I am trying to take snapshot from analog camera for every 10 sec, it worked fine.
command:
gst-launch-1.0 v4l2src device=/dev/video0 ! queue ! vspmfilter !
video/x-raw,width=640,height=480 ! videorate !
video/x-raw,width=640,height=480,framerate=1/10 ! jpegenc quality=30 !
multifilesink location=/home/root/images/image_%d.jpg
I am getting images in the specified directory like
image0.jpg
image1.jpg
...
But I want image to be saved with time stamp after every 10 sec like image_yymmddhhmmss for example
image_20180817104333.jpg
image_20180817104343.jpg
....
How I can achieve this with above command?
I guess you could use a pad probe for the multifilesink. And whenever a buffer is received you set a new desired location property on the sink. Not sure though if that property can be set while in PLAYING state though.
Related
I'm a newbie to gstreamer so i would be appreciated if you could help me.
I'm trying to listen to a pipeline and record frames to a file.
I have tried the following pipeline:
gst-launch-1.0 udpsrc port=5600 do-timestamp=true ! application/x-rtp, payload=96 ! rtph264depay ! avdec_h264 ! clockoverlay ! jpegenc ! avimux ! filesink location=stream.avi
I want to record whole timeline even if the sender doesn't provide any frame data.
In default, recorder appends the frames when pipeline receive some valid frames. But I want to see some black frames when sender doesn't send data.
I experimented a bit and I don't think you'll be able to do this with a plain gst-launch command. Unfortunately what it would probably involve is to write an application that detects when packets/buffers are not coming in any more, and then modifying the pipeline. If you want to give it a go I'd suggest the input-selector element in something like this:
gst-launch-1.0 videotestsrc pattern=black ! video/x-raw ! input-selector name=selector ! clockoverlay ! jpegenc ! avimux ! filesink location=stream.avi
Then I'd create a method to attach the stream to the input-selector:
udpsrc port=5600 do-timestamp=true ! application/x-rtp, payload=96 ! rtph264depay ! avdec_h264 ! identity name=buffer-checker
To detect no packets coming in, you can listen for the handoff signal on the identity element, and then remove the stream when it times out and switch over to the black test pattern from the videotestsrc by using the active-pad property on the input-selector.
Using the videomixer element almost works, but I don't believe it will handle multiple stops and starts of the stream.
Anyway, hope someone else comes up with a better idea. You could also re-analyze your top level approach and see if there is a way you can work with multiple video clips instead of the one.
I have two AXIS IP cameras streaming H264 stream over RTSP/RTP. Both cameras are set to synchronize with same NTP server so I assume both cameras will have same exact clock (may be minor diff in ms).
In my application, both cameras are pointing to same view and its required to process both camera images of same time. Thus, I want to synchronize the image capture using GStreamer.
I have tried invoking two pipelines separately on different cmd prompts but the videos are 2-3 seconds apart .
gst-launch rtspsrc location=rtsp://192.168.16.136:554/live ! rtph264depay ! h264parse ! splitmuxsink max-size-time=100000000 location=cam1_video_%d.mp4
gst-launch rtspsrc location=rtsp://192.168.16.186:554/live ! rtph264depay ! h264parse ! splitmuxsink max-size-time=100000000 location=cam2_video_%d.mp4
Can someone suggest a gstreamer pipeline to synchronize both H264 streams and record them into separate video files?
Thanks!
ARM
I am able to launch a pipeline using gst-launch as shown below. It shows good improvement on captured frame synchronization compare to lanuching two pipelines. Most times they differ by 0-500 msec. Though, I still want to synchronize them less than 150 msec accuracy.
rtspsrc location=rtsp://192.168.16.136:554/axis-media/media.amp?videocodec=h264 \
! rtph264depay ! h264parse \
! splitmuxsink max-size-time=10000000000 location=axis/video_136_%d.mp4 \
rtspsrc location=rtsp://192.168.16.186:554/axis-media/media.amp?videocodec=h264 \
! rtph264depay ! h264parse \
! splitmuxsink max-size-time=10000000000 location=axis/video_186_%d.mp4
Appreciate if someone can point other ideas!
~Arm
What do you mean synchronize? if you record to separate video files you do not need any synchronization.. as this is going to totaly separate them.. each RT(S)P stream will contain different timestamps, if you want to align them somehow to the same time (I mean real human time.. like "both should start from 15:00") then you have to configure them this way somehow (this is just idea)..
Also you did not tell us whats inside those rtp/rtsp streams (is it MPEG ts or pure IP.. etc). So I will give example of mpeg ts encapsulated rtp streams.
We will go step by step:
Suppose this is one camera just to demonstrate how it may look like:
gst-launch-1.0 -v videotestsrc ! videoconvert ! x264enc ! mpegtsmux ! rtpmp2tpay ! udpsink host=127.0.0.1 port=8888
Then this would be reciever (it must use rtmp2tdepay. We are encapsulating metadata inside MPEG container):
gst-launch-1.0 udpsrc port=8888 caps=application/x-rtp\,\ media\=\(string\)video\,\ encoding-name\=\(string\)MP2T ! rtpmp2tdepay ! decodebin ! videoconvert ! autovideosink
If you test this with your camera .. the autovideosink means that new window will popup displaying your camera..
Then you can try to store it inside file.. we will use mp4mux..
So for same camera input we do:
gst-launch-1.0 -e udpsrc port=8888 caps=application/x-rtp\,\ media\=\(string\)video\,\ encoding-name\=\(string\)MP2T ! rtpmp2tdepay ! tsdemux ! h264parse ! mp4mux ! filesink location=test.mp4
Explanation: We do not decode and reencode(waste of processing power) so I will just demux the MPEG ts stream and then instead of decoding H264 I will just parse it for the mp4mux which accepts video/x-h264.
Now you could use the same pipeline for each camera.. or you can just copypaste all elements into the same pipeline..
Now as you did not provide any - at least partial - attempt to make something out this is going to be your homework :) or make yourself more clear about the synchronization as I do not understand it..
UPDATE
After your update to question this answer is not very useful, but I will keep it here as reference. I have no idea how to synchronize that..
Another advise.. try to look at timestamps after udpsrc.. maybe they are synchronized already.. in that case you can use streamsynchronizer to synchronize two streams.. or maybe video/audio mixer:
gst-launch-1.0 udpsrc -v port=8888 ! identity silent=false ! fakesink
This should print the timestamps (PTS, DTS, Duration ..):
/GstPipeline:pipeline0/GstIdentity:identity0: last-message = chain ******* (identity0:sink) (1328 bytes, dts: 0:00:02.707033598, pts:0:00:02.707033598, duration: none, offset: -1, offset_end: -1, flags: 00004000 tag-memory ) 0x7f57dc016400
Compare PTS of each stream.. maybe you could combine two udpsrc in one pipeline and after each udpsrc put identity (with different name=something1) to make them start reception together..
HTH
I'm trying to use a jpg-File as a virtual webcam for Skype (or similar). The image file is reloading every few seconds and the Pipeline should also transmit always the newest image.
I started creating a Pipeline like this
gst-launch filesrc location=~/image.jpg ! jpegdec ! ffmpegcolorspace ! freeze ! v4l2sink device=/dev/video2
but it only streams the first image and ignores the newer versions of the image file. I read something about concat and dynamically changing the Pipeline but I couldn't get this working for me.
Could you give me any hints on how to get this working?
Dynamic refresh the input file is NOT possible (at least with filesrc).
Besides, your sample use freeze, which will prevent the image change.
One possible method is using multifilesrc and videorate instead.
multifilesrc can read many files (with a provided pattern similar to scanf/printf), and videorate can control the speed.
For example, you create 100 images with format image0000.jpg, image0001.jpg, ..., image0100.jpg. Then play them continuously, with each image in 1 second:
gst-launch multifilesrc location=~/image%04d.jpg start-index=0 stop-index=100 loop=true caps="image/jpeg,framerate=\(fraction\)1/1" ! jpegdec ! ffmpegcolorspace ! videorate ! v4l2sink device=/dev/video2
Changing the number of image at stop-index=100, and change speed at caps="image/jpeg,framerate=\(fraction\)1/1"
For more information about these elements, refer to their documents at gstreamer.freedesktop.org/documentation/plugins.html
EDIT: Look like you use GStreamer 0.10, not 1.x
In this case, please refer to old documents multifilesrc and videorate
You can use a general file name with multifilesrc if you add some parameter adjustments and pair it with an identity on a delay. It's a bit fragile but it'll do fine for a temporary one-off program as long as you keep your input images the same dimensions and format.
gst-launch-1.0 multifilesrc loop=true start-index=0 stop-index=0 location=/tmp/whatever ! decodebin ! identity sleep-time=1000000 ! videoconvert ! v4l2sink
I need to capture a video using a webcam and output a single image for each video frame captured.
I have tried using gstreamer with a multifilesink, e.g.:
gst-launch v4l2src device=/dev/video1 ! video/x-raw-yuv,framerate=30/1 ! ffmpegcolorspace ! pngenc ! multifilesink location="frame%d.png"
However, this does not actually output every frame, meaning that if I record for 2 seconds at 30 fps, I don't get 60 images. I'm assuming this is because the encoding can't go that fast, so I need another method.
I figured it might work if I have one pipeline capture a video, and a separate pipeline convert that video to frames, but I don't know enough about codecs. Do I need to encode the video to a file like h264 or mp4 just to then decode it again?
Does anyone have any thoughts or suggestions? Keep in mind that I need to be able to do this in code, not using an application like Adobe Premiere, for example.
Thanks!
You could simply add a queue in there like this:
gst-launch v4l2src device=/dev/video1 ! video/x-raw-yuv,framerate=30/1 ! queue ! ffmpegcolorspace ! pngenc ! multifilesink location="frame%d.png"
This should make sure the video-capture is allowed to run at 30 fps, and then writing it to
disk can happen in its own tempo. Just be aware that the queue will grow to quite a large size
if you leave this setup for too long.
the solution I have to offer doesn't use gstreamer but ffmpeg. I hope that's fine for you too.
As described in this forum post, you can use something like this:
ffmpeg -i movie.avi frame%d.png
to get a png/jpg image for each frame of the video.
But depending on the input file you use, you might have to convert it to an MPEG vid before running ffmpeg.
Note:
If you want leading zeroes in your image file names, use %05d instead (for 5-digit numbers, like in C's printf()):
ffmpeg -i movie.avi frame%05d.png
The output file format depends on the file extension, so you might use .jpg, .bmp, ... instead of .png.
I ended up doing this in two parts.
Write video to file.
gst-launch v4l2src device=/dev/video2 ! video/x-raw-yuv,framerate=30/1 ! xvidenc ! queue ! avimux ! filesink location=test.avi
Post process.
gst-launch-1.0 --gst-debug-level=3 filesrc location=test.avi ! decodebin ! queue ! autovideoconvert ! pngenc ! multifilesink location="frame%d.png"
My pipe line is simply trying to mux an audiotestsrc with a videotestsrc and output to a filesink.
videotestsrc num-buffers=150 ! video/x-raw-yuv,width=1920, height=1080 !
timeoverlay ! videorate ! queue ! xvidenc ! avimux name=mux mux.
! filesink sync=true location=new.avi
audiotestsrc num-buffers=150 !
queue ! audioconvert ! audiorate ! mux.
new.avi is produced.
Video is exactly 5 seconds long as expected
Audio is about 3.5 seconds long and the remaining 1.5 seconds is
slient.
What am I missing here? I've tried every combination of sync="" properties, etc.
What pipeline would generate a test clip with autotestpattern and videotest pattern muxed together where audio and video are the same duration?
Thanks
audiotestsrc num-buffers=150
By default each buffer contains 1024 samples: samplesperbuffer
Which means you are generating 150*1024=153600 samples.
Asuming 44.1kHz, the duration would be 153600/44100=3.48 seconds.
So if you need 5 seconds audio, you need 5*44100=220500 samples. with samplesperbuffer==1024, this means 220500/1024=215.33 buffers. (ie 215 or 216 buffers).
It would be easier if you set samplesperbuffer to 441, then you need exactly 100 buffers for every second audio:
audiotestsrc num-buffers=500 samplesperbuffer=441
You can make use of the blocksize roperty of audiotesrc to match the duration of a frame. this is in bytes and thus you might want to use a caps filter after audiotestsrc to select a sampling-rate and sample format.