SK-AM69: gstreamer RTSP with v4l2h264dec

Hi Jason,

Can you try the below pipeline and see if it works?

GST_DEBUG=3 gst-launch-1.0 rtspsrc location=rtsp://192.168.109.242/live_st1 latency=50 ! rtph264depay wait-for-keyframe=1 ! h264parse ! v4l2h264dec ! videoconvert ! waylandsink

Best Regards,

Suren

Hi Suren,

Thanks your reply.

I try, but it doesn't work, see the log below.

Best Regards,

LOG:
weston@am69-sk:~$ GST_DEBUG=3 gst-launch-1.0 rtspsrc location=rtsp://192.168.109.242/live_st1 latency=50 ! \
> rtph264depay wait-for-keyframe=1 ! h264parse ! v4l2h264dec ! videoconvert ! waylandsink
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Pipeline is PREROLLED ...
Prerolled, waiting for progress to finish...
Progress: (connect) Connecting to rtsp://192.168.109.242/live_st1
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
0:00:00.078091736 1321 0xffff80000b70 WARN udpsrc gstudpsrc.c:1638:gst_udpsrc_open:<udpsrc0> warning: Could not create a buffer of requested 524288 b?
0:00:00.078179742 1321 0xffff80000b70 WARN udpsrc gstudpsrc.c:1648:gst_udpsrc_open:<udpsrc0> have udp buffer of 212992 bytes while 524288 were requesd
Progress: (request) SETUP stream 0
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending PLAY request
Redistribute latency...
Progress: (request) Sending PLAY request
0:00:00.088038871 1321 0xffff80000d20 FIXME default gstutils.c:4036:gst_pad_create_stream_id_internal:<fakesrc0:src> Creating random stream-id, consided
Redistribute latency...
Progress: (request) Sent PLAY request
Redistribute latency...
0:00:00.104402936 1321 0xffff80001230 WARN v4l2 gstv4l2object.c:3408:gst_v4l2_object_reset_compose_region:<v4l2h264dec0:sink> Failed to get defaultt
Redistribute latency...
0:00:00.427265413 1321 0xffff80001230 WARN GST_CAPS gstpad.c:5813:pre_eventfunc_check:<v4l2h264dec0:sink> caps video/x-h264, stream-format=(string)byted
0:00:00.428262048 1321 0xffff80001230 WARN GST_CAPS gstpad.c:5813:pre_eventfunc_check:<v4l2h264dec0:sink> caps video/x-h264, stream-format=(string)byted
0:00:00.429020946 1321 0xffff80001230 WARN GST_CAPS gstpad.c:5813:pre_eventfunc_check:<v4l2h264dec0:sink> caps video/x-h264, stream-format=(string)byted
0:00:00.429670442 1321 0xffff80001230 WARN GST_CAPS gstpad.c:5813:pre_eventfunc_check:<v4l2h264dec0:sink> caps video/x-h264, stream-format=(string)byted
0:00:00.429914935 1321 0xffff80000ed0 WARN basesrc gstbasesrc.c:3134:gst_base_src_loop:<udpsrc0> error: Internal data stream error.
0:00:00.429937645 1321 0xffff80000ed0 WARN basesrc gstbasesrc.c:3134:gst_base_src_loop:<udpsrc0> error: streaming stopped, reason not-negotiated (-4)
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal data stream error.
Additional debug info:
/usr/src/debug/gstreamer1.0/1.22.12/libs/gst/base/gstbasesrc.c(3134): gst_base_src_loop (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.342770967
Setting pipeline to NULL ...
0:00:00.430861939 1321 0xffff80000b70 WARN default gstrtspconnection.c:1417:writev_bytes: Operation was cancelled
0:00:00.430907170 1321 0xffff80000b70 WARN rtspsrc gstrtspsrc.c:6796:gst_rtspsrc_try_send:<rtspsrc0> error: Could not send message. (Received end-of-f)
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not write to resource.
Additional debug info:
/usr/src/debug/gstreamer1.0-plugins-good/1.22.12/gst/rtsp/gstrtspsrc.c(6796): gst_rtspsrc_try_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Could not send message. (Received end-of-file)
0:00:00.431066461 1321 0xffff80000b70 WARN rtspsrc gstrtspsrc.c:9267:gst_rtspsrc_pause:<rtspsrc0> error: Could not send message. (Received end-of-file)
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not write to resource.
Additional debug info:
/usr/src/debug/gstreamer1.0-plugins-good/1.22.12/gst/rtsp/gstrtspsrc.c(9267): gst_rtspsrc_pause (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Could not send message. (Received end-of-file)
Freeing pipeline ...

Hi Jason,

Is there a possibility you can enable this rtsp stream accessible? For us to validate on our end.

Best Regards,

Suren

Hi Suren,

Due to the company’s security policy, this RTSP stream cannot be made accessible externally.
The vendor and model are Aver CAM520 Pro3.
Here is the related webpage:
https://www.aver.com/Downloads/search?q=CAM520%20Pro3

This is the information obtained using ffprobe:

$ ffprobe -hide_banner -i rtsp://192.168.109.242/live_st1
Input #0, rtsp, from 'rtsp://192.168.109.242/live_st1':
Metadata:
title : RTSP/RTP stream from AVer
comment : LIVE555 Streaming Media v2019.03.06
Duration: N/A, start: 1.100100, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p(tv, gbr/reserved/reserved, progressive), 1920x1080, 30 fps, 29.97 tbr, 90k tbn, 60 tbc

Hope this helps.

I will update its firmware to see if that makes any difference.

Best Regards,
Jason

Hi Jason,

I suspect this could be a problem handling the colorimetries in Gstreamer. Can you apply the below patch on V4L2 object that handles colorimetries and let us know if it solves the problem.

diff --git a/subprojects/gst-plugins-good/sys/v4l2/gstv4l2object.c b/subprojects/gst-plugins-good/sys/v4l2/gstv4l2object.c
index f3e0e99829..03628fa559 100644
--- a/subprojects/gst-plugins-good/sys/v4l2/gstv4l2object.c
+++ b/subprojects/gst-plugins-good/sys/v4l2/gstv4l2object.c
@@ -2614,13 +2614,13 @@ return_data:
       "height", G_TYPE_INT, (gint) height, NULL);   gst_v4l2_object_add_aspect_ratio (v4l2object, s);
-
+#if 0
   if (!v4l2object->skip_try_fmt_probes) {
     gst_v4l2_object_add_interlace_mode (v4l2object, s, width, height,
         pixelformat);
     gst_v4l2_object_add_colorspace (v4l2object, s, width, height, pixelformat);
   }
-
+#endif
   if (G_IS_VALUE (&rates)) {
     gst_v4l2src_value_simplify (&rates);

Hope this helps

Best Regards,

Suren