstrtspsrc.c(7469): gst_rtspsrc_retrieve_sdp (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source:
Failed to connect. (Generic error)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

*** 文件源的 Playbin 有效。

如何使用 GStreamer 播放 RTSP 視頻?

編輯:根據格雷戈里的回答:

Lnx-Workstation:~$ gst-launch-1.0 -v rtspsrc location="rtsp://192.168.100.50:19112/ipcam_h264.sdp" ! queue max-size-buffers=2 ! rtph264depay ! h264parse ! decodebin ! autovideosink sync=false

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://192.168.100.50:19112/ipcam_h264.sdp
0:00:20.108314006  3310 0x563cd86e8850 ERROR                default gstrtspconnection.c:1004:gst_rtsp_connection_connect_with_response: failed to connect: Socket I/O timed out
0:00:20.108425505  3310 0x563cd86e8850 ERROR                rtspsrc gstrtspsrc.c:4702:gst_rtsp_conninfo_connect:<rtspsrc0> Could not connect to server. (Generic error)
0:00:20.108449668  3310 0x563cd86e8850 WARN                 rtspsrc gstrtspsrc.c:7469:gst_rtspsrc_retrieve_sdp:<rtspsrc0> error: Failed to connect. (Generic error)
0:00:20.108540016  3310 0x563cd86e8850 WARN                 rtspsrc gstrtspsrc.c:7548:gst_rtspsrc_open:<rtspsrc0> can't get sdp
0:00:20.108569689  3310 0x563cd86e8850 WARN                 rtspsrc gstrtspsrc.c:5628:gst_rtspsrc_loop:<rtspsrc0> we are not connected
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not open resource for reading and writing.
Additional debug info:
gstrtspsrc.c(7469): gst_rtspsrc_retrieve_sdp (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Failed to connect. (Generic error)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

錯誤似乎是一樣的。

您需要從RTP加載H264 ,但您忘記在將其傳遞給depay之前autovideosink進行解析和解碼。 我也不知道你為什么需要demux部分,因為你只使用視頻和一個 stream。嘗試以下操作:

gst-launch-1.0 -v rtspsrc location="rtsp://192.168.100.50:19112/ipcam_h264.sdp" ! queue max-size-buffers=2 ! rtph264depay ! h264parse ! decodebin ! autovideosink sync=false

不確定您的情況,但 IIRC 如果安裝了 plugins-ugly,某些 gstreamer 版本可能會出現 RTSP 身份驗證問題。 您可以嘗試:

sudo apt-get remove gstreamer1.0-plugins-ugly

如果還不夠,您可以分享 sdp 以獲得進一步的建議。

在這里使用mplayer

mplayer rtsp://username:password@IPADDRESS:PORT/your-path-to-stream

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