I am running below gstreamer command for live streaming:
gst-launch-1.0 imxv4l2videosrc device=/dev/video0 fps-n=30 imx-capture-mode=0 ! textoverlay name=overlay text=\"Overlay text here\" valignment=top halignment=left font-desc=\"Sans, 22\"! gdkpixbufoverlay name=imageoverlay location=/home/user/LZ_50/CamOverlay.png ! imxg2dvideotransform ! imxg2dvideosink framebuffer=/dev/fb1 use-vsync=true sync=false"
I want to change the text overlay dynamic in the GStreamer pipeline.
How can I get the pipeline object pointer to change the text overlay dynamic?
Thank
I had written an application but I have an issue with crashing the application using the GStreamer pipeline with overlay image & text.
Finally got the issue of crashing the application.more detail can be found at : imx Gstreamer plugins .
This also helps to reduce CPU usage(almost 20% 1 core).
Related
I want to preview RTMP using gstreamer xvimagesink. i can see the output if i use autovideosink like this:
gst-launch-1.0 -v rtmpsrc location='rtmp://127.0.0.1:1935/live/stream' ! decodebin3 ! autovideosink
but if i replace "autovideosink" with "xvimagesink" i get this:
Setting pipeline to PAUSED ...
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: Could not initialise Xv output
Additional debug info:
xvimagesink.c(1773): gst_xv_image_sink_open (): /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0:
Could not open display (null)
Setting pipeline to NULL ...
Freeing pipeline ...
Both decodebin3 and autovideosink are auto-plugging GStreamer elements. It means that both elements are auto-selecting available and the most appropriate GStreamer plugins to demux/decode (decodebin3) and render video (autovideosink) from, in this case, live RTMP stream.
So it is very possible that, for example,
decodebin3 decodes video in format that xvimagesink cannot show on your platform/hardware and/or with your Gstreamer version,
xvimagesink is not set properly on your platform and it is not related with available display/monitor.
To find out more details about
video format decoded by decodebin3
video sink element "chosen" by autovideosink,
you can set higher (more detailed) debug level of GStreamer with, for example, export GST_DEBUG=3, rerun pipeline and inspect output.
I'm reading a byte-stream YUV420 at 972x720 pixels from a file with Gstreamer using the following command:
gst-launch-1.0 filesrc location=testfile blocksize=1049760 ! rawvideoparse width=972 height=720 framerate=1/1 ! xvimagesink
This works in so far that I get an image but it isn't displayed correctly. When exporting the frames seperately using command:
gst-launch-1.0 filesrc location=testfile blocksize=1049760 ! rawvideoparse width=972 height=720 framerate=1/1 ! multifilesink location="rvp_%d.raw"
I see that when using the element 'rawvideoparse' it will create a file of 1051200 bytes per frame instead of the expected 1049760. When I remove 'rawvideoparse' the frames are exported correctly but my objective is to read them directly from the file into an 'xvimagesink'
Where am I messing up?
Thanks to the GStreamer Development mailing list I got an answer. The problems was that the rawvideoparse element can't handle this resolution. When I switched to 976 width it works.
I'm designing a program to stream an icecast server (radio.clarkson.edu). Ultimately it will be written in Python3, but for now I'm using gst-launch to test the pipeline. I've been working on Debian Jessie and using gstreamer-1.0. Using a file on Wikimedia, I was able to play pretty easily using:
url=https://upload.wikimedia.org/wikipedia/commons/0/0c/Muriel-Nguyen-Xuan-Korsakov-Flight-of-the-bumblebee.flac.oga
gst-launch-1.0 -v souphttpsrc location =$url ! decodebin ! audioconvert ! audioresample ! alsasink
Running the same commands with my stream, I get the output:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = text/uri-list
Missing element: text/uri-list decoder
ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0: Your GStreamer installation is missing a plug-in.
Additional debug info:
gstdecodebin2.c(3977): gst_decode_bin_expose (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0:
no suitable plugins found
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = "NULL"
Freeing pipeline ...
I have tried too many other pipelines to put on one post, but I can answer any other questions.
Thank you
By now you probably have solved that problem, but still here's an idea: text/uri-list indicates that you didn't hand an actual stream to gstreamer, but rather a (textual) playlist that contains stream addresses. I guess gstreamer can't handle those, hence you need to parse them beforehand and then hand an actual audio stream address to it.
I am trying to run certain pipelines on the Command prompt for playing a video and I am often getting these errors/messages/warnings :
WARNING: erroneous pipeline: no element "qtdemux"
WARNING: erroneous pipeline: no element "playbin2"
WARNING: erroneous pipeline: no element "decodebin2"
ERROR: pipeline could not be constructed: no element "playbin".
Following are the pipelines :
gst-launch filesrc location=path to the mp4 file ! playbin2 ! queue ! ffmpegcolorspace ! autovideosink
or
gst-launch -v filesrc location=path to the mp4 file ! qtdemux name=demuxer ! { queue ! decodebin ! sdlvideosink } { demuxer. ! queue ! decodebin ! alsasink }
or
gst-launch -v playbin uri=path to the mp4 file
or
gst-launch -v playbin2 uri=path to the mp4 file
Questions
I wanted to know, if I am I missing the plugins to execute this.
How do I know which plugin is responsible for which or found where?
What is the benefit of implementing the pipeline via c code.Are the missing plugins still required.
Is it good to install the missing plugins form the Synaptic manager or form the Gstreamer site(base,good,bad,ugly)
When we do gst-inspect we get output like this:
postproc: postproc_hdeblock: LibPostProc hdeblock filter
libvisual: libvisual_oinksie: libvisual oinksie plugin plugin v.0.1
flump3dec: flump3dec: Fluendo MP3 Decoder (liboil build)
vorbis: vorbistag: VorbisTag
vorbis: vorbisparse: VorbisParse
vorbis: vorbisdec: Vorbis audio decoder
vorbis: vorbisenc: Vorbis audio encoder
coreindexers: fileindex: A index that stores entries in file
coreindexers: memindex: A index that stores entries in memory
amrnb: amrnbenc: AMR-NB audio encoder
amrnb: amrnbdec: AMR-NB audio decoder
audioresample: audioresample: Audio resampler
flv: flvmux: FLV muxer
flv: flvdemux: FLV Demuxer
What does the x : y ( x and y mean ) ?
Answers,
It looks like gstreamer at your ends was not installed correctly. playbin2, decodebin2 are basic and part of the base plugins
1 Yes you may be missing some plugins
2 Use gst-inspect command to check if it is available
3 From C code you can manage states, register callback, learn more
Yes missing plugins are still required
4 I guess gstreamer site would be better
5 Not sure about this one, would help if you arrange the result in a proper way
Most probably the GST_PLUGIN_PATH is incorrect. Please set the correct path to where the gstremer has been installed.
I'm constructing a gstreamer pipeline that receives two RTP streams from an networked source:
ILBC Audio stream + corresponding RTCP stream
H263 Video stream + corresponding RTCP stream
Everything is put into one gstreamer pipeline so it will use the RTCP from both streams to synchronize audio/video. So far I've come up with this (using gst-launch for prototyping):
gst-launch -vvv gstrtpbin name=rtpbin
udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-2000" port=40000 ! rtpbin.recv_rtp_sink_0
rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink
udpsrc port=40001 ! rtpbin.recv_rtcp_sink_0
rtpbin.send_rtcp_src_0 ! udpsink port=40002 sync=false async=false
udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMU,encoding-params=(string)1,octet-align=(string)1" port=60000 rtpbin.recv_rtp_sink_1
rtpbin. ! rtppcmudepay ! autoaudiosink
udpsrc port=60001 ! rtpbin.recv_rtcp_sink_1
rtpbin.send_rtcp_src_1 ! udpsink port=60002 sync=false async=false
This pipeline works well if the networked source starts out with sending both video and audio. If the videostream is paused later on, gstreamer will still playback audio and even will start playing back the video when the networked source resumes the video stream.
My problem is however that if the networked source starts out with only an audio stream (video might be added later on), the pipeline seems to pause/freeze until the video stream starts as well.
Since video is optional (and can be added/removed at will by the user) in my application, is there any way I can hook up for instance an 'videotestsrc' that will provide some kind of fallback video data to keep the pipeline running when there is no networked video data?
I've tried experimenting with 'videotestsrc' and a thing called 'videomixer' but I think that mixer still requires both streams to be alive. Any feedback is greatly appreciated!
I present a simple function for pause resume by changing bins. In the following example I provide the logic to change destination bin on the fly dynamically. This shall not completely stop the pipeline which is what you seek I believe. A similar logic could be used for src bins. Here you may remove your network source bin and related decoder/demux bins and add videotestsrc bins.
private static void dynamic_bin_replacement(Pipeline pipe, Element src_bin, Element dst_bin_new, Element dst_bin_old) {
pipe.pause();
src_bin.unlink(dst_bin_old);
pipe.remove(dst_bin_old);
pipe.add(dst_bin_new);
dst_bin_new.syncStateWithParent();
src_bin.link(dst_bin_new);
pipe.ready();
pipe.play();
}
The other logic you may want to try is "PADLOCKING". Please take a look at the following posts
http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-block.txt
and
http://web.archiveorange.com/archive/v/8yxpz7FmOlGqxVYtkPb4
and
Adding and removing audio sources to/from GStreamer pipeline on-the-go
UPDATE
Try output-selector and input-selector bins as they seem to be better alternative. I found them most reliable and have had immense luck with them. I use fakesink or fakesrc respectively as the other end of the selector.
valve bin is another alternative that I found doesn't even need fakesink or fakesrc bins. It is also extremely reliable.
Also the correct state transition order for media file source
NULL -> READY -> PAUSED -> PLAYING (Upwards)
PLAYING -> PAUSED -> READY -> NULL (Downwards)
My order in the above example should be corrected where ready() should come before pause(). Also I would tend to think un-linking should be performed after null() state and not after pause(). I haven't tried these changes but theoretically they should work.
See the following link for detailed info
http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-states.txt?h=BRANCH-RELEASE-0_10_19