I stream h264 contents as rtsp from a filesrc to a udpsink
gst-launch-0.10 filesrc location=./test.movie ! h264parse ! rtph264pay ! udpsink host=127.0.0.1 port=9004
This pipeline works well but gives a quite high CPU load and my question is if there is any way of minimizing the CPU load, limiting the processing rate?
Feels like since there is a filesrc and udpsink there is really nothing stopping gstreamer from just working as quick as it can. A 35sec movie takes <0,5 seconds for gstreamer to stream with the given pipeline.
Related
I'm working on a video streaming wearable device. During the tests, it came up that the pipeline clock and stream stop while fast walking or running. It's bizarre behaviour because in debug messages there are no errors about the broken pipeline, besides lost frames. It's frizzed and only restarting help. May you guys guess what causes the problem?
The pipelines I use:
streaming device:
gst-launch-1.0 -vem --gst-debug=3 v4l2src device=/dev/video0 ! video/x-raw,width=640,height=480,framerate=\(fraction\)30/1 ! v4l2h264enc extra-controls=s,video_bitrate=250000 capture-io-mode=4 output-io-mode=4 ! "video/x-h264,level=(string)4" ! rtph264pay config-interval=1 ! multiudpsink clients="127.0.0.1:5008,10.123.0.2:5008"
client:
udpsrc port=5008 do-timestamp=true ! application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96 ! rtpjitterbuffer latency=100 drop-on-latency=true drop-messages-interval=100000000 ! queue max-size-buffers=20000 ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! glupload ! qmlglsink name=qmlglsink sync=false
The hardware I use is a PS3 Eye cam, and LTE modem to transmit video with a pretty low uplink of 1-2 Mbit/s, and everything running on RaspberryPi 3b+ 1GB.
For more debug info there are also pictures of the log file after last registered dropped frame and every next "cycle" sends a new query, loops over GST Element from sink to the source which is my camera and ends with max query duration(highlighted query to v4l2src)
Do you know how to overcome this problem?
The problem has been resolved. The issue was not variable encoder bitrate.
A more detailed inspection and pipeline that works for me is in this GStreamer issue page
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The objective I am trying to achieve is streaming 1080p video from Raspberry pi camera and record the video simultaneously.
I tried recording the http streaming as source but didn't work on 30fps. A lot of frames were missing and almost got 8fps only.
As a second approach, I am trying to record the file directly from camera and then streaming the "recording in progress/buffer" file. For the same I am trying to use GStreamer. Please suggest if this is good option or should I try any other?
For Recording using GStreamer I used
gst-launch-1.0 -v v4l2src device=/dev/video0 ! capsfilter caps="video/x-raw,width=1920,height=1080,framerate=30/1" !
videoflip method=clockwise ! videoflip method=clockwise ! videoconvert ! videorate ! x264enc! avimux ! filesink location=test_video.h264
Result : recorded video shows 1080p and 30fps but frames are dropping heavily.
For Streaming the video buffer I have used UDP in Gstreamer as,
gst-launch-1.0 -v v4l2src device=/dev/video0 ! capsfilter caps="video/x-raw,width=640,height=480,framerate=30/1" ! x264enc ! queue ! rtph264pay ! udpsink host=192.168.5.1 port=8080
Result : No specific errors on terminal but can't get stream on vlc.
Please suggest the best method here.
I am developing an application where I am using a wave file from a location at one end of a pipeline and udpsink at the other end of it.
gst-launch-1.0 filesrc location=/path/to/wave/file/Tornado.wav ! wavparse ! audioconvert ! audio/x-raw,channels=1,depth=16,width=16,rate=44100 ! rtpL16pay ! udpsink host=xxx.xxx.xxx.xxx port=5000
The Above wave file is having sampling rate = 44100 Hz and single-channel(mono)
On the same PC I am using a c++ program application to catch these packets and depayload to a headerless audio file (say Tornado.raw)
The pipeline I am creating for this is basically
gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay ! filesink location=Tornado.raw
Now This works fine. I get the headerless data and when I play it using the Audacity It plays great!
I am trying to resample this audio file while it is in pipeline from 44100 Hz to 8000 Hz
Simply changing the clock-rate=(int)44100 to clock-rate=(int)8000 is not helping (also absurd logically)
I am looking for how to get the headerless file at the pipeline output with 8000 Hz sampling.
Also the data that I am getting now is Big-endian, but I want Little-endian as output. how do I set that in the pipeline?
You might relate this to one of my earlier question.
First, you have some weird caps in your pipeline - width and height are for video here. They probably will be just ignored.. but still.. not sure on others there as well but meh..
For the actual question. Just use audioresample and audioconvert elements of Gstreamer to transfer in your desired format.
E.g.
[..] ! rtpL16depay ! audioresample ! audioconvert ! \
audio/x-raw, rate=8000, format=S16LE ! filesink location=Tornado.raw
I am on gstreamer 1.2.4.
I have developed a video sink based on xvimagesink.
When I set the sync property of the sink to 0 the pipeline hangs after displaying a few frames.
I used the following pipeline
gst-launch-1.0 filesrc location=test.mp4 ! qtdemux ! h264parse ! avdec_h264 ! myvideosink sync=0
I noticed that even the decoder stops receiveing frames.
This issue occurs even with videotestsrc
gst-launch-1.0 videotestsrc ! myvideosink sync=0
This does not occur with xvimagesink.
What could be the problem?
I have h264 video track and aac audio track inside mp4 container and I want to play it, but when I run my pipeline there's just first frame shown and no sound.
Here's my pipeline:
gst-launch filesrc location=/home/dmitry/Downloads/big_buck_bunny.mp4 ! qtdemux name=demux \
demux.audio_00 ! queue ! faad ! audioconvert ! audioresample ! autoaudiosink \
demux.video_00 ! queue ! ffdec_h264 ! ffmpegcolorspace ! autovideosink
Your queues might not be large enough for this scenario. You should try using playbin2 or decodebin for decoding and it will automatically adjust the queue sizes for playback.
If you have to stick to this pipeline, try setting larger values to the max-size-* properties on the queues.
On a plus side: please move to 1.2 version, 0.10 is obsolete for 2 years now.