I'm using FFMPEG on windows with direct show.
I'm streaming RTMP (command below) and i need very low latency.
Once run I get the following errors: [dshow # 024ce800] real-time buffer 204% full! frame dropped!
ffmpeg -threads 6 -f dshow -i video=UScreenCapture -s 1920x1080 -an -vco
dec libx264 -x264opts keyint=25:min-keyint=20 -b:v 1024k -preset ultrafast -tune zerolatency -crf 22 -r 10 -pix_fmt yuv420p -f flv rtmp://server...
Do you have an idea how to handle this kind of error?
Thanks
Ronen
That message means "dshow got an incoming packet, but you haven't finished sending the previous packet yet" so in reality, this should be contributing to as low a latency possible. If your goal is to avoid dropping packets then increase rtbufsize. With rtmp hopefully there will be some improvements soon so it has better throughput.
Related
I tried various ways to make a live streaming script with a bit rate according to YouTube's recommendation of 4500 Kbps bit rate.
The code:
ffmpeg -re -stream_loop -1 -i live1.mp4 -c copy -preset veryfast -b:v 7000k -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 -c:a aac -b:a 160k -ac 2 -ar 44100 -f flv -flvflags no_duration_filesize rtmp://a.rtmp.youtube.com/live2/(streamkey)
and in my current code, there is an error when live: Use a keyframe frequency of four seconds or less. Currently, keyframes are sent infrequently which can cause buffering. The current keyframe frequency is 5.0 seconds. Please note that errors in the transfer process can cause the size of the GOP (group of images) to be incorrect.
How to fix my code?
I've tried several ways, but the bit rate is still high, and the error is in YouTube Studio
I create a webrtc peer connection with my server(only stun)
Using pion webrtc for the server
I write the received RTP packets as VP8 and opus streams, as described here, to two pipes (the writers; created with os.Pipe() in golang)
The read ends of these two pipes are received by ffmpeg as inputs (via exec.Command.ExtraFiles) for transcoding using libx264 and aac into a single stream. The command:
ffmpeg -re -i pipe:3 -re -r pipe:4 -c:a aac -af aresample=48000 -c:v libx264 -x264-params keyint=48:min-keyint=24 -profile:v main -preset ultrafast -tune zerolatency -crf 20 -fflags genpts -avoid_negative_ts make_zero -vsync vfr -map 0:0,0:0 -map 1:0,0:0 -f matroska -strict -2 pipe:5
The above command outputs to a pipe(:5) the read end of which is being taken as input by the following:
ffmpeg -hide_banner -y -re -i pipe:3 -sn -vf scale=-1:'min(ih,360)' -c:v libx264 -pix_fmt yuv420p -ca aac -b:a 128k -b:v 1400k -maxrate 1498k -bufsize 2100k -hls_time 1 -hls_playlist_type event -hls_base_url /workdir/streamID/360p -hls_segment_filename /workdir/streamID/360p/360_%%03d.ts -f hls /workdir/streamID/360p.m3u8
This works fine as long as there are no movements of my webcam. The moment that happens the video speed suddenly increases for a split second and audio delay gets introduced. This delay keeps increasing each time I shake my webcam.
The first command in point 4 above - if written to a file separately will be absolutely fine, in terms of a/v sync, even with vigorous camera shaking. The weird audio delay is only when transcoding for hls output irrespective of whether I'm actually viewing it live or playing it back later.
This is my first time working with ffmpeg/hls/webrtc - would be really helpful if I could be pointed in the correct direction at least to be able to debug this or even know why this happens. Any and all help is greatly appreciated
I am trying to make a reliable stream from my Icecast/Shoutcast servers to Youtube live. The command that I use is:
ffmpeg -v verbose -framerate 30 -loop 1 -i /var/image.jpg -re -i http://127.0.0.1:4700/radio -c:v libx264 -preset ultrafast -b:v 2250k -maxrate 6000k -bufsize 6000k -c:a copy -ab 128k -s 1920x1080 -framerate 30 -g 60 -keyint_min 60 -f flv rtmp://a.rtmp.youtube.com/live2/xxx
As you can see I am using recommended bitrate for Youtube, insert keyframes every 2 seconds and streaming at 30 frames per second.
The stream is working but after running for some time two thing are happening:
FFMPEG speed falls from 1x to something like 0.998x
Youtube starts complaining that video stream speed is slow, markes the quality as bad and sometimes video starts buffering.
Why is this happening? CPU load is normal, connectivity is ok (the stream is running on a 1Gg/s dedicated server).
Since in my example above I am streaming a single image as a logo of the stream I also tried to generate a short 30 seconds video with that image and broadcast that video instead of an image, but that did not help as well.
The command I used for conversion:
ffmpeg -framerate 30 -loop 1 -i /var/image.jpg -c:v libx264 -preset ultrafast -tune stillimage -b:v 2250k -minrate 2250k -maxrate 6000k -bufsize 6000k -framerate 30 -g 60 -keyint_min 60 -t 30 out4.mp4
And broadcast with
ffmpeg -stream_loop -1 -i out4.mp4 -re -i http://127.0.0.1:4700/radio -c:v copy -c:a copy -framerate 30 -g 60 -keyint_min 60 -f flv rtmp://a.rtmp.youtube.com/live2/xxx
ffmpeg version is 4.1.1
Are you sure that your original stream is really keeping up with the wall-clock?
Depending on how it's encoded there are possibilities that it gets heavily skewed. This ultimately leads to buffer under (or overruns if it's too fast) and the player complaining/skipping.
Can you try and dump several hours worth of stream to a file and then stream that with FFmpeg? If that works, then it's a strong indication that your original stream timing (sample rate) is off.
Getting the sample rate right is why professional/expensive sound cards use high precision Quartz-Crystal controlled oscillators. Purely virtual processing (e.g. files get encoded into a stream) can easily get skewed, especially inside virtual machines. Also, cheap USB sound cards are often among the worst offenders in terms of frequency accuracy and stability.
FFmpeg might have an option to deal with too slow input. Keywords could be 'padding' or 'missing samples'.
Youtube's error saying "...buffer....." is not a buffer issue on your PC, but simply data you are sending to youtube is too small.
1)note that [-preset ultrafast] and [-preset fast] does not make big difference.
2) change your ffmpeg comannd for broadcast one. like, [-b:v 2250k] to [-b:v 15000k],and set fps to 12→[-r 12] option.
I's gonna be.
ffmpeg -stream_loop -1 -i out4.mp4 -re -i http://127.0.0.1:4700/radio -preset fast -r 12 -framerate 30 -g 60 -video_track_timescale 1000 -b:v 15000k -f flv rtmp://a.rtmp.youtube.com/live2/xxx
I hope this will be good for you !!(^v^)Y
I'm using FFMPEG(h264) and I want to reduce latency as much as possible. Now it's about 700 ms and I can't really make it lower. I tried almost all, so maybe anyone has idea how to help me?
ffmpeg -f dshow -i video="screen-capture-recorder" -pix_fmt yuv420p -probesize 32 -r 100 -an -vcodec libx264 -crf 40 -preset ultrafast -tune zerolatency -threads 8 -thread_type slice -f mpegts udp://192.168.88.228:1234
The weird thing is I got this latency even on 127.0.0.1....
(on the other side I use just ffplay udp:// .......)
I would try to set -threads to 1 to disable multi-threaded decoding. Multi-threaded decoding introduces delay by adding a lag of 1 frame for each thread.
Works with a GoPro Hero 8 Black and Linux
ffmpeg -threads 1 -i 'udp://#0.0.0.0:8554?overrun_nonfatal=1&fifo_size=50000000' -f:v mpegts -fflags nobuffer -vf format=yuv420p -f v4l2 /dev/video0
like above question, I want find out what ffmpeg command can help me reduce cpu usage when running 50 IP camera (running same 50 command).
My ffmpeg command:
ffmpeg -f mjpeg -y -use_wallclock_as_timestamps 1 -i 'http://x.x.x.x:8090/test1?.mjpg' -r 3 -reconnect 1 -loglevel 16 -c:v mjpeg -an -qscale 10 -copyts '1.mp4'
50 command like that take my computer (4 core) 200% CPU
I want this computer can run for 150 camera, any advise?
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using -c:v copy can make it faster but fize size is terrible
I try slow down frame rate by 3 with -r 3 or -framerate 3 to decrease file size but not succesful (because vcodec copy can't do that).
Have any option to force input frame rate by 3?
(sorry for my bad English)
by setting -c:v mjpeg you are decoding and re-encoding the stream. set -c:v copy to copy the data without re-encoding it.
ffmpeg -re -i 'rtsp://user:password#10.10.10.30/rtsp_tunnel' -pix_fmt yuv420p -c:v libx264 -preset ultrafast -profile baseline -crf 18 -f h264 udp://0.0.0.0:3001