I don't understand why I'm getting a green image. I'd appreciate any insights.
Producer:
gst-launch-0.10 -v videotestsrc ! vp8enc ! rtpvp8pay ! udpsink host=127.0.0.1 port=9001
Consumer:
gst-launch-0.10 udpsrc port=9001 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)VP8-DRAFT-IETF-01, payload=(int)96, ssrc=(uint)2990747501, clock-base=(uint)275641083, seqnum-base=(uint)34810" ! rtpvp8depay ! vp8dec ! ffmpegcolorspace ! autovideosink
This is using the gstreamer-sdk-2013.6-universal.pkg package published from Gstreamer on OS X Mavericks.
REF: http://delog.wordpress.com/2011/04/14/stream-webm-video-over-rtp-with-gstreamer/
REF: http://delog.wordpress.com/2011/05/20/vp8-video-streaming-over-rtp-using-the-rtpbin-plugin-of-gstreamer/
I've tried it and it works on my machine (Arch linux).
Have you tried tweaking the ssrc and clock-base to make them equal ? Not sure if it can solve your problem because it works as-is on my machine.
Related
I have a third party application that reads data from a thermal camera and generates a RTP stream to a given UDP source. I am trying to wrap this RTP into a RTSP stream but I am running into problems...
The third party application basically runs gstreamer with this command
appsrc format=GST_FORMAT_TIME is-live=true block=true caps=video/x-raw,width=640,height=480,format=GRAY8,clock-rate=90000,framerate=10/1 ! openjpegenc ! rtpj2kpay ! udpsink host=127.0.0.1 port=3000
Using the command below I can visualize the stream on my machine
gst-launch-1.0 udpsrc port=3000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)JPEG2000, sampling=(string)GRAYSCALE, width=(int)640, height=(int)480, payload=(int)96" ! queue ! rtpj2kdepay ! openjpegdec ! videoconvert ! xvimagesink
However when trying to use the default RTP to RTSP application example using https://github.com/freedesktop/gstreamer-gst-rtsp-server/blob/master/examples/test-launch.c to just forward it with a RTSP container the connection fails with VLC. Command below:
./rtp-src-to-rtsp '( udpsrc port=3000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)JPEG2000, sampling=(string)GRAYSCALE, width=(int)640, height=(int)480, payload=(int)96" ! queue ! rtpj2kdepay ! rtpj2kpay )'
Any light on what I am doing wrong? VLC gives only a non-descriptive error
live555 error: Nothing to play for rtsp://{IP}:{PORT}/test
It might be a lack of support of J2K in VLC (I'm using revision 3.0.8-0).
Simulating your source with:
gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,framerate=10/1,format=GRAY8 ! openjpegenc ! rtpj2kpay ! udpsink host=127.0.0.1 port=3000
and relaying as RSTP with:
./test-launch "udpsrc port=3000 auto-multicast=0 ! application/x-rtp,encoding-name=JPEG2000,sampling=GRAYSCALE ! queue ! rtpj2kdepay ! image/x-jpc ! jpeg2000parse ! rtpj2kpay name=pay0 "
works on Linux with X using:
gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test ! application/x-rtp, encoding-name=JPEG2000,sampling=GRAYSCALE ! rtpj2kdepay ! jpeg2000parse ! openjpegdec ! videoconvert ! xvimagesink -v
Though, I haven't been able to receive with VLC, nor able to make a correct J2K/RTP SDP for VLC nor ffmpeg. Someone better skilled may further advise.
I have an issue in sending H264 streams to linphone.
When I play the following pipeline :
gst-launch-1.0 -v filesrc location=C:/ test.mp4 ! qtdemux ! avdec_h264 ! x264enc bitrate=192 ! rtph264pay ! udpsink host=127.0.0.1 port=9078
Everything is ok and the video played in linphone screen.
But what I want to do is to save first the video streams into a file then send this file to the linphone. What I did is the following:
Saving into a file
gst-launch-1.0 -v filesrc location=C:/ test.mp4 ! qtdemux ! avdec_h264 ! x264enc bitrate=192 ! filesink location=C: /videosample
Send to linphone:
gst-launch-1.0 -v filesrc location=C: /videosample ! h264parse ! rtph264pay ! udpsink host=127.0.0.1 port=9078
The packets received but the linphone played a black screen.
I want to know what I am missing in my pipelines or if there is a specific parameter to set.
Note that it is working when a play a gst receiver :
gst-launch-1.0 -v udpsrc port=9078 caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" ! rtph264depay ! avdec_h264 ! autovideosink
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'm trying to stream a h264 encoded movie file from a server to multiple clients at once by sending the RTP Stream to the broadcast address.
The solution I've got works but is very slow. Playing the video locally works fine.
Here's my Server:
gst-launch-0.10 -v filesrc location=/home/zeroc8/Videos/bunny.mov \
! qtdemux ! h264parse ! rtph264pay pt=96 ! udpsink host=192.168.1.255 port=5000
This is the Client:
gst-launch-0.10 udpsrc port=5000 \
caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"J01AHqkYGwe83gDUBAQG2wrXvfAQ\\,KN4JyA\\=\\=\", payload=(int)96, ssrc=(uint)786848209, clock-base=(uint)101553131, seqnum-base=(uint)64602"
! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink
Am I doing something bad here? Why is this so slow?
Just got the answer from the gstreamer mailing list.
In case anyone else is having the same problem, adding the gstrtpjitterbuffer element fixes it.
gst-launch-0.10 udpsrc port=5000 \
caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"J01AHqkYGwe83gDUBAQG2wrXvfAQ\\,KN4JyA\\=\\=\", payload=(int)96, ssrc=(uint)786848209, clock-base=(uint)101553131, seqnum-base=(uint)64602" \
! gstrtpjitterbuffer latency=1000
! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink
Here is what I'm trying:
gst-launch -v udpsrc port=1234 ! fakesink dump=1
I test with:
gst-launch -v audiotestsrc ! udpsink host=127.0.0.1 port=1234
And everything works fine, I can see the packages arriving from the audiotestsrc
Now lets test with the webcam source:
gst-launch -v v4l2src device=/dev/video0 ! queue ! videoscale method=1 ! "video/x-raw-yuv,width=320,height=240" ! queue ! videorate ! "video/x-raw-yuv,framerate=(fraction)15/1" ! queue ! udpsink host=127.0.0.1 port=1234
And nothing happens, no package appears in the dump.
Here is a logdump of what verbose shows in the server.
Does anyone have a clue on this?
Try these (You may have to install gstreamer-ugly plugins for this one)
UDP streaming from Webcam (stream over the network)
gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480' ! x264enc pass=qual quantizer=20 tune=zerolatency ! rtph264pay ! udpsink host=127.0.0.1 port=1234
UDP Streaming received from webcam (receive over the network)
gst-launch udpsrc port=1234 ! "application/x-rtp, payload=127" ! rtph264depay ! ffdec_h264 ! xvimagesink sync=false
Update
To determine the payload at the streaming end simply use verbose option with gst-launch -v ...
Maybe packets are too large for udp? They are limited to 64K. Try resizing frames to really small size to check if this is the reason. If so, you may be interested in some compression and payloaders/depayloaders (gst-inspect | grep pay).
gstreamer1-1.16.0-1.fc30
gst-launch-1.0 -v filesrc location=/.../.../.../sample-mp4-file.mp4 ! qtdemux ! h264parse ! queue ! rtph264pay config-interval=10 pt=96 ! udpsink port=8888 host=127.0.0.1
https://en.wikipedia.org/wiki/RTP_audio_video_profile