For bandwidth reasons I've modified the slice-header spacing to use more slices per I-frame, this causes tearing on the receiving end.
The problem appears as if individual slices are getting decoded without an entire I-frame getting buffered up for the omxh264dec? This is a bit strange as the Tegra decoder is supposed to only work on a frame level..
Perhaps this problem can be alleviated by correct synchronization of GstBuffer:s on the receiving end?
Repro case: (Jetson TX2)
# Sender:
gst-launch-1.0 nvcamerasrc ! 'video/x-raw(memory:NVMM), width=(int)1280, height=(int)720, format=(string)I420, framerate=(fraction)60/1' ! nvvidconv flip-method=0 ! 'video/x-raw(memory:NVMM), format=(string)I420' ! omxh264enc iframeinterval=1 bit-packetization=TRUE slice-header-spacing=450000 control-rate=2 preset-level=0 profile=1 qp-range=-1,-1:10,10:-1,-1 ! 'video/x-h264, stream-format=(string)byte-stream' ! h264parse ! rtph264pay mtu=60000 ! udpsink host=127.0.0.1 port=5000
# Reciever:
gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,encoding-name=H264,payload=96" ! rtph264depay ! h264parse ! omxh264dec ! videoconvert ! xvimagesink async=TRUE sync=TRUE
As Florian Zwoch suggested an rtpjitterbuffer solves this issue.
Related
I want to stream a h264 video over UDP to another pc.
I am using this pipeline to produce the stream:
videotestsrc ! video/x-raw,width=400,height=400,framerate=7/1 ! videoconvert ! x264enc ! h264parse config-interval=1 ! video/x-h264,stream-format=byte-stream,alignment=nal ! rtph264pay ! udpsink host=192.168.1.100 port=2705
I can play this on the same machine (with ip address 192.168.1.100) with this pipeline:
udpsrc port=2705 ! application/x-rtp,width=400,height=400,encoding-name=H264,payload=96,framerate=7/1 ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink
But when I try to stream it from another pc to the same machine I get only this output and it waits forever:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Redistribute latency...
Redistribute latency...
What can be the problem here?
I found the solution. A videoconvert element is needed in the playing pipeline.
The working playing pipeline is:
udpsrc port=2705 ! application/x-rtp,width=400,height=400,encoding-name=H264,payload=96,framerate=7/1 ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! autovideosink
I am trying to make UDP Multicast screen streaming using GStreamer. My screen casting server should run on Windows and my client should run on Linux.
If I start the client before the server, everything is fine.
The problem is when I start the client and the server had already been launched. The video is received, but it is terribly distorted.
This is the result.
Server:
gst-launch-1.0 -e gdiscreencapsrc ! queue ! video/x-raw, framerate=25/1 ! videoconvert ! \
x264enc noise-reduction=10000 tune=zerolatency bitrate=2500 speed-preset="fast" byte-stream=true threads=4 key-int-max=15 intra-refresh=true ! \
h264parse ! rtph264pay config-interval=1 \
! udpsink host=224.1.1.1 port=5000 auto-multicast=true
Client:
gst-launch-1.0 -v udpsrc multicast-group=224.1.1.1 auto-multicast=true port=5000 ! application/x-rtp ! rtph264depay ! h264parse ! queue ! decodebin ! videoconvert ! autovideosink caps='video/x-raw, format=RGB'
I have already tried using dx9screencapsrc, but the behaviour is the same.
The issue is fixed only if I replace gdiscreencapsrc with videotestsrc.
If I launch the server on Linux, using ximagesrc, I still have some issues, but the video is improving over time.
Any help would be appreciated!
Adding cabac=false to my x264enc element fixed the ussue.
gst-launch-1.0 -v gdiscreencapsrc ! queue ! video/x-raw,framerate=60/1 ! decodebin ! videoscale ! videoconvert ! \
x264enc cabac=false tune=zerolatency bitrate=4000 speed-preset="fast" ! \
h264parse ! rtph264pay config-interval=-1 \
! udpsink host=224.1.1.1 port=5000 auto-multicast=true sync=false
I'm trying to stream a video with h264. Source is a Axis camera. I managed to stream jpeg with multicast but not h264.
With jpeg I used following command:
gst-launch-1.0 udpsrc uri=udp://239.194.0.177:1026 ! application/x-rtp,encoding-name=JPEG,payload=26 ! rtpjpegdepay ! jpegdec ! autovideosink
I tried to stream h264 but it fails, used following command:
gst-launch-1.0 -v udpsrc host=239.194.0.177 port=1026 ! rtph264depay ! ffdec_h264 ! xvimagesink
I get the following error:
ERROR: pipeline could not be constructed: no element "udpsrc".
With this line:
gst-launch-1.0 udpsrc uri=udp://239.194.0.177:1026 ! application/x-rtp,media=video,clock-rate=90000,encoding-name=H264 ! rtph264depay ! h264parse
I did not get any errors but no video streamed and this was printed in terminal:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
I tried the commands from following pages:
Stream H.264 video over rtp using gstreamer
https://developer.ridgerun.com/wiki/index.php/Using_UDP_Multicast_with_GStreamer
http://labs.isee.biz/index.php/Example_GStreamer_Pipelines#H.264_RTP_Streaming
But could not get it to work.
When running in verbos mode I get litte more info.
Command:
gst-launch-1.0 -v udpsrc uri=udp://239.194.0.177:1026 ! application/x-rtp, media=video, payload=96, encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! fakesink
Output:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ payload\=\(int\)96\,\ encoding-name\=\(string\)H264\,\ clock-rate\=\(int\)90000"
/GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0.GstPad:sink: caps = "application/x-rtp\,\ media\=\(string\)video\,\ payload\=\(int\)96\,\ encoding-name\=\(string\)H264\,\ clock-rate\=\(int\)90000"
How do I stream H264 via multicast with gstreamer?
Too long for comment - and since nobody is answering posting this draft of thoughts as answer..
The first error about no element udpsrc is really weird. But I think its complaining about missing uri parameter. What version are you using? I do not have the host parameter for udpsrc..
In third pipeline it ends with h264parse - is this s typo? you need to decode the h264.. not just parse it:
gst-launch-1.0 udpsrc uri=udp://239.194.0.177:1026 ! application/x-rtp,media=video,clock-rate=90000,encoding-name=H264 ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! autovideosink
Also add some logs (maybe with pastebin if too long) with running GST_DEBUG=3 gst-launch-1.0 .... or so.
What does it mean:
But could not get it to work
This does not say too much ;)
Usually when working with rtp you need to provide really all capabilities otherwise it may not link or play at all..
Maybe try with uridecodebin? Not sure if its the best idea:
gst-launch-1.0 uridecodebin uri=udp://etcetc:port ! videoconvert ! autovideosink
If you get any new infos/questions add them as updates to make the picture whole (for others as well..)
HTH
I would like to capture a video stream (+audio) in MJPEG from my webcam into .mts container using this pipeline:
gst-launch-1.0 v4l2src do-timestamp=true device=/dev/video0 \ !
'image/jpeg,framerate=30/1,width=1280,height=720' ! videorate \
! queue ! mux2. pulsesrc do-timestamp=true \
device="alsa_input.pci-0000_00_1b.0.analog-stereo" ! \
'audio/x-raw,rate=88200,channels=1,depth=24' ! audioconvert ! \
avenc_aac compliance=experimental ! queue ! \
mux2. mpegtsmux name="mux2" ! filesink location=/home/sina/Webcam.mts
it seems that my pipeline doesn't recognize the mpegtsmux (?)
when i use avimux or even matroskamux it works but as far as I know for MPEG-TS I need to use the correct muxer which is "mpegtsmux"
This is the warning:
WARNING: erroneous pipeline: could not link queue0 to mux2
Can you please tell me what part of my pipeline is wrong? or what shall I change in order to get a timestamped video stream at the end (duration of the video must be shown when I play it via kdenlive or VLC)?
Best,
Sina
I think you are missing some encoder before mux.
Just try this without audio(added x264enc):
gst-launch-1.0 v4l2src device=/dev/video0 ! videorate ! queue ! x264enc ! mpegtsmux name="mux2" mux2. ! filesink location=bla.mts
The warning you are getting is saying it clearly.. it cannot link mux because the mux does not support capabilities image/jpeg.. just check the Capabilities section of sink pad with command:
gst-inspect-1.0 mpegtsmux
But it supports for example video/x-h264 - therefore the need for x264enc
I have the following command I am using but somehow its not letting me send it over network(local). if I view local it works but not when I try with IP-address.
gst-launch-0.10 v4l2src device=/dev/video0 ! video/x-raw-yuv,width=320,height=240 ! videobox left=-320 border-alpha=0 ! queue ! videomixer name=mix ! ffmpegcolorspace ! xvimagesink v4l2src device=/dev/video1 ! video/x-raw-yuv,width=320,height=240 ! videobox left=1 ! queue ! send-config=true ! udpsink host=127.0.0.1 port=5000
this gives me error:
WARNING: erroneous pipeline: link without source element
but without the udp it works fine.
gst-launch-0.10 v4l2src device=/dev/video0 ! video/x-raw-yuv,width=320,height=240 ! videobox left=-320 border-alpha=0 ! queue ! videomixer name=mix ! ffmpegcolorspace ! xvimagesink v4l2src device=/dev/video1 ! video/x-raw-yuv,width=320,height=240 ! videobox left=1 ! queue ! mix.
my client side is this:
gst-launch udpsrc uri=udp://127.0.0.1:5000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string)000001b001000001b58913000001000000012000c48d88007d0a041e1463000001b24c61766335322e3132332e30, payload=(int)96, ssrc=(uint)298758266, clock-base=(uint)3097828288, seqnum-base=(uint)63478" ! rtpmp4vdepay ! ffdec_mpeg4 ! autovideosink
what am I doing wrong? Any help would be great.
The warning is the reason you are not able to send:
"queue ! send-config=true ! udpsink" is the 'link without source element'
What is send-config=true? Isn't this a property for some element you didn't type there?