[Bug 767483] Maximum number of clients reached, memory accumulation

GStreamer (GNOME Bugzilla) bugzilla at gnome.org
Wed Jun 29 06:21:13 UTC 2016


https://bugzilla.gnome.org/show_bug.cgi?id=767483

--- Comment #19 from Joona Laine <joonarlaine at gmail.com> ---
Hi,

I'm coming back to this since I just noticed that when I'm running my pipeline
from terminal and I have GST_DEBUG=3 I get the same warnings about PAUSE and
TEARDOWN interruption. I had only used the -v flag on the gst-launch and those
messages didn't appear then. 

gst-launch-1.0 rtspsrc
location=rtsp://root:root@10.128.1.82/axis-media/media.amp latency=0 !
rtph264depay ! h264parse ! queue ! vaapidecode ! videoconvert ! videocrop !
glimagesink
Setting pipeline to PAUSED ...
libva info: VA-API version 0.39.0
libva info: va_getDriverName() returns 0
libva info: Trying to open /usr/local/lib/dri/i965_drv_video.so
libva info: Found init function __vaDriverInit_0_39
libva info: va_openDriver() returns 0
Pipeline is live and does not need PREROLL ...
Got context from element 'sink': gst.gl.GLDisplay=context,
gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayX11\)\ gldisplayx11-0";
0:00:00.136132553  6985      0x11ce640 WARN               structure
gststructure.c:1935:priv_gst_structure_append_to_gstring: No value transform to
serialize field 'gst.vaapi.Display' of type 'GstVaapiDisplay'
Got context from element 'vaapidecode0': gst.vaapi.Display=context,
gst.vaapi.Display=(GstVaapiDisplay)NULL;
Progress: (open) Opening Stream
Progress: (connect) Connecting to
rtsp://root:root@10.128.1.82/axis-media/media.amp
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
Progress: (request) SETUP stream 0
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending PLAY request
Progress: (request) Sending PLAY request
0:00:00.527462710  6985 0x7f69d8021e80 FIXME                default
gstutils.c:3764:gst_pad_create_stream_id_internal:<fakesrc0:src> Creating
random stream-id, consider implementing a deterministic way of creating a
stream-id
Progress: (request) Sent PLAY request
Redistribute latency...
Redistribute latency...
0:00:03.079206448  6985      0x119cb70 WARN                 default
gstglutils.c:470:gst_gl_context_set_error: Output window was closed
0:00:03.102938067  6985      0x11c9990 WARN             glimagesink
gstglimagesink.c:1552:gst_glimage_sink_show_frame:<sink> error: Output window
was closed
ERROR: from element
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLImageSink:sink:
Output window was closed
Additional debug info:
gstglimagesink.c(1552): gst_glimage_sink_show_frame ():
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLImageSink:sink
Execution ended after 0:00:02.575881824
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
0:00:03.103959581  6985      0x11c98f0 WARN                 rtspsrc
gstrtspsrc.c:5477:gst_rtspsrc_try_send:<rtspsrc0> receive interrupted
0:00:03.103982304  6985      0x11c98f0 WARN                 rtspsrc
gstrtspsrc.c:7500:gst_rtspsrc_pause:<rtspsrc0> PAUSE interrupted
0:00:03.108056070  6985      0x11c98f0 WARN                 rtspsrc
gstrtspsrc.c:5477:gst_rtspsrc_try_send:<rtspsrc0> receive interrupted
0:00:03.108092328  6985      0x11c98f0 WARN                 rtspsrc
gstrtspsrc.c:6971:gst_rtspsrc_close:<rtspsrc0> TEARDOWN interrupted
Freeing pipeline ...


Above log comes from launcing the pipeline and closing it after streaming
begins. Can this be a issue in the RTSP server or is it still my element
linking that causes this? I'm yet to fix the onPadAdded() callback...

-Joona

-- 
You are receiving this mail because:
You are the QA Contact for the bug.
You are the assignee for the bug.


More information about the gstreamer-bugs mailing list