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<BtbN> you can consider support for those dropped, but it might just as well just work
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<BtbN> nobody tests it anymore
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<Juest> last time i tried it appears that ffmpeg 5.1 doesn't run on windows 7
<Juest> or 5.0.2
<Juest> i dont remember
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<kepstin> i'd honestly be surprised if you couldn't build and run modern ffmpeg on windows 7, but prebuilt ones might not work and you might have to disable some features if they require newer windows.
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<FlorianBad> If I was making -thread_queue_size proportional to something so that it stops complaining, but still doesn't run out of memory, what should it in theory be proportional to? I was thinking resolution surface (width x height)
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<FlorianBad> My script encodes for example 12 resolutions of input each to 5 bitrates of output (in 12 commands, 1 per resolution, no rescale). And of course with the low rest it's fine, but when we get to 8K footage it becomes difficult to nail the right thread_queue_size
<FlorianBad> And I can't just try and see because it takes absolutely forever. So that leads me back to my question: in theory to what should it be proportional?
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<FlorianBad> I know I already asked this but I'd like to give it one more try ;) > Is there a way to define the (approximate) duration of EACH INDIVIDUAL segments, so just like seg_duration but as a list (similar to what -force_key_frames does for keyframes)
<FlorianBad> (that's for audio, which is why I can't just use the keyframes for that)
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<FlorianBad> What I mean is something like: seg_duration_seconds_list=2.5,2.7,2,3.1,2.4 as opposed to an average: seg_duration=2.5
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<dv_> can SPS / PPS NALUs be inserted in front of any frame, not just IDR frames?
<dv_> in h264 I mean
<dv_> with intra refresh, you don't have IDR frames (other than the very first frame), so this sounds logical
<JEEB> you could verify the H.264 spec since it is free, but indeed I am not sure there is a requirement for them to be only at random access points (IDR or random access point SEI + I)
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<TuxJobs> I now have almost exclusively lossless FLAC files of the only music I want to listen to. But now I'm feeling nostalgic for how lossy MP3s sounded in the late 1990s and early 2000s. Since I haven't kept those ancient MP3s and they cannot be found online anymore, I'm trying to make copies of some of my FLACs as lossy MP3s to listen to sometimes. The command I use is `ffmpeg -i lossless.flac -b:a 32k lossy.mp3`, and it works, but I can't hear any of those
<TuxJobs> characteristic "pops" that always were present at seemingly random locations in all those old MP3s. Does ffmpeg have a feature to add those in?
<TuxJobs> I assumed that they were the result of the low bitrate chosen, but I also tried 16k and it sounds worse than I remember the old MP3s, yet zero "pops".
<kepstin> "pops" were likely caused by either download errors introducting artifacts or poor quality mp3 encoders. ffmpeg will normally use lame by default, which is a lot better encoder now than most people used in the 90s)
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<rodeo> you can't even build current git master with a very old lame, requires 3.98.3
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<rodeo> n0.5.10 has no official minimum lame requirement although that doesn't necessarily means it will work with any version
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<TuxJobs> :(
<TuxJobs> Was checksumming the downloaded files not done by programs like Napster?
<TuxJobs> Although I got the bulk of them from a LAN.
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<rodeo> in the 90s? possibly not
<furq> you want l3enc or mp3enc for the authentic napster sound
<furq> i doubt there's anything but windows binaries available though
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<furq> well i guess there's i386 linux binaries but i can't imagine those still work 27 years later
<Lynne> go older than that
<Lynne> vivo video + siren audio
<CounterPillow> furq: if they're static, they probably will
<furq> sadly i only see linux binaries for the demo version of mp3enc
<galad> probably some bad CD rips with read errors
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<bencc1> hi
<bencc1> I'm using ffmpeg to capture the screen
<bencc1> I pause and resume the capture by sending SIGSTOP and SIGCONT signals to the ffmpeg pid
<bencc1> in the video output the capture is frozen when I pause the video but the duration of the video includes the paused parts
<bencc1> I've tried to rewrite the timestamps with "-vf setpts=N/FR/TB" but it doesn't help
<bencc1> this is the original command
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<bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -draw_mouse 0 -framerate 25 -video_size 1280x720 -i :0.0+0,0 -acodec aac -b:a 96k -af aresample=async=1 -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 -vf fps=25 rec.nut
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<bencc1> and this is the command when adding setpts:
<bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -draw_mouse 0 -framerate 25 -video_size 1280x720 -i :0.0+0,0 -acodec aac -b:a 96k -af aresample=async=1 -af asetpts=N/SR/TB -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 -vf fps=25 -vf setpts=N/FR/TB rec.nut
<bencc1> what am I missing?
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<kepstin> indeed; write some stuff to cd, throw it in a box of sand and shake it a bit, then rip it on a windows xp box with windows media player and the fraunhofer mp3 encoder, should sound suitably terrible :)
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<Marth64> i dream of those days
<Marth64> bad quality but good times
<furq> you forgot that it should be a cd-r that was created from 128k mp3s in the first place
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<TuxJobs> Just to be clear, the "pops" I mentioned earlier seemed frequent and sounded more like some kind of sound effect from a NES game than any sound that a vinyl record would make while playing back a record.
<TuxJobs> I wish I had an example.
<TuxJobs> "Bad early MP3s" is apparently not nostalgic for too many more people besides myself.
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<kepstin> hmm, yeah, a recurring noise of that sort does seem like the sort of thing you'd get from a cd ripper using realtime or fast extraction and relying on the cd drive's builtin error hiding on a cheap drive, rather than anything specific to the mp3 encoding.
<kepstin> there's a reason projects like cdparanoia and exact audio copy exist; to prevent that sort of thing :)
<bencoh> so decode to pcm, replace random bits with 0s, and re-encode, basically
<kepstin> if you want to get tricky about it, you'd look into the cd audio format and make sure to align your random chunks of 0s with the frame size.
<kepstin> (and space them based on the typical length of music encoded in one revolution of a cd)
<kepstin> (might also try a "hold"; take a sample and repeat the same sample value for a period of time, instead of replacing with 0s - that would change the character of the glitch a bit, and might be more representative of how some drives worked)
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<rodeo> for all we know some of these came from the playback hardware and not the mp3 encode, I remember a long long time ago I bought a cheap CD player w/MP3 support and burned some CDs with multiple albums in MP3 format, and it sounded bad, but eventually I realised regular CDs sounded bad as well, with artifacts including cracks and pops; I returned it and exchanged for a high-end CD player also w/MP3 and the CDs I burned for the bad player
<rodeo> sounded about just fine on the new one
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<TuxJobs> I'm pretty sure it was not the playback, though.
<TuxJobs> I actually thought or assumed that this was a famous, known thing.
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<TuxJobs> "Oh, yeah, those weird pops/blops in early MP3s! I remember those!"
<tykling> the ntfs effect
<tykling> (and fat32 before that)
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<TuxJobs> tykling: Hmm?
<tykling> :) the blips and errors in my mp3s were from various crashes poorly handled by bad filesystems over the years, the problem kind of just stopped once the world moved to zfs and other checksummed and journalled filesystems
<tykling> same reason we used to get warez in a bunch of rar files with checksums and now we dont
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<bencc1> I'm trying to pause/resume screen capture with SIGSTOP/SIGCONT with continuous timestamps
<bencc1> but the video has gaps every time the ffmpeg process is paused
<bencc1> I've tried to add setpts and asetpts but it doesn't help
<TuxJobs> tykling: "from various crashes" <-- On the computers making the MP3s?
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<bencc1> this is my command with setpts and asetpts:
<bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -draw_mouse 0 -framerate 25 -video_size 1280x720 -i :0.0+0,0 -acodec aac -b:a 96k -af aresample=async=1 -af asetpts=N/SR/TB -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 -vf fps=25 -vf setpts=N/FR/TB rec.nut
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<furq> bencc1: did you check ffprobe -show_frames
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<furq> TuxJobs: i guess he means bit flips etc on any computer between the ripper and the downloader
<bencc1> furq: I'll check now
<furq> including through bad piracy software or an overloaded switch at the lan party that was set up by someone's mate who reckons he knows what he's doing
<furq> and that ends up ruining the quake tournament
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<furq> but what you described sounds like ancient mp3 encoder artifacts to me
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<furq> they were not very good in the 90s especially since they insisted on the lie that 128k cbr was CD QUALITY
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<BtbN> I doubt anything in ffmpeg is prepared to sensibly handle SIGSTOP for a live capture
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<furq> if it works at all i guess you'd need -fps_mode vfr
<bencc1> BtbN: so how can I pause and resume x11grab and pulseaudio capture?
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<BtbN> stop the whole process, and start it again later
<bencc1> furq: I'm trying to get a constant framerate
<BtbN> and then stitch together the two videos
<TuxJobs> furq: That just doesn't add up. You're talking about the 1990s... not some sort of ancient, alternate steampunk timeline with crude metal pipes and stuff...
<bencc1> will I get constant audio and video framerate after concatenating the videos?
<furq> you can't have cfr if there are seconds-long gaps though
<TuxJobs> Things swoooshed through the network cables reliably on LANs.
<furq> without dup frames
<bencc1> furq: I thought I can rewrite the pts
<furq> TuxJobs: yeah like i said i think it's probably artifacts
<TuxJobs> furq: But you're implying that it's from unreliable network transfer and whatnot?
<furq> it could be
<furq> or just people having jank files and then sharing them
<bencc1> here the suggest to use SIGSTOP and SIGCONT
<bencc1> and here the suggest to use setpts=N/FR/TB
<bencc1> you are saying that it is not supposed to work?
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<TuxJobs> ChatGPT just answered: 'The "pops" or "blip" sounds that were often heard in MP3 songs in the late 1990s were generally a result of the encoding process used at that time. MP3 is a lossy audio compression format, which means it discards some audio data in order to reduce file size. This compression technique employs a "psychoacoustic model" to determine which sounds are less perceptible to human hearing and can be safely removed. During the encoding
<TuxJobs> process, data related to certain sounds might be discarded or altered. This can sometimes create artifacts or glitches in the audio, leading to the presence of pops or blips in the resulting MP3 files. These artifacts are typically more noticeable when higher levels of compression are used or when the encoding algorithm is less sophisticated. Furthermore, in the late 1990s, when MP3 technology was still relatively new, encoding software and algorithms
<BtbN> it's not anything anyone ever thought about when writing the code
<TuxJobs> were not as advanced as they are today. This also contributed to the prevalence of those glitch sounds in MP3 files during that era.' But who knows if that's just made up nonsense like it often is?
<BtbN> Can you not?
<furq> that sounds like a much more verbose version of the guess that i made
<TuxJobs> Damn FFMPEG for using such a sophisticated MP3 encoder!
<furq> which is better than those things normally manage
<furq> i guess you didn't ask it to do any maths
<TuxJobs> furq: You were talking about network transfers corrupting the files, though? Not the encoding process?
<furq> i said it was probably artifacts and earlier i mentioned using l3enc or mp3enc if you want that sound
<furq> other people said it was bad cd rippers or old network transfers with no checksumming onto filesystems with no checksumming
<furq> idk what you mean by pops so it could be all of the above
<TuxJobs> I wish I could just physically travel back to that time and hear for myself, at my old computer... ;S
<furq> there are still l3enc and mp3enc windows binaries online
<TuxJobs> If I could find an example MP3 song with the glitch in it, I would've loved to link to it.
<furq> as well as a million other awful mp3 encoders
<furq> i think mp3enc is the correct one for the era though
<furq> there was a windows acm component for it that was very commonly pirated back in the day
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<TuxJobs> I think I have a Windows 98 VM left to use it on.
<furq> it'll probably run in wine
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<TuxJobs> Well, that shall be an experiment for tomorrow. 'night.
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<bencc1> furq: I have run ffprobe -show_frames. what should I look for?
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<bencc1> I don't understand why I can't rewrite the pts to make the timestamps continuous
<bencc1> the number of frames when I send SIGSTOP and when I don't is the same
<bencc1> if I'll create separate video files and stich them together, will I get constant framerate?
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<bencc1> shouldn't it work?
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<FlorianBad> If anyone is interested in very high quality test/encoder-challenging footage I gathered a few videos downloaded from Youtube here: https://florian.club/pub/video-test-footage/
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<another|> "very high quality" "downloaded from Youtube"
<another|> lol
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<FlorianBad> another|, well, yeah they're pretty damn good if your goal is to test codecs because it's well beyond the kind of loss you are trying to notice when encoding, especially since you'll obviously downscale that a bit
<FlorianBad> Also you would't want a 2TB file for 2 minutes of test video
<another|> not even 8k 420p would be 2TB
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<BtbN> Let me introduce you to the Phantom T4040
<BtbN> "The industry’s fastest 4Mpx camera at over 9300 fps at 2560 x 1664 and up to 444K fps."
<another|> okay, that's a super specific use case
<furq> what an ugly sentence
<BtbN> Yeah, it will not be 2560x1664 anymore at 444k fps
<bencc1> I'm trying to use select between to pause/resume screen capture:
<bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -draw_mouse 0 -framerate 25 -video_size 1280x720 -i :0.0+0,0 -filter_complex "[0:v]select='between(t,0,5.0)+between(t,10.0,15.0)',setpts=N/FR/TB,fps=25;[0:a]aselect='between(t,0,5.0)+between(t,10.0,15.0)',asetpts=N/SR/TB,aresample=async=1" -acodec aac -b:a 96k -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 rec.nut
<bencc1> I'm getting an error: Stream specifier ':v' matches no streams
<bencc1> what's wrong with my syntax?
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<BtbN> input 0 is pulse, which has no video
<bencc1> ok. I'll try to switch them
<BtbN> so 0:v matches no streams
<BtbN> well, if you switch them, input 0 will have no audio :D
<furq> also you need to explicitly map those outputs
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<BtbN> You need to use the correct selectors
<bencc1> yes. I switched and changed video to 1
<another|> ...
<bencc1> what do you mean by "map those outputs"?
<another|> if you switched them, video won't be input 1
<bencc1> because I'm getting: aresample:default has an unconnected output
<bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -draw_mouse 0 -framerate 25 -video_size 1280x720 -i :0.0+0,0 -filter_complex "[0:a]aselect='between(t,0,5.0)+between(t,10.0,15.0)',asetpts=N/SR/TB,aresample=async=1;[1:v]select='between(t,0,5.0)+between(t,10.0,15.0)',setpts=N/FR/TB,fps=25" -acodec aac -b:a 96k -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 rec.nut
<bencc1> this is my command now
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<FlorianBad> BtbN, haha a phantom :)
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<bencc1> it's working now. thanks
<bencc1> Is there a syntax for specifiying infinity or end of stream when using select=between(t,0,5.0) ?
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