I'm having a problem trying to record audio+video from my webcam to a file. If I use videotestsrc and autoaudiosrc I get everything right (read as in I get a file with audio recorded from the webcam's mic, and test-video image), but as soon as I replace videotestsrc with v4l2src (or autovideosrc) I get Error starting streaming on device '/dev/video0'.
The command I'm using:
gst-launch-0.10 videotestsrc ! queue ! ffmpegcolorspace! theoraenc ! queue ! oggmux name=mux autoaudiosrc ! queue ! audioconvert ! vorbisenc ! queue ! mux. mux. ! queue ! filesink location = test.ogg
Why is that happening? What am I doing wrong?
EDIT:
In fact, something as simple as
gst-launch-0.10 autovideosrc ! autovideosink autoaudiosrc ! autoaudiosink
is failing with the same error (Error starting streaming on device '/dev/video0')
Replacing autovideosrc with videotestsrc gives me test image + real audio.
Replacing autoauidosrc with audiotestsrc gives me real image + test audio.
I'm starting to think that this is some kind of limitation of my webcam. Is that possible?
EDIT:
GST_DEBUG=2 log here: http://pastie.org/4755009
EDIT 2:
GST_DEBUG="v4l2*:5" (gstreamer 0.10): http://pastie.org/4810519
GST_DEBUG="v4l2*:5" (gstreamer 1.0): http://pastie.org/4810502
Please do a
gst-launch-1.0 v4l2src ! videoscale ! videoconvert ! autovideosink
Does that run? If not repeat as
GST_DEBUG="v4l2*:5" GST_DEBUG_NO_COLOR=1 gst-launch 2>debug.log ...
and check the log for errors. You also might want to run v4l-info (install v4l-conf under debian/ubuntu) and report what formats your camera supports.
Related
As title, how can I change this so it also plays the files audio too?
gst-launch-1.0 filesrc location='/usr/share/myfile.mp4' ! qtdemux ! h264parse ! imxvpudec ! imxipuvideosink framebuffer=/dev/fb2 &
The I can get the file to play with audio using
gst-launch-1.0 -v playbin uri=file:///path/to/somefile.mp4
But I need the output to be onto device fb2 like in the first example
Many thanks
I posted a link to this question into the gstreamer reddit and a hero called Omerzet saved the day.
The following is the solution:
gst-launch-1.0 filesrc location='/usr/share/myfile.mp4' ! qtdemux name=demux demux.video_0 ! queue ! h264parse ! imxvpudec ! imxipuvideosink framebuffer=/dev/fb2 demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! alsasink device="sysdefault:CARD=imxhdmisoc"
Where framebuffer diverts the video to device /dev/fb2.
And
alsasink device="sysdefault:CARD=imxhdmisoc"
Diverts the audio to my define sound card.
I'm using the following pipeline to listen to a RTSP stream and save a video file:
gst-launch-1.0 -q rtspsrc location=rtsp://<ip>:<port>/video ! decodebin ! autovideoconvert ! x264enc pass=5 quantizer=25 speed-preset=6 ! h264parse ! matroskamux ! filesink location=<filename>
But even though I can see the files generated, they lack the duration of the video when playing on VLC.
I can fix it by passing it through ffmpeg later, but I want to generate the video from gstreamer already completely valid. How can I fix this pipeline?
gst-launch-1.0 -e rtspsrc location=rtsp://<ip>:<port>/video ! decodebin ! videoconvert ! x264enc ! mp4mux ! filesink location="filename.mp4"
This will create a video with duration shown correctly
I'm really running out of ideas. Here is my problem: I need to stream on demand mp4 (H264) through the network. I'm new with gstreamer and after lot of tries with versions > 1.0 I decided to use 0.10 because seems to be most promising so far.
Command below works perfect ( I see window with my movie )
gst-launch filesrc location=/home/zuko/sintel_trailer-368p.mp4 ! decodebin2 name=dec ! queue ! ffmpegcolorspace ! autovideosink dec. ! queue ! audioconvert ! audioresample ! autoaudiosink
Now I'm trying to build TCP stream using commands (so far on localhost only):
Server side:
gst-launch filesrc location=/home/zuko/sintel_trailer-368p.mp4 ! decodebin2 name=dec ! tcpserversink host=127.0.0.1 port=5000
Client side:
gst-launch tcpclientsrc host=127.0.0.1 port=5000 ! decodebin2 name=dec ! queue ! ffmpegcolorspace ! autovideosink dec. ! queue ! audioconvert ! audioresample ! autoaudiosink
But response from the "client side" command is following:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstDecodeBin2:dec/GstTypeFindElement:typefind: Could not determine type of stream.
Additional debug info:
gsttypefindelement.c(813): gst_type_find_element_chain_do_typefinding (): /GstPipeline:pipeline0/GstDecodeBin2:dec/GstTypeFindElement:typefind
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
What is missing, or what I'm doing wrong?
I'm testing on: VirtualBox 4.3.12 with Ubuntu 14.04, kernel 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:30:00 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
Full error with (GST_DEBUG_NO_COLOR=1 GST_DEBUG=*:3 ) attached here:
https://app.box.com/s/4ntyk6am2ibg0pohtg9h
First off, using 0.10 is an absolutely bad idea, you should really stick to 1.0, for which you will have community support.
Second, to your problem itself, you are trying to stream the decoded stream over the network ("decodebin2 ! tcpserversink") and to decode it again on the other side ("tcpclientsrc ! decodebin2"). Not only is it very wrong with respect to bandwidth usage, it also straight up won't work.
I'll advise you to have a look at the rtp plugins provided by gstreamer.
Using gstreamer 1.0 the server side can share h264 streams with:
gst-launch-1.0 filesrc location="C:\\Videos\\videotestsrc.avi" ! decodebin ! x264enc ! mpegtsmux ! queue ! tcpserversink host=127.0.0.1 port=8080
While the client side receives with:
gst-launch-1.0 tcpclientsrc host=127.0.0.1 port=8080 ! decodebin ! videoconvert ! autovideosink sync=false
Alternatively, the client could be simulated with VLC through:
Media >> Open Network Stream >> tcp://127.0.0.1:8080 >> Play
I have a question about gstreamer.
i made a streaming server using gst-rtsp-server. I'm trying to send camera capture to another machine (on the local network) and to parse it into an .ogv file.
The transmission of the streaming works fine, and i'm able to parse the informations into the file; but i can't read it or use it with any application after this parsing. It seems that there are some information missing (probably in relation with the encoding technique, i don't really know much about it)
Server side command (inside c++ code):
....
gst_rtsp_media_factory_set_launch (factory, "( v4l2src device=/dev/video0 ! videorate !
video/x-raw-yuv,width=320,height=240,framerate=30/1 ! videoscale ! ffmpegcolorspace !
theoraenc ! rtptheorapay name=pay0 pt=96 )");
gst_rtsp_media_factory_set_shared (factory, TRUE);
/* attach the test factory to the /test url */
gst_rtsp_media_mapping_add_factory (mapping, "/stream", factory);
....
Client side command (terminal command) :
gst-launch -v rtspsrc location=rtsp://192.168.0.115:8554/stream !
rtptheoradepay name=pay0 ! oggmux ! filesink location=/home/jean/Desktop/stream.ogv
Any help any kind of help is well appreciated !
Jean
I could decode the pipeline as follows to view it gst-launch -v rtspsrc location="rtsp://localhost:8554/test" name=demux demux. ! queue ! rtptheoradepay ! theoradec ! ffmpegcolorspace ! autovideosink
To decode it
gst-launch -v rtspsrc location="rtsp://localhost:8554/test" ! application/x-rtp, payload=96 ! rtptheoradepay ! theoradec ! videorate ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=GIBBERISH.ogg
I decode it and encode it back with the videorate before writing to the file. There may be a more optimal way to perform the same but it is just a work around.
hi I am trying to visualize a music file in gstreamer using the following command:
gst-launch filesrc location=file.mp3 ! decodebin ! audioconvert !
tee name=myT myT. ! queue ! autoaudiosink myT. ! queue ! goom !
colorspace ! autovideosink
But I get this error : "There may be a timestamping problem, or this computer is too slow."
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock
WARNING: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0/GstDshowVideoSink:autovideosink0-actual-sink-dshowvideo: A lot of buffers are being dropped.
Additional debug info:
..\Source\gstreamer\libs\gst\base\gstbasesink.c(2572): gst_base_sink_is_too_late (): /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0/GstDshowVideoSink:autovideosink0-actual-sink-dshowvideo:
There may be a timestamping problem, or this computer is too slow.
ERROR: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0
Assuming this is something to do with the thread, I tried the following command:
gst-launch filesrc location=file.mp3 ! decodebin ! audioconvert ! tee name=myT
{ ! queue ! autoaudiosink } { tee. ! queue ! goom ! colorspace ! autovideosink }
But then it gives the folloiwng link error:
** (gst-launch-0.10:5308): WARNING **: Trying to connect elements that don't share a common ancestor: tee and queue1
0:00:00.125000000 5308 003342F0 ERROR GST_PIPELINE grammar.tab.c:656:gst_parse_perform_link: could not link tee to queue1
WARNING: erroneous pipeline: could not link tee to queue1
Can anyone tell what is wrong? Thanks
I cannot give you an exact answer because i don't have windows installed.
For debugging this use your first pipeline (in linux works). Use parameter -v with gst-launch and put element identity just before autovideosink. This will print buffer information that passes through element identity, look for anything strange.
Also you could try to use directdrawsink instead of autovideosink. Another test that i will do is to generate the audio with audiotestsrc.
Remember that if you find a bug you can open a bug report in gnome bugzilla so GStreamer developers are aware that there is a problem. Even you could fix it yourself and send a patch.
For There may be a timestamping problem, or this computer is too slow. Error Try sync=false like
`gst-launch filesrc location=file.mp3 ! decodebin ! audioconvert ! tee name=myT myT. ! queue ! autoaudiosink myT. ! queue ! goom ! colorspace ! autovideosink sync=false`
or you may have to try at both sink ends of the Tee like
`gst-launch filesrc location=file.mp3 ! decodebin ! audioconvert ! tee name=myT myT. ! queue ! autoaudiosink sync=false myT. ! queue ! goom ! colorspace ! autovideosink sync=false`
I also observed that if you replace autovideosink with xvimagesink or ximagesink the timestamping problem apparently seems to be solved.