I'm using ffmpeg to restream a live feed. Unfortunately occasionally the input resolution changes but ffmpeg continues running. The nginx rtmp server I'm using doesn't cope well with this, and continues the stream with audio, but the video is mostly black or green with some artifacts.
Ideally what I want to happen is for ffmpeg to stop on an input resolution change, as I have a script that detects ffmpeg stopping and will restart it again.
I'm using -c:v copy in my ffmpeg command as unfortunately my machine is not powerful enough to re-encode the live video on the fly to a constant resolution (not without a significant quality reduction at least)
ffmpeg -i "http://mpegts-live-stream" -c:v copy -c:a aac -ac 2 -f flv "rtmp://nginxserver/live/streamname"
So, ffmpeg can't detect input parameter change during streamcopy but we can work around that by adding a minimal decoding load to the process.
ffmpeg -xerror -skip_frame:v nokey -flags:v +drop_changed -i "http://mpegts-live-stream" -c:v copy -c:a aac -ac 2 -f flv "rtmp://nginxserver/live/streamname" -an -f null -
-xerror : exits upon error
-skip_frame:v nokey : only decodes keyframes of video stream
-flags:v +drop_changed : drops frame and signals error when parameters (like resolution) change
-an -f null - : maps a video output stream which needs a decoded stream, needed else no decoding will occur.
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
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.
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've been testing different parameters to capture my desktop video and audio (desktop audio, not mic) and I find that no matter what settings I have, the resulting webm file's framerate is around 5fps and is horribly inconsistent. It starts at around 20fps and slowly drops over time until about 4-5fps. I'm not really sure what I'm doing wrong, but here is the basic command I'm using:
ffmpeg -y -video_size 1920x1080 -f gdigrab -framerate 60 -i desktop -c:v libvpx-vp9 -acodec libvorbis -c:a libopus -b:v 2M -threads 4 output.webm
I've tried anywhere between 30-60 fps and tested different bitrates but nothing seems to affect the output framerate.
Also, I know that acodec and c:a are for audio but I'm not sure how to specify the audio device to use.
So my issues are horrible framerate for webm and how to include desktop audio in the recording.
You can use arecord and pipe it through stdout and ffmpeg can read it from stdin.
aplay piping to arecord using a file instead of stdin and stdout
Replacing the aplay command with your ffmpeg. Dont forget to add '-i -' in ffmpeg.
A doubt: why are you defining audio encoder two times?
It's impossible to say why the video frame rate is low from the question. It can be an issue with encoder. Or issue in reading input. Remove the video encoding option. See if the issue persists. If it's working fine, try some other encoders.
Use -c:v libx264 instead of -c:v libvpx-vp9. libvpx-vp9's realtime encoding quality is really bad, even regular libvpx (i.e. VP8) is much better. If you insist on using libvpx, use options like -deadline realtime and -cpu-used -4
everyone.
I'm trying to use FFmpeg to record video and 3 audio sources and use it to generate 3 different video files - each file should contain the same video stream but the different audio stream. The problem is that I got audio sync issues. The first audio stream is synced perfectly, but the second one has 1 sec lag, and the third one has like 2 sec lag.
I've made a few tests so far and it seems that root cause of the issue is initialization time of video/audio devices. So, one device is already recording something but the second is still being opened and so on. I've tried to change input devices order and after that audio streams still have the same issue BUT if before 2nd and 3rd audio streams were some time ahead of video, after reordering they became to lag after the audio (audio for the same event appears with some delay). So this test confirms my version about device initialization times.
But the question still, why the first audio stream is synchronized properly, while other 2 are not. and also, how could I overcome this issues? Any workarounds and ideas are highly appreciated.
Here is FFmpeg command I'm using and it's output.
ffmpeg.exe -f dshow -video_size 1920x1080 -i video="Logitech HD Webcam
C615" -f dshow -i audio="Microphone (HD Webcam C615)" -f dshow -i
audio="Microphone Array (Realtek High Definition Audio)"
-filter_complex "[1:a]volume=1[a1];[2:a]volume=1[a2]" -vf scale=h=1080:force_original_aspect_ratio=decrease -vcodec libx264
-pix_fmt yuv420p -crf 23 -preset ultrafast -acodec aac -vbr 5 -threads 0 -map v:0 -map [a1] -map [a2] -f tee
"[select=\'v,a:0\']C:/Users/vshevchu/Desktop/123/111/111_jjj1.avi|
[select=\'v,a:1\']C:/Users/vshevchu/Desktop/123/111/111_jjj2.avi"
OUTPUT
PS. Actually, the issue is exactly the same when I'm not using "tee" muxer but writing all the audio streams to one container. So, "tee" isn't a suspect.