I'm currently working on Gstreamer and my goal is to take video from camera(coded natively in h264) decode it, then encode in AV1 and send it in udp to another computer on the network.
My pipelines currently are :
Server :
gst-launch-1.0 -v rtspsrc location= rtsp://192.168.33.104:8554/vis.0 latency=1 is-live=TRUE ! decodebin ! autovideoconvert ! x265enc tune=zerolatency bitrate=300 speed-preset=3 ! rtph265pay ! udpsink host=192.168.33.39 port=8123
Client :
gst-launch-1.0 udpsrc address=192.168.33.39 port=8123 ! application/x-rtp,media=video,clock-rate=90000,encoding-name=H265,payload=96 ! rtph265depay ! avdec_h265 ! autovideosink
So with h265 it works but i cannot find how to do it with AV1 because i can't find a rtpav1pay (and depay).
Thanks in advance.
I tried to search for rtpav1pay but found nothing. I tried rtpgstpay(and depay) didn't work. The main goal is to use as little as possible the network without lag so maybe it's not the best solution. If you have any other idea please share it.
There are rtpav1pay and rtpav1depay plugins provided by gst-plugins-rs; they can be built along with GStreamer if you enable the Rust plugins option, but you could also build them separately from their own repo (instructions on the README).
Related
I am very much new to the whole GStreamer-thing, therefore I would be happy if you could help me.
I need to stream a near-zero-latency videosignal from a webcam to a server and them be able to view the stream on a website.
The webcam is linked to a Raspberry Pi 3, because there are space-constraints on the mounting plattform. As a result of using the Pi I really can't transcode the video on the Pi itself. Therefore I bought a Logitech C920 Webcam, which is able to output a raw h264-stream.
By now I managed to view the stream on my windows-machine, but didn't manage to get the whole website-thing working.
My "achivements":
Sender:
gst-launch-1.0 -e -v v4l2src device=/dev/video0 ! video/x-h264,width=1920,height=1080,framerate=30/1 ! rtph264pay pt=96 config-interval=5 mtu=60000 ! udpsink host=192.168.0.132 port=5000
My understanding of this command is: Get the signal of video-device0, which is a h264-stream with a certain width, height and framerate. Then pack it into a rtp-package with a high enough mtu to have no artefacts and capsulate the rtp-package into a udp-package and stream in to a ip+port.
Receiver:
gst-launch-1.0 -e -v udpsrc port=5000 ! application/x-rtp, payload=96 ! rtpjitterbuffer ! rtph264depay ! avdec_h264 ! fpsdisplaysink sync=false text-overlay=false
My understanding of this command is: Receive a udp-package at port 5000. Application says it is a rtp-package inside. I don't know what rtpjitterbuffer does, but it reduces the latency of the video a bit.
rtph264depay says that inside the rtp is a h264-encoded stream. To get the raw data, which fpsdisplaysink understands we need to decode the h264 signal by the use of avdec_h264.
My next step was to change the receiver-sink to a local tcp-sink and output that signal with the following html5-tag:
<video width=320 height=240 autoplay>
<source src="http://localhost:#port#">
</video>
If I view the website I can't see the stream, but I can view the videodata, which arrived as plain text, when I analyse the data.
Am I missing a videocontainer like MP4 for my video?
Am I wrong with decoding?
What am I doing wrong?
How can I improve my solution?
How would you solve that problem?
Best regards
I'm trying to find a way to use VP8 or VP9 compressed video, a part of Googles WebM project with Gstreamer.
Is there a already a module that can handle VP8? If so, can I get some simple example of how to use it in a broadcast/receive over RTP?
So far there is nothing on the Gstreamer official documentation. They have Matroska support but that seems only be for demuxing the container.
Edit
There obviously are ways
Server:
gst-launch-0.10 -v v4l2src ! video/x-raw-yuv,width=640,height=480 ! vp8enc ! rtpvp8pay ! udpsink host=127.0.0.1 port=9001
Client:
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
But the latency is higher than I expected.
Yes VP8 is already supported..
VP9 is missing de/payloader more to that below (not really - gst 1.8 added support - details at bottom)
They are contained in vpx module.
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-plugin-vpx.html
VP8:
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-vp8enc.html
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-vp8dec.html
check with gst-inspect-1.0 vp8enc if you have it
For RTP you can use webmmux, rtpvp8depay, rtpvp8pay, vp8enc/dec etc..
However, as Burak Arslan stated pay/depayloader for RTP are not ready for now (not even in 1.6.1 I checked)
For the examples - post some pipe with that and we can check it when its not working :)
EDIT
GStreamer 1.8 was released with support for VP9 - new elements rtpvp9pay/rtpvp9depay added
I am trying to stream audio with the following GStreamer pipeline:
Server:
gst-launch-1.0 -v audiotestsrc ! audioconvert ! vorbisenc ! rtpvorbispay ! udpsink host=127.0.0.1 port=5000
Client:
gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp, media=audio, clock-rate=44100, encoding-name=VORBIS, encoding-params=1, payload=96" ! rtpvorbisdepay ! vorbisdec ! audioconvert ! autoaudiosink
I get the following message from GStreamer:
WARNING: from element /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not decode stream.
Additional debug info: gstrtpvorbisdepay.c(614): gst_rtp_vorbis_depay_process (): /GstPipeline:pipeline 0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not switch codebooks
And I don't get any sound on the client. Can anyone help?
[EDIT:]
When I copy-paste the caps from the server side... It works! But among those caps there is a configuration parameter which looks really ugly (link here). I noticed that if I just delete this parameter it doesn't work anymore. Moreover I used gst-inspect on udpsrc and rtpvorbisdepay elements and there is nothing about this parameter. Can someone explain me what this parameter corresponds to? Is there a way to avoid it?
I think this is Theora Vorbis thing.. those are some configuration parameters for initialization of decoder if I understand that properly..
Theora makes the same controversial design decision that Vorbis made to
include the entire probability model for the DCT coecients and all the quan-
tization parameters in the bitstream headers. This is often several hundred
elds. It is therefore impossible to decode any frame in the stream without
having previously fetched the codec info and codec setup headers.
~ from here
some similar question
I am trying to use latest gstreamer and x265enc together. I saw that someone have already posted some commits in http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/log/ext/x265/gstx265enc.c
Can anyone please give an example pipeline where it is known to working (gst-launch-1.0 pipeline example will be very helpful)
1)
What is the current status of x265enc plugin for gstreamer ? does it work really ?
Which branch of gstreamer I need to use to build x265enc? I want to build whole gsteamer source code which will be compatible with x265enc plugin.
What are the system requirement for x265enc and how to build it ? Any wiki/basic instructions will be very helpful.
My goal is to broadcast my ip cameras (h264 streams) as h265 stream on vaughnlive.tv
Currently, I am using following pipeline to broadcast in h264 format:
GST_DEBUG=2 gst-launch-1.0 flvmux name=mux streamable=true ! rtmpsink
sync=true location="rtmp://xxxxxxxxxxxx" rtspsrc
location="rtsp://xxxxxxx" caps="application/x-rtp,
media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA,
payload=(int)96" ! rtpmpadepay ! mpegaudioparse ! queue ! mad !
audioconvert ! queue ! voaacenc bitrate=128000 ! aacparse !
audio/mpeg,mpegversion=4,stream-format=raw ! mux. rtspsrc
location="rtsp://xxxxxxx"
caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,
encoding-name=(string)H264" ! rtph264depay !
video/x-h264,stream-format=avc,alignment=au,byte-stream=false ! queue
! decodebin ! queue ! videorate ! "video/x-raw,framerate=30/1" ! queue
! x264enc threads=4 speed-preset=ultrafast bitrate=3072 ! mux.
2)
Can anyone please suggest on how should I change this pipeline to broadcast in h265 format using x265enc element?
A little late but, maybe some people will find this question when seeking info about H.265 support in gstreamer nowadays. This is with gstreamer 1.6.1 compiled from source on Ubuntu 15.10 which has packages ready for libx265..
1,
Encoder
There is x265enc which will be enabled when we have library libx265-dev.
The encoder is inside gst-plugins-bad so after doing autogen.sh you should see x265enc enabled.
You may also need h265parse, rtph265pay/depay
Decoder
I see two decoders, dont know which one is working, I guess libde265dec there is also avdec_h265.
mux
For mux for x264 I was using mpegtsmux, but this does not support video/x265, some work has to be done. The matroskamux should be working when using filesink etc..
[16:39] hi, which container is suitable for x265enc, for x264enc I was using mpegtsmux?
[16:54] otopolsky: mpegts would work if you add support for h265 there, not very difficult[16:55] slomo_: so we need to just add the caps compatibility?
[16:55] otopolsky: otherwise, matroskamux supports it. mp4mux/qtmux could get support for it relatively easily too
[16:55] otopolsky: a bit more than that. look at what tsdemux does for h265
[16:56] otopolsky: and check the gst_mpegts_descriptor_from_registration related code in tsmux
[17:00] slomo_: thanks
2,
Questioned flvmux also does not support h265 only h264..
matroskamux cannot be used for streaming, so only way is to patch mpegtsmux or flvmux etc.
I have got one HD video "ed_hd.avi" on System#1. Would like to stream it over network and play the content from System#2. I am using GStreamer on Ubuntu 11.04, tried a lot on this. Variety of errors makes this objective difficult to diagnose. Will be thankful for getting a working command for the System#1-end and System#2-end.
What I have tried is as follows:
System #1:
gst-launch filesrc location=ed_hd.avi ! decodedin ! x263enc ! video/x-h264 ! rtph264pay ! udpsink host=127.0.0.1 port=5000
System #2:
gst-launch udpsrc port=5000 ! rtph264depay ! decodebin ! xvimagesink
Objective is : Convert avi file to raw video. Stream it from the second System#2.
Thank You.
Could you try the following
gst-launch filesrc location=ed_hd.avi ! decodebin ! ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=127.0.0.1 port=5000
If that doesn't work then proceed:
Did you try the following? You need to replace his audio bins with video bins
http://delog.wordpress.com/2011/06/01/stream-raw-vorbis-audio-over-udp-or-tcp-with-gstreamer/
Also take a look at
http://pastebin.com/PtD21Bx7
Here replace v4l2src with your video src
Also
https://metalab.at/wiki/Gstreamer_One_Liners
I think your problem is the 127.0.0.1 portion. That is a loopback address (check ifconfig lo0 to see the Link encap:Local Loopback for the 127.0.0.1 address). This won't work across two systems, though it might work fine on a single system.
Instead, use the address that is publicly visible for the second machine; check ip addr show or ifconfig output to find the address. Write the actual address for System #2 in the command line on System #1.