I found different articles on changing the fps with ffmpeg but none of them is matching for my exact purposes.
There is an ffmpeg command like below:
ffmpeg -i RTSPCAMERAPRODUCEH264 -c:v copy -an -movflags +frag_keyframe+empty_moov -f mp4
This will remux my camerastream to fragmented mp4 perfectly.
Is there a way to force ffmpeg to lower the FPS to save bandwidth?
I.e. camera streams 30fps, it needs 1Mbps for fmp4 (sample numbers!):
I'd like to know if it's possible to lower the FPS and get an output stream for which 500kbps (50% of original is enough) without re-encoding.
ffmpeg -r 1 -i RTSPCAMERAPRODUCEH264 -c:v copy -an -movflags +frag_keyframe+empty_moov -f mp4
and
ffmpeg -i RTSPCAMERAPRODUCEH264 -c:v copy -an -movflags +frag_keyframe+empty_moov -r 1 -f mp4
do not seem to work.
A temporally coded video stream (like one with H264 codec) cannot arbitrarily drop intermediate packets, so this is not possible. Only whole or trailing part of GOPs may be dropped.
Related
Before posting I have searched and found similar questions on stackoverflow (I list some below) - none have helped me towards a solution, hence this post. The duration that each image is shown within the movie file differs from many posts that I have seen thus far.
A camera captures 1 image every 30 seconds. I need stream them, preferably via HLS, thus I wrap 2 images in an MP4. I then convert MP4 to mpegts. Each MP4 and TS file play fine individually (each contain two images, each image transitions after 30seconds, each movie file is 1minute long).
When I reference the two TS files in an M3U8 playlist, only the first TS file gets played. Can anyone advise why it stops and how I can get it to play all the TS files that I expect to create, not just the first TS file? Besides my ffmpeg commands, I also include my VLC log file (though I expect to stream to Firefox/Chrome clients). I am using ffmpeg 4.2.2-static installed on an AWS EC2 with AMI2 Linux.
I have four jpgs named image11.jpg, image12.jpg, image21.jpg, image22.jpg - The images look near identical as only the timestamp in top left changes.
The following command creates 1.mp4, using image11.jpg and image12.jpg, each image displayed for 30 seconds, total duration of the mp4 is 1 minute. It plays like expected.
ffmpeg -y -framerate 1/30 -f image2 -i image1%1d.jpg -c:v libx264 -vf "fps=1,format=yuvj420p" 1.mp4
I then convert 1.mp4 to an mpegts file, creating 1.ts. It plays like expected.
ffmpeg -y -i 1.mp4 -c:v libx264 -vbsf h264_mp4toannexb -flags -global_header -f mpegts 1.ts
I repeat the above steps except specific to image21.jpg and image22.jpg, creating 2.mp4 and 2.ts
ffmpeg -y -framerate 1/30 -f image2 -i image1%1d.jpg -c:v libx264 -vf "fps=1,format=yuvj420p" 2.mp4
ffmpeg -y -i 1.mp4 -c:v libx264 -vbsf h264_mp4toannexb -flags -global_header -f mpegts 2.ts
Thus now I have 1.mp4, 1.ts, 2.mp4, 2.ts and all four play individually just fine.
Using ffprobe I can confirm their duration is 60seconds, for example:
ffprobe -i 1.ts -v quiet -show_entries format=duration -hide_banner -print_format json
My m3u8 playlist follows:
#EXTM3U
#EXT-X-VERSION:4
#EXT-X-PLAYLIST-TYPE:VOD
#EXT-X-MEDIA-SEQUENCE:1
#EXT-X-TARGETDURATION:60.000
#EXTINF:60.0000,
1.ts
#EXTINF:60.000,
2.ts
#EXT-X-ENDLIST
Can anyone advise where I am going wrong?
VLC Error Log (though I expect to play via web browser)
I have researched the process using these (and other pages) as a guide:
How to create a video from images with ffmpeg
convert from jpg to mp4 by ffmpeg
ffmpeg examples page
FFMPEG An Intermediate Guide/image sequence
How to use FFmpeg to convert images to video
Take a look at the start_pts/start_time in the ffprobe -show_streams output, my guess is that they all start at zero/near-zero which will cause playback to fail after your first segment.
You can still produce them independently but you will want to use something like -output_ts_offset to correctly set the timestamps for subsequent segments.
The following solution works well for me. I have tested it uninterrupted for more than two hours and believe it ticks all my boxes. (Edited because I forgot the all important -re tag)
ffmpeg will loop continuously, reading test.jpg and stream it to my RTMP server. When my camera posts an image every 30seconds, I copy the new image on top of the existing test.jpg which in effect changes what is streamed out.
Note the command below is all one line, I have put new lines in to assist reading and The order of the parameters are important - the loop and fflags genpts for example must appear before the -i parameter
ffmpeg
-re
-loop 1
-fflags +genpts
-framerate 1/30
-i test.jpg
-c:v libx264
-vf fps=25
-pix_fmt yuvj420p
-crf 30
-f fifo -attempt_recovery 1 -recovery_wait_time 1
-f flv rtmp://localhost:5555/video/test
Some arguments explained:
-re implies play in real time
loop 1 (1 turns the loop on, 0 off)
-fflags +genpts is something I only half understand. PTS I believe is the start/end time of the segment and without this flag, the PTS is reset to zero with every new image. Using this arguement means I avoid EXT-X-DISCONTINUITY when a new image is served.
-framerate 1/30 means one frame for 30seconds
-i test.jpg is my image 'placeholder'. As new images are received via a separate script, it overwrites this image. When combined with loop it means the ffmpeg output will reference the new image.
-c:v libx264 is for H264 video output formating
-vf fps=25 Removing this, or using a different value resulted in my output stream not being 30seconds.
-pix_fmt yuvj420p (sometimes I have seen yuv420p referenced but this did not work on my environment). I believe there are different jpg colour palettes and this switch ensures I can process a wider choice.
-crf 30 implies highest quality image, lowest compression (important for my client)
-f fifo -attempt_recovery 1 -recovery_wait_time 1 -f flv rtmp://localhost:5555/video/test is part of the magic to go with loop. I believe it keeps the connection open with my stream server, reduces the risk of DISCONTINUITY in the play list.
I hope this helps someone going forward.
The following links helped nudge me forward and I share as it might help others to improve upon my solution
Creating a video from a single image for a specific duration in ffmpeg
How can I loop one frame with ffmpeg? All the other frames should point to the first with no changes, maybe like a recusion
Display images on video at specific framerate with loop using FFmpeg
Loop image ffmpeg HLS
https://trac.ffmpeg.org/wiki/Slideshow
https://superuser.com/questions/1699893/generate-ts-stream-from-image-file
https://ffmpeg.org/ffmpeg-formats.html#Examples-3
https://trac.ffmpeg.org/wiki/StreamingGuide
According to the ffmpeg documentation
-vsync parameter
Video sync method. For compatibility reasons old values can be specified as numbers. Newly added values will have to be
specified as strings always.
drop
As passthrough but destroys all timestamps, making the muxer
generate fresh timestamps based on frame-rate.
It appears that the mpegts mux does not regenerate the timestamps correctly (PTS/DTS); however, piping the output after vsync drop to a second process as raw h264 does force mpegts to regenerate the PTS.
Generate test stream
ffmpeg -f lavfi -i testsrc=duration=20:size=1280x720:rate=50 -pix_fmt yuv420p -c:v libx264 -b:v 4000000 -x264-params ref=1:bframes=0:vbv-maxrate=4500:vbv-bufsize=4000:nal-hrd=cbr:aud=1:bframes=0:intra-refresh=1:keyint=30:min-keyint=30:scenecut=0 -f mpegts -muxrate 5985920 -pcr_period 20 video.ts -y
Generate output ts that has correctly spaced PTS values
ffmpeg -i video.ts -vsync drop -c:v copy -bsf:v h264_mp4toannexb -f h264 - | ffmpeg -fflags +igndts -fflags +nofillin -fflags +genpts -r 50 -i - -c:v copy -f mpegts -muxrate 5985920 video_all_pts_ok.ts -y
Generate output ts where all PTS are zero
ffmpeg -i video.ts -vsync drop -c:v copy -bsf:v h264_mp4toannexb -f mpegts - | ffmpeg -fflags +igndts -fflags +nofillin -fflags +genpts -r 50 -i - -c:v copy -f mpegts -muxrate 5985920 video_all_pts_zero.ts -y
It appears that vsync drop does destroy them but the mpegts doesn't regenerate them? Any ideas on what needs adding to get it to work as a single ffmpeg command?
Tested on both Linux and Windows with the same result
Try recoding the video just using -vsync 1, without -fflags +genpts. I found some good information here. This guy talking about streaming video. So highest quality isn't his objective. But there is useful info.
https://videoblerg.wordpress.com/2017/11/10/ffmpeg-and-how-to-use-it-wrong/
Section one – Constant frame rate
"-r is used to specify the output frame rate. This must be the same as the input frame rate to eliminate judder. This is used in conjunction with the -vsync parameter using the 1 option which will retime the PTS/DTS timestamps accordingly"
Section six – Audio [Has some good advice too]
"-af "aresample=async=1:min_hard_comp=0.100000:first_pts=0" helps to keep your audio lined up with the beginning of your video. It is common for a container to have the beginning of the video and the beginning of the audio start at different points. By using this your container should have little to no audio drift or offset as it will pad the audio with silence or trim audio with negative PTS timestamps if the audio does not actually start at the beginning of the video."
I haven't tried this yet, no videos with sync problems at the moment.
I need to capture an audio/video rtsp stream uncompressed in a file from ipcamera. Audio (pcm_alaw) and video (h264) must be synchronized. It is necessary that the file does not get corrupted if the camera loses the connection for a few moments (mp4).
At the moment I use the command below, but the ts codec does not support pcm_alaw and therefore the audio is not heard:
ffmpeg -stimeout 2000000 -rtsp_transport tcp -i rtsp://admin:1234#192.168.5.22/h264 -c:v copy -c:a copy -f mpegts -y main.ts
I use the mpegts codec because I need to check the duration of the capture in real time with the command:
ffprobe -i /home/pi/NAS/main.mov -show_entries format=duration -v quiet -of csv="p=0"
If i use mkv or avi its output would be:
N/A
The verification of the duration is important because I capture files of about 3 hours and at my choice I perform some data while the capture is in progress. I prefer not to compress the audio because I have often noticed some asynchrony with respect to the video when cutting.
Thank you.
Instead of -c:a copy you can use -c:a aac or -c:a mp3 to convert the audio stream before you save it.
MPEG-TS h264 is only compatible with mp3 or aac (source).
I'm currently using the streaming plugin as follows
Fancy artchitecture here
OBS--------RTMP--------->NGINX-Server------FFMPEG(input RTMP output RTP)--------->JANUS---------webrtc-------->Client
When using the ffmpeg command (bellow), on the Janus streaming interface, we only see the bitrate that corresponds to that of the ffmpeg output in the console but we don't see any video.
ffmpeg -i rtmp://localhost/live/test -an -c:v copy -flags global_header -bsf dump_extra -f rtp rtp://localhost:8004
(using "-c:v copy" so that no encoding is used and hence reducing the
latency)
The video shows fine if I use "-c:v libx264", the only issue is that it is CPU intensive and adds latency.
Previously I had tried using RTSP as input for FFMPEG and in this case the video show fine with almost no latency even though I use "-c:v copy".
So I don't realy get why for RTSP the copy works fine, but for RTMP I have to use the libx264 codec. If anyone has an idea about this I am all ears :)
I had similar issue and my problem was that the stream / video that I used has large GOP size.
For WebRTC, latency is sub-second, so the input source should have short interval I frames. Better to remove B frames since they referring backward and forward as well.
Here are commands that you could use for small GOP size (4) and remove B frames.
Using RTMP streaming src:
ffmpeg rtmp://<your_src> -c:v libx264 -g 4 -bf 0 -f rtp -an rtp://<your_dst>
Using a mp4 file:
ffmpeg -re -i test.mp4 -c:v libx264 -g 4 -bf 0 -f rtp -an rtp://<your_dst>
-c:v copy does not reduce latency. It merely tells ffmpeg not to transcode.
i want convert video from any format to mp4. so i am using command:
ffmpeg -i ttt.mp4 -vcodec copy -acodec copy test.mp4
this is working perftectly but now i also add scale in this -s 320:240.
There also many other command for convert LIKE :
ffmpeg -i inputfile.avi -s 320x240 outputfile.avi
but after convert by this command video not play in html5 player
BUT this is not working so tell me in my command how i add scale;
So please provide me solution for this .
Thanks in advance.
You have several problems:
In your command, you have -vcodec copy you cannot scale video without reencoding.
In the command you randomly found on the Internet, they are using AVI, which is not HTML5-compatible.
What you should do is:
ffmpeg -i INPUT -s 320x240 -acodec copy OUT.mp4
Adding to Timothy_G:
Video copy will ignore the video filter chain of ffmpeg, so no scaling is available (man ffmpeg is a great source of information that you will not find on Google). Notice that once you start decoding-filtering-encoding (i.e., no copy) the process will be much slower (x100 time slower or even more). The libx264 is recommended if you want compatibility with all browsers.
$ ffmpeg -i INPUT -s 320x240 -threads 4 -c:a copy -c:v libx264 OUT.mp4
vp9 will provide nearly 50% extra bandwidth saving, but only for supported browsers (Firefox/Chrome), and the encoding will much slower compared to libx264 (that itself is much slower that v:c copy):
$ ffmpeg -i INPUT -s 320x240 -c:a copy -c:v vp9 OUT.webm
Notice that there is a set of formats (containers) accepted by browsers (most admit mp4, some also webm, ...) and for each format there is a set of audio/video codecs accepted. For example you can use mp3 or aac with an mp4 file (container), but not with webm files.
http://en.wikipedia.org/wiki/HTML5_video#Supported_video_formats