I have been trying to get gstreamer rtp working on my osx lion I have following:
sender:
gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay ! udpsink -vvv
receiver:
gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string)000001b001000001b58913000001000000012000c48d8800f50a041e1463000001b24c61766335332e33352e30, payload=(int)96, ssrc=(uint)1613325455, timestamp-offset=(uint)3778351903, seqnum-offset=(uint)5013" ! rtpmp4vdepay ! avdec_mpeg4 ! autovideosink
but all i get is a green screen on the receiver. Am i setting up my pipelines incorrectly?
I tried using the tcpserversink and tcpclientsrc as follows:
gst-launch-1.0 videotesavenc_mpeg4 ! rtpmp4vpay config-interval=2 ! tcpserversink port=5555 -vvv --gst-debug=3
gst-launch-1.0 tcpclientsrc port=5555 ! "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string)000001b001000001b58913000001000000012000c48d8800f50a041e1463000001b24c61766335332e33352e30, payload=(int)96, ssrc=(uint)970028597, timestamp-offset=(uint)2609711508, seqnum-offset=(uint)59228" ! rtpmp4vdepay ! decodebin ! videoconvert ! autovideosink --gst-debug=3 -vvv
which results in a flickering test source and the following error repeated on the client side:
0:00:06.213137000 10134 0x7fab6b051680 ERROR libav :0:: Error at MB: 311
0:00:06.214311000 10134 0x7fab6b051680 ERROR libav :0:: ac-tex damaged at 19 14
can someone construct a test pipeline to output the videotestsrc to mpeg4 video with both a sender and receiver please??
Try setting "host" for the udpsink like this:
gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay ! udpsink host=127.0.0.1 -vvv
Related
I'm trying to play audio (WAV) file using Gstreamer udpsrc element with no success. Playing audio from a file works just fine:
gst-launch-1.0 -v filesrc location=sp5.wav ! wavparse ! audioconvert ! volume volume=1 ! autoaudiosink
But when running this in one terminal:
gst-launch-1.0 -v udpsrc port=5200 ! wavparse ! audioconvert ! volume volume=1 ! autoaudiosink
and this in another:
$ gst-launch-1.0 -v filesrc location=sp5.wav ! udpsink host=localhost port=5200
can't get any audio out (and Wireshark shows packets are going across). Trying the last command from a different PC didn't make any difference.
Running Gstreamer version 1.14.4.
Am i missing something obvious?
Currently, i'm using OpenCV 4.2 C++ in order to encode and stream outputs from a Allied Vision Manta camera.
I already can grabe a frame, compress with MJPEG and the stream to other pipeline using OpenCV and GStreamer.
However, i need to try with H.264 (and later with H.265) but it's not working. The pipelines created are these in VideoWriter function are the follow:
1 - "appsrc ! queue ! videoconvert ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5015";
2 - "appsrc ! autovideoconvert ! videoconvert ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5015";
With the 1st pipe, i get:
[ WARN:0] global /opt/opencv/modules/videoio/src/cap_gstreamer.cpp (1759) handleMessage OpenCV | GStreamer warning: Embedded video playback halted; module x264enc0 reported: Can not initialize x264 encoder.
[ WARN:0] global /opt/opencv/modules/videoio/src/cap_gstreamer.cpp (1665) writeFrame OpenCV | GStreamer warning: Error pushing buffer to GStreamer pipeline
With the 2nd pipe, I get:
libva info: VA-API version 0.39.0
libva info: va_getDriverName() returns 0
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/vmwgfx_drv_video.so
libva info: va_openDriver() returns -1
The others arguments of VideoWritter are:
cv::CAP_GSTREAMER, 0, 5, Size(1080, 720), true
The input pixel format is BGR, because i need to grab the frames with BayerRG8, and then convert to BGR to minimize the traffic.
The receiver is done with
udpsrc port=5015 ! application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264,framerate=15/1 ! rtph264depay ! decodebin ! videoconvert ! appsink
One detail that i don't know if it's important or not, i'm doing this in the VirtualBox.
Hello I am trying to encode and decode a "live" video using the webcam of and windows pc. The gstreamer version that I am currently using is 1.15.1
For the encoding line I use:
gst-launch-1.0 ksvideosrc ! \
'video/x-raw, width=640, height=480, framerate=30/1' ! \
videoconvert ! \
x264enc pass=qual quantizer=20 tune=zerolatency ! \
rtph264pay ! \
filesink location=D:\\cam.ts
This make a file on the D drive, but when is try to decode it I get an error.
line used:
gst-launch-1.0 filesrc location=D:\\cam.ts ! \
rtph264depay ! \
avdec_h264 ! \
videoconvert ! \
autovideosink sync=false
This is the answer Gstreamer gives me
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data stream error.
Additional debug info:
../libs/gst/base/gstbasesrc.c(3064): gst_base_src_loop ():
/GstPipeline:pipeline0/GstFileSrc:filesrc0:
streaming stopped, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
I am currently getting the following error
WARNING: erroneous pipeline: no element "Qtdemux" when I run the following command
gst-launch-1.0 -v filesrc location=~/Desktop/Dog.mp4 ! Qtdemux ! h264parse ! ffdec_h264 ! ffmpegcolorspace ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5000
I have tried making sure I have gstreamer-plugin-good and get the following response
gstreamer1.0-plugins-good is already the newest version (1.8.3-1ubuntu0.4).
However when I look at gst-inspect-1.0 Qtdemux I get the following response No such element or plugin 'Qtdemux'.
Any thoughts on what may be the problem I am facing?
It is qtdemux, with a samll q.
I'm working on some HLS app with latest version(1.2.0) of GStreamer on Ubuntu 13.10
and be troubled by hlssink plugin.
While getting segments from a ts file, the cmd as follow :
gst-launch-1.0 filesrc location=test.ts ! tsdemux name=demuxer ! multiqueue name=mq ! 'video/x-h264,stream-format=byte-stream,alignment=nal' ! mpegtsmux name=mux ! hlssink mq. ! 'audio/mpeg,mpegversion=2,stream-format=adts' ! mux.
I found it dumps the segments ts files with wrong type, and no player can play back those files.
But when I demux from mp4 format, it works :
gst-launch-1.0 filesrc location=test.mp4 ! qtdemux name=demuxer demuxer. ! multiqueue name=mq ! h264parse ! mpegtsmux name=mux ! hlssink demuxer. ! mq. mq. ! aacparse ! mux.
really troubled a lot by this. Can anyone help me?