rstp-server

James Linder jam at tigger.ws
Mon Oct 4 04:45:23 UTC 2021


It seems to me that the whole gst is very nice but fraught with gotchas if you ventutre off the trodden path

eg1)
based on test-appsrc.c if I set GST_DEBUG at *:4 everything works beautifully
if I set GST_DEBUG at less eg *:3 then the output audio from pulsesrc stutters terribly

eg2)
if I connect to rst://192.168.5.118:8554/dvr on either of 2 machines all is perfect
if I connect both then:


0:05:25.152405142 22329 0x7fab08097980 FIXME                default gstutils.c:4025:gst_pad_create_stream_id_internal:<pulsesrc1:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:05:25.154860664 22329 0x7faa80003060 FIXME                default gstutils.c:4025:gst_pad_create_stream_id_internal:<audiosrc:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:05:25.154886046 22329 0x7fab08097b00 WARN                 default v4l2-utils.c:189:gst_v4l2_error:<v4l2src1> error: Device '/dev/video2' is busy
0:05:25.154896706 22329 0x7fab08097b00 WARN                 default v4l2-utils.c:192:gst_v4l2_error:<v4l2src1> error: Call to S_FMT failed for H264 @ 1920x1080: Device or resource busy
0:05:25.154910273 22329 0x7fab08097b00 WARN                 basesrc gstbasesrc.c:3347:gst_base_src_prepare_allocation:<v4l2src1> Subclass failed to decide allocation
0:05:25.154921644 22329 0x7fab08097b00 WARN                 basesrc gstbasesrc.c:3127:gst_base_src_loop:<v4l2src1> error: Internal data stream error.
0:05:25.154925087 22329 0x7fab08097b00 WARN                 basesrc gstbasesrc.c:3127:gst_base_src_loop:<v4l2src1> error: streaming stopped, reason not-negotiated (-4)
0:05:25.154964230 22329 0x7faa800029e0 FIXME                default gstutils.c:4025:gst_pad_create_stream_id_internal:<videosrc:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:05:45.153773460 22329      0x197b0c0 WARN               rtspmedia rtsp-media.c:3594:wait_preroll: failed to preroll pipeline
0:05:45.153811734 22329      0x197b0c0 WARN               rtspmedia rtsp-media.c:3964:gst_rtsp_media_prepare: failed to preroll pipeline

does that mean

video to a sink eg ipcpipesink
audio to a sink eg ipcpipesink

Then the server pipeline reads from the pipesrc’s and every new connection uses a new src?
James


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