I run GStreamer exsample:qglwtextureshare. But I found it will crash at
qglrenderer.cpp Line168 Q_ASSERT (gst_is_gl_memory (mem)); I modified the pipeline like this :
m_pipeline =
GST_PIPELINE (gst_parse_launch
("videotestsrc ! "
"video/x-raw, width=640, height=480, "
"framerate=(fraction)30/1 ! "
"fakesink sync=1", NULL)); because gleffects effect=5 will pop error can not get fisheye shader.
And I don't change otherthings. I don't know why it can not get the frame data. Please help me, thanks.
Related
I'm using this Gstreamer pipeline to send an RTSP stream of a camera.
./gst-rtsp-launch --port 8554 "( v4l2src device=/dev/video0 ! video/x-raw,framerate=30/1,width=640,height=480 ! rtpvrawpay name=pay0 pt=96 )"
I want to using playbin, so I don't need to specify the type of video from the rtsp stream. If I use this pipeline, I can get a single image from the camera:
gst-launch-1.0 playbin uri=rtsp://(ip-of-camera):8554 video-sink="jpegenc ! filesink location=capture1.jpeg"
But if I try this pipeline, to save as a file:
gst-launch-1.0 playbin uri=rtsp://(ip-of-camera):8554 video-sink="videoconvert ! video/x-h264,width=320,height=240 ! mp4mux ! filesink location=test.mp4"
I get this error:
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source/GstUDPSrc:udpsrc1: Internal data stream error.
Additional information for debugging:
../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source/GstUDPSrc:udpsrc1:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.094968753
Defining the processing queue to NULL...
Freeing the processing queue...
Note: I had translate the last two lines.
Is there a problem in the pipeline I'm using to save the stream as a file?
I copied your pipeline and in worked.
I only added -e after gst-launch-1.0, so that GStreamer writes all necessary information to the file when ctrl-C is pressed:
gst-launch-1.0 -e playbin uri=(rtsp url) video-sink="videoconvert ! video/x-h264,width=320,height=240 ! mp4mux ! filesink location=test.mp4"
Maybe you are using an older version of GStreamer?
I am using GStreamer 1.20.0
I am building my pipeline like this:
gst-launch-1.0 -v filesrc location=" + video_filename + " ! qtdemux ! queue max-size-buffers=0 max-size-time=0 ! vpudec frame-drop=0 ! queue ! imxv4l2sink device=/dev/video16
This is how I stop video playback
gst_element_set_state (m_gst_pipeline, GST_STATE_NULL);
g_main_loop_quit (m_mainloop);
gst_object_unref(m_gst_pipeline);
The stopping itsself works fine, only issue is that I get the last video frame displayed even when exiting the program. How do I flush the pipeline before destruction so I have a black screen?
Initial Issue
I would like to use gstreamer plugin opusdec to decode an Opus bitstream. The final purpose is to make glue around it with appsrc and appsink as input/output, to decode 20 ms Opus packets coming from a RTP packet payload and provide PCM sample.
Remark: I can't use gstreamer rtpopusdepay
The following pipeline works:
gst-launch-1.0 filesrc location=testvector01.bit.opus ! oggdemux !
opusdec ! fakesink
In my final application I'm no expected OGG contained data so I did the following:
1) Desencapsulate Opus bitstream
gst-launch-1.0 filesrc location=testvector01.bit.opus ! oggdemux !
filesink location = testvector01.bit.demux
That works. And then:
2) Decode Opus bitstream
gst-launch-1.0 filesrc location=testvector01.bit.demux ! opusdec !
fakesink
and I have the following error:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2865): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc0:
streaming task paused, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ..
Input File
testvector01.bit.opus
From Opus test vector :https://people.xiph.org/~greg/opus_testvectors/
My question is:
What is the proper way to use gstreamer plugin opusec without transport container?
Update
Gstreamer version 1.2.4
As recommended I tried to add opusparse after filesrc and got the following error.
Pipeline is PREROLLING ...
(gst-launch-1.0:5147): GStreamer-WARNING **:
gstpad.c:4555:store_sticky_event:<opusparse0:src> Sticky event
misordering, got 'caps' before 'stream-start'
(gst-launch-1.0:5147): GStreamer-WARNING **:
gstpad.c:4555:store_sticky_event:<opusdec0:sink> Sticky event
misordering, got 'caps' before 'stream-start' Pipeline is PREROLLED
... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock
ERROR: from element /GstPipeline:pipeline0/GstOpusDec:opusdec0:
Decoding error: -4 Additional debug info: gstopusdec.c(460):
opus_dec_chain_parse_data ():
/GstPipeline:pipeline0/GstOpusDec:opusdec0 Execution ended after
0:00:00.063372478 Setting pipeline to PAUSED ... Setting pipeline to
READY ... Setting pipeline to NULL ... Freeing pipeline ...
GStreamer 1.8.1
The following pipeline
gst-launch-1.0 filesrc location = testvector01.bit.demux ! opusparse !
opusdec ! audioconvert ! alsasink
halt here:
Setting pipeline to PAUSED ... Pipeline is PREROLLING ...
Gstreamer 1.13.1
gst-launch-1.0 filesrc location = testvector01.bit.demux ! opusparse !
opusdec ! alsasink
Playback just produce a short audio glitch while no gstreamer error is raised.
gst-launch-1.0 filesrc location = testvector01.bit.opus ! oggdemux ! opusparse !
opusdec ! alsasink
Playback is choppy while no gstreamer error is raised.
Regards,
appsrc is-live=true do-timestamp=true name=audiosrc ! opusparse ! oggmux ! filesink location=test.ogg
gstreamer 1.14.1 works fine
You need to have a parser (opusparse) in between as opusdec doesn’t know what format it is, try the following pipeline:
gst-launch-1.0 filesrc location=testvector01.bit.demux ! opusparse !
opusdec ! fakesink dump=true
I'm trying to stream H263 via RTP with gstreamer 1.0. It works just fine aside from no key frames being sent. The command line looks like this:
gst-launch-1.0 videotestsrc pattern=ball ! avenc_h263 ! rtph263pay pt=34 ! udpsink host=10.0.75.196 port=25782 sync=true
The result is that it starts from black and only works with changes thereafter. Could it have anything to do with avenc_h263 using stuff that only H263+ or H263++ handles?
I would be very grateful for any help on this!
I finally found the problem! Standard rtp-payload-size is 0. Changing this parameter to anything above zero, I tried 1 and 20, makes it run smooth and with full frames.
gst-launch-1.0 videotestsrc pattern=ball ! avenc_h263 rtp-payload-size=10 ! rtph263pay pt=34 ! udpsink host=10.0.75.196 port=25782 sync=true
I'm trying to play a local mpeg2 TS file with gstreamer with this:
gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink
The first frame appears as big blocks of color and then stops. Any thoughts about what I'm doing wrong here? Does a -TS file need to be handled differently than this?
Here's the log:
$ gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ....
WARNING: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: Internal data flow problem..
Additional debug info:.
gstbasesink.c(3492): gst_base_sink_chain_unlocked (): /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0:
Received buffer without a new-segment. Assuming timestamps start from 0.
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Got EOS from element "pipeline0".
Execution ended after 6866757291 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ..
I think first you should first try to play the file with the help of playbin2. If you are able to play it then u should use decodebin2 ,debug its output and construct your pipeline accordingly.
The syntax for playbin2 is as follows :-
gst-launch playbin2 uri = file:///home/user1031040/Desktop/file.mpg
The syntax for decodebin2 is as follows:-
gst-launch filesrc location = file.mpg ! decodebin2 ! autovideosink