I'm trying to stream a video from a rtsp server to a rtmp one using FFmpeg.
Tried multiple arguments for my command :
ffmpeg.exe -re -i "rtsp://10.65.28.251:11442/video/live" -pix_fmt yuv420p -codec:v libx264 -tune animation -preset fast -crf 23 -maxrate 4M -bufsize 8M -f flv "rtmp://10.65.58.21:1935/rec/XB"
ffmpeg.exe -re -i "rtsp://10.65.28.251:11442/video/live" -preset ultrafast -vcodec libx264 -tune zerolatency -b 900k -f flv "rtmp://10.65.52.131:1935/rec/XB
I'm loosing a lot of packages as seen in the picture. I'm pretty new to FFmpeg so I'm pretty sure I'm messing up the parameters somehow.
My goal is to get a video on rtmp with min 30fps and as least lost packages as possible. If needed a downsize of the video quality would be fine.
Any idea what I'm doing wrong?
Thanks!
As kesh pointed above removing -re made a big difference. I ended up with this command which holds pretty good quality at 30fps.
ffmpeg.exe -i "rtsp://serversource:11442" -filter:v fps=fps=30 -crf 40 -preset ultrafast -vcodec libx264 -f flv "rtmp://servertarget:1935"
I am trying to get ffmpeg to work as expected however I am having all kinds of trouble getting it to work.
I need to output a webm and h264 for web play. However, the command I am using, while it used to work a few years ago, does not work at all now.
Both my webm and h264 do not have audio, and neither will play in any browser.
My command for webm is:
ffmpeg -y -i "$KMVAR_File" -c:v libvpx -crf 24 -b:v 1000k -vf scale=720:-2 -c:a libvorbis "$KMVAR_webmPath"
and my command for mp4 is:
ffmpeg -y -i "$KMVAR_File" -c:v libx264 -pix_fmt yuv420p -profile:v baseline -level 3.0 -crf 32 -b:v 1M -minrate 1M -maxrate 1M -bufsize 2M -vf scale=720:-2 -c:a aac -strict experimental -movflags +faststart "$KMVAR_mp4Path"
When playing with multiple audio, downmixing or extracting, there's no "one size fit all" solution with ffmpeg.
Look at https://trac.ffmpeg.org/wiki/AudioChannelManipulation as it provides multiple possible solution to your problem.
(I usually go with the pan filter : not the easiest to use, but more powerful than the map_channel approach)
I'm doing screen recordings of gameplay (Dota2) using my NVIDIA graphics card GeForce experience hardware recording (NVEC Encoder). This creates a variable frame rate output video. My NVIDIA settings are 60 fps 15000 kbps. I have paid a guy to make a program that generates scripts that given start/stop timepoints can extract clips from the video and merge them with crossfade. See example code below. The script works for many input recordings but fails often: The audio and video are desynchronized (usually audio delay) in many of the clips, ca 0.5 seconds. I think it fails more when frame rate dropped more during recording. He does not know how to fix the problem, and I wonder if anyone could point out if anything could be fixed in the script (example below)?
Processing speed is quite important (now making a 10 min 'highlight' video takes ca 7-10 min). Solutions increasing that amount very much more is not of too big interest, unfortunately. His approach has been to work separately with audio and video and merge in the end. He already has a program to make ffmpeg code for working with different scenarios (also adding overlays, adding music, intro/outro) so it would be preferable with some easy fixes to his code and not dramatic redesigning of the logic. But if nothing else can fix the problem, a redesign in logic is ok. Using other tools than ffmpeg is also ok, but should be automatable (scripts/cli) and not increase processing times too much.
Running the program "mediainfo" on the input video shows that framerate dropped quite low for this input video:
Frame rate mode: Variable
Frame rate : 60.000 FPS
Minimum frame rate: 3.059 FPS
Maximum frame rate: 63.739 FPS
Full report here: https://pastebin.com/TX061Wih
The input video can be downloaded from dropbox here (6 GB):
https://www.dropbox.com/s/ftwdgapazbi62pr/fullgame.mp4?dl=0
Here the example of a script when asked to extract two clips from input video at 9:57 (41 sec length) and 15:45 (28 sec length) and crossfade merge them with a 0.5 crossfade time. There might be some code-remnants from options that are not used in this example (overlays, music, intro/outro). Using the input video above, this creates audio/video desync.
6 commands excecuted in sequence:
ffmpeg.exe -loglevel warning -ss 00:09:57 -i fullgame.mp4 -t 00:00:41 -filter_complex "[0:a]afade=t=out:st=40.5:d=0.5[a1]" -map "[a1]" -y out_temp_00.mp4.wav
ffmpeg.exe -loglevel warning -i fullgame.mp4 -ss 00:09:57 -t 00:00:41 -an -vcodec copy -f mpegts -avoid_negative_ts make_zero -y out_temp_00.mp4.ts
ffmpeg.exe -loglevel warning -ss 00:15:45 -i fullgame.mp4 -t 00:00:28 -filter_complex "[0:a]afade=t=in:st=0:d=0.5[a1]" -map "[a1]" -y out_temp_01.mp4.wav
ffmpeg.exe -loglevel warning -i fullgame.mp4 -ss 00:15:45 -t 00:00:28 -an -vcodec copy -f mpegts -avoid_negative_ts make_zero -y out_temp_01.mp4.ts
ffmpeg.exe -loglevel warning -i out_temp_00.mp4.wav -i out_temp_01.mp4.wav -y -filter_complex "[0:a]adelay=0|0[a0];[1:a]adelay=40500|40500[a1];[a0][a1]amix=inputs=2:dropout_transition=68.5,atrim=duration=68.5[outa0];[outa0]loudnorm[outa]" -map "[outa]" -ar 48000 -acodec aac -strict -2 fullgame_Output.mp4.aac
ffmpeg.exe -loglevel warning -i out_temp_00.mp4.ts -i out_temp_01.mp4.ts -y -i fullgame_Output.mp4.aac -filter_complex "[0:v]trim=start=0.5,setpts=PTS-STARTPTS[0c];[1:v]trim=start=0.5,setpts=PTS-STARTPTS[1c];[0:v]trim=40.5:41,setpts=PTS-STARTPTS[fo];[1:v]trim=0:0.5[fi];[fi]format=pix_fmts=yuva420p,fade=t=in:st=0:d=0.5:alpha=1[z];[fo]format=pix_fmts=yuva420p,fade=t=out:st=0:d=0.5:alpha=1[x];[z]fifo[w];[x]fifo[q];[q][w]overlay[r];[0c][r][1c]concat=n=3[outv]" -map "[outv]" -map 2:a -shortest -acodec copy -vcodec libx264 -preset ultrafast -b 15000k -aspect 1920:1080 fullgame_Output.mp4
P.S.
I already asked for help at an ffmpeg chat room. One guy said he knew what the problem was, but didnt know how to fix it(?):
[00:10] <kepstin> oh, wait, you're using -vcodec copy
[00:10] <kepstin> that explains everything.
[00:10] <kepstin> when you're using -vcodec copy, the start time (set with -ss) is rounded to the nearest keyframe
[00:10] <kepstin> it's not exact
[00:11] <kepstin> depending on the keyframe interval, this will result in possibly quite large shifts
[00:11] <kepstin> (also, your commands are applying audio filters on commands with -an, which is confusing/contradictory)
[00:12] <birdboy88> so the problem is that the audio temporary clips are not being extracted from the same excat timepoints?
[00:13] <kepstin> birdboy88: yeah, your audio is being re-encoded to wav so it's being cut sample-accurate, but the video's not being precisely cut.
[00:16] <birdboy88> kepstin: so I need to use slow seek (?) to extract video accurately? Or somehow extract audio only where there are video keyframes?
[00:17] <kepstin> birdboy88: i don't know how to extract audio starting at video keyframes with ffmpeg cli. You're already doing slow seek, which doesn't help (you should move the -ss option to before the -i option to speed it up)
[00:17] <kepstin> if you want accurate video cutting when saving to a file, you have to re-encode the video
[00:18] <kepstin> (doing this in a single ffmpeg command means you don't have to save to a file, so you can avoid the issue)
[00:18] * kepstin is off for a bit now
EDIT:
Everything is done with the latest ffmpeg version.
I was unable to get Gyan's code to work. It always loses some audio (audio is either 40.5 or 27.5, so only one audio is used). This is the only one working for me (changes were adelay=40500|40500 and amix=inputs=2[a0];[a0]loudnorm):
ffmpeg -i fullgame.mp4 -filter_complex "[0]split=2[vpre][vpost];
[0]asplit=2[apre][apost];
[vpre]trim=start='00:09:57':duration='00:00:41',setpts=PTS-STARTPTS[vpre-t];
[apre]atrim=start='00:09:57':duration='00:00:41',asetpts=PTS-STARTPTS,afade=t=out:st=40.5:d=0.5[apre-t];
[vpost]trim=start='00:15:45':duration='00:00:28',setpts=PTS-STARTPTS,format=yuva420p,fade=t=in:st=0:d=0.5:alpha=1,setpts=PTS+40.5/TB[vpost-t];
[apost]atrim=start='00:15:45':duration='00:00:28',asetpts=PTS-STARTPTS,afade=t=in:st=0:d=0.5,adelay=40500|40500[apost-t];
[vpre-t][vpost-t]overlay[v];
[apre-t][apost-t]amix=inputs=2[a0];[a0]loudnorm[a]" -map "[v]" -map "[a]" -y -c:v libx264 -preset ultrafast -b:v 15000k -aspect 1920:1080 -c:a aac fullgame_Output.mp4
Then I tried using a similar setup but with 3 clips, but on one machine I got error: "Error while filtering: Cannot allocate memory". And my 16 GB memory machine the processing speed is 0.02x! Any way to avoid this? This is the code I tried:
ffmpeg -i fullgame.mp4 -filter_complex "[0]split=3[vpre][vpost][v3];
[0]asplit=3[apre][apost][a3];
[vpre]trim=start=357:duration=41,setpts=PTS-STARTPTS[vpre-t];
[apre]atrim=start=357:duration=41,asetpts=PTS-STARTPTS,afade=t=out:st=40.5:d=0.5[apre-t];
[vpost]trim=start=795:duration=28,setpts=PTS-STARTPTS,format=yuva420p,fade=t=in:st=0:d=0.5:alpha=1,fade=t=out:st=40.5:d=0.5:alpha=1,setpts=PTS+40.5/TB[vpost-t];
[apost]atrim=start=795:duration=28,asetpts=PTS-STARTPTS,afade=t=in:st=0:d=0.5,afade=t=out:st=27.5:d=0.5,adelay=40500|40500[apost-t];
[v3]trim=start=95:duration=30,setpts=PTS-STARTPTS,format=yuva420p,fade=t=in:st=0:d=0.5,setpts=PTS+41+28-0.5/TB[v3-t];
[a3]atrim=start=95:duration=30,asetpts=PTS-STARTPTS,afade=t=in:st=0:d=0.5,adelay=68500|68500[a3-t];
[vpre-t][vpost-t]overlay[v1];
[v1][v3-t]overlay[v];
[apre-t][apost-t][a3-t]amix=inputs=3[a0];
[a0]loudnorm[a]" -map "[v]" -map "[a]" -y -c:v libx264 -preset ultrafast -b:v 15000k -aspect 1920:1080 -c:a aac fullgame_Output.mp4
Just do it in one command.
Besides the keyframe seek issue, which is true, your present sequence has an error in the last command. You have [0:v]trim=start=0.5...[0c] which trims out the first 0.5 seconds and will cause a desync of its own. Since this is the first clip, it should be [0:v]trim=0:40.5.
The full single command should be
ffmpeg -i fullgame.mp4 -filter_complex
"[0]split=2[vpre][vpost];[0]asplit=2[apre][apost];
[vpre]trim=start='00:09:57':duration='00:00:41',setpts=PTS-STARTPTS[vpre-t];
[apre]atrim=start='00:09:57':duration='00:00:41',asetpts=PTS-STARTPTS,afade=t=out:st=40.5:d=0.5[apre-t];
[vpost]trim=start='00:15:45':duration='00:00:28',setpts=PTS-STARTPTS,format=yuva420p,fade=t=in:st=0:d=0.5:alpha=1,setpts=PTS+40.5/TB[vpost-t];
[apost]atrim=start='00:15:45':duration='00:00:28',asetpts=PTS-STARTPTS,afade=t=in:st=0:d=0.5[apost-t];
[vpre-t][vpost-t]overlay[v];
[apre-t][apost-t]acrossfade=d=0.5,loudnorm,aresample=48000[a]"
-map "[v]" -map "[a]" -c:v libx264 -preset ultrafast -b:v 15000k -aspect 1920:1080 -c:a aac fullgame_Output.mp4
Your original sequence had -strict -2 for audio AAC encoding. That hasn't been needed since Dec 2015. You have a very old version of ffmpeg if your ffmpeg throws an error without it. Upgrade first.
I did not test the above with your file, as it will take too long to filter 16 min of Full HD 60 fps video, but I tested the below faster command and it works fine with the latest git build of ffmpeg:
ffmpeg -ss 00:09:57 -t 00:00:41 -i fullgame.mp4 -ss 00:15:45 -t 00:00:28 -i fullgame.mp4 -filter_complex
"[0]afade=t=out:st=40.5:d=0.5[apre-t];
[1]format=yuva420p,fade=t=in:st=0:d=0.5:alpha=1,setpts=PTS+40.5/TB[vpost-t];
[1]afade=t=in:st=0:d=0.5[apost-t];
[0][vpost-t]overlay[v];
[apre-t][apost-t]acrossfade=d=0.5,loudnorm,aresample=48000:ocl=stereo[a]"
-map "[v]" -map "[a]" -c:v libx264 -preset ultrafast -b:v 15000k -aspect 1920:1080 -c:a aac fullgame_Output.mp4
I have problem with live http flv stream. I'm using the following command and it works great except for the audio. Audio has delay that increasing over time. I can fix the this by removing refresh rate option on the output (-r 30) but then stream latency goes higher for about half second.
ffmpeg -f v4l2 -threads 0 -video_size 672X420 -i /dev/video1 -f alsa -thread_queue_size 512 -i hw:1,0 -c:a aac -ar 44100 -b:a 128k -c:v libx264 -s 672x420 -r 30 -g 60 -preset superfast -tune zerolatency -strict -2 -f flv rtmp://localhost/live/primary
Can someone explain why this happening and what possible fixes are?
On the client I'm using Chrome with flv.js library.
On the server Node-Media-Server.
I'm trying to send a stream of video that's coming into a BlackMagic DeckLink Studio 4K capture card over a few different RTMP streams at once with FFmpeg. The command that I am doing it with is this:
ffmpeg -re -format_code Hi59 -f decklink -i 'DeckLink Studio 4K' -map 0 -flags +global_header -vcodec libx264 -crf 25 -preset medium -pix_fmt yuv422p -acodec aac -f tee "[f=flv]rtmp://ip1/live/test|[f=flv]rtmp://ip2/live/test.
However, whenever I send this video out, I just get color bars when looking at the stream. I tried using a different video source (the testsrc supplied by FFmpeg), and that sends out fine over RTMP to multiple stream destinations.
Is there something weird with how tee and the decklink stuff work in FFmpeg? Or is there an issue with my command?
If you see colors bars, that means that ffmpeg is connecting to the card and streaming fine but the card is giving the bars. Your command says that ffmpeg is expecting 1920X1080#29.97 interlaced, make sure that is the format into the Decklink. You can also try explicitly setting the connection type, example:
ffmpeg -re -format_code Hi59 -video_input sdi -f decklink -i 'DeckLink Studio 4K' -map 0 -flags +global_header -vcodec libx264 -crf 25 -preset medium -pix_fmt yuv422p -acodec aac -f tee "[f=flv]rtmp://ip1/live/test|[f=flv]rtmp://ip2/live/test
If you are still running into issues, make sure that the BlackMagic software can see the video signal, and it's the format you expect.
One last thing to check, if its an HDMI input make sure it is not HDCP; it's not supported.