BtbN changed the topic of #ffmpeg to: Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: https://ffmpeg.org/bugreports.html | Wiki: https://trac.ffmpeg.org/ | This channel is publically logged | FFmpeg 7.0 is released
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<chiselfuse> JEEB: but what's the point? how is it useful to know
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<BtbN> How else would you tell two libx264 encoders apart that are working in parallel?
<chiselfuse> oh so it's meant as an identifier?
<chiselfuse> i thought it was for debugging in gdb or something
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<BtbN> You can use it for that as well I guess. It's the pointer to the AVCodecContext after all
<BtbN> or whatever the log context happens to be
<BtbN> But you don't want to look at that context thaaat often for debugging
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<Guest0> which is better for naming photos and videos?
<Guest0> (4 digit year)(2 digit month)(2 digit date) (24 hour)(2 digit minute)(2 digit second) (File counter).(Extension)
<Guest0> OR
<Guest0>  (4 digit year)(2 digit month)(2 digit date)_(24 hour)(2 digit minute)(2 digit second)_(File counter).(Extension)
<Guest0> the only difference between the underscore vs the space
<Guest0> versus hyphens and dots
<BtbN> Whatever you prefer
<BtbN> spaces always need escaped or quotes
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<aaabbb> Guest0: i prefer underscore for filenames generated automatically
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<noobaroo> Most vids are h264 so I can use the h264_metadata bsf, but with basically all of the open-source codecs the only way I can set crop into mkv container is by using mkvtoolnix. It then shows up in ffprobe as sidedata for the video stream specifier. But IIRC -vf sidedata in ffmpeg does not work with codec copy. So I havent even tried to look if there is a way to set this cropping sidedata on cmdline ffmpeg.
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<noobaroo> I have a 12GB video file right now that Im using mkvtoolnix to add the cropping sidedata and then im going to remux again with ffmpeg since mkvtoolnix cant do cues_to_front = true, and its a 12GB file so this is even more important because of the size
<noobaroo> I considered doing it in tmpfs so it doesnt take 10mins but im not sure if i have enough ram.
<noobaroo> Is there a way to add cropping sidedata with ffmpeg? I dont see it under matroska section in the ffmpeg.org docs
<noobaroo> and IIRC -vf sidedata doesnt work with stream copy
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<noobaroo> If something says "1920x1080 [SAR 1:1 DAR 16:9], SAR 20:27 DAR 320:243, 23.98 fps, 23.98 tbr, 1k tbn (default)" Does that mean the real SAR and DAR are the ones in the brackets or is it the one right after?
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<aaabbb> noobaroo: it means that the ratio stored in the container and the ratio stored in the bitstream differ. one is not automatically correct. but for a hd1080 video, you can probably assume that the sar 1:1 one is correct and the sar 20:27 one is not
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<noobaroo> Which one is the container?
<noobaroo> I guess its the 1920x1080
<furq> xx: those are colour primaries, not pixel formats
<furq> those are stored in the container
<furq> they should also be copied from the source with recent enough ffmpeg
<xx> yeah I realized it some time later that pixel formats are not the same thing, my bad
<xx> still strange that they change the color primaries (I'm unsure what it is anyway)
<furq> well i guess next time i won't take 24 hours to reply
<xx> recent enough ffmpeg would be what version?
<furq> no idea
<furq> about a year old maybe
<furq> it used to not set any primaries unless you set them explicitly
<furq> now it tries to copy them from the source
<furq> i guess what mediainfo is reporting is some default value from somewhere
<furq> hard to say because 0x0 is down and i can't see those pastes
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<noobaroo> i have no idea where ffprobe is getting the "SAR 20:27 DAR 320:243" from
<noobaroo> mediainfo and mkvtoolnix only show 1920x1080 and the raw video stream to a file only shows 1080
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<noobaroo> Is the only reason we still use ac3/eac3 is for compatibility? Even if thats the case in 2024, why in all of time did nobody ever use ac3/eac3 for music? Is it that bad?
<noobaroo> I would think a movie audio codec would need better VBR than music codecs. You know... Because of so many silent or quiet scenes for several seconds at a time
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<kepstin> iirc the main use cases for ac3 have actually been for broadcast and optical discs, where vbr can at times be counterproductive since you have limit bitrate for transfer speed.
<kepstin> as far as I understand it, ac3 is fine for music, but other codecs are more efficient (lower average bitrate for same quality)
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<BtbN> Yeah, movies typically have super strict cbr requirements.
<BtbN> I think the ac3 family of codecs stems from cinema even? Where the physical film reel is rolling at constant speed, so the digital audio on there had to be perfect cbr no matter what's going on
<BtbN> And in broadcast TV it's also strict cbr, though mpeg-ts can pad the data just fine
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<noobaroo> I understand the physical layout difference of 5.1(side) vs 5.1, at least I think, it seems pretty simple and true to the name. But why is there a different ffprobe channel map layout? Apparently 5.1=FL+FR+FC+LFE+BL+BR vs 5.1(side)=FL+FR+FC+LFE+SL+SR and in audacity, the program doesnt ever name channels or care about intended channel maps, it just numbers them. The first 4 channels are the same between these, with both using 2 as the main
<noobaroo> dialogue channel and 3 as LFE. The only difference is the last two channels
<noobaroo> And its really a superficial difference because maybe I have my speakers slightly on the side and slightly behind. What matters is that SL and BL are both in the same channel number. And SR and BR are in the same spot
<noobaroo> I dont understand why there is a software labeling difference
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<ePirat> noobaroo, I would assume ffmpeg follows some of the standards for that there
<JEEB> plus certain definitions of channel layouts between different specifications. doesn't help that then retroactively like 10 years later one of the specs retroactively redefines stuff and you're left with a mess :)
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<furq> 15:30:23 ( BtbN) I think the ac3 family of codecs stems from cinema even?
<furq> the one in the middle
<Marth64> yes always thought that was cool
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<iive> furq, Even the emblem in the middle looks like Dolby Digital
<furq> that's because it is
<iive> :DD
<JEEB> both of the D companies had their digital formats first in cinema
<furq> yeah that's the DTS timecode on the right hand side
<furq> to sync with an external cd with the audio on it
<furq> and SDDS on the left if you didn't look the photo up already
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<realies> is it expected for loudnorm to clip the output when asking for i=-19,lra=6,tp=-1? getting a true peak of 3.5 dbfs on the file im testing
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<realies> top track - after loudnorm, bottom track - input to loudnorm; could this be because of a potential DC offset? https://s3.amazonaws.com/i.snag.gy/eF03zP.jpg
<kepstin> are you sure you actually want loudnorm (dynamic loudness normalization) rather than a fixed offset based on the measured ebur128 integrated loudness?
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<realies> i want to conform to the specific ebur128 specs using that input file
<kepstin> if this is a file and you don't need the dynamic range adjustment stuff, I'd recommend doing a 2 pass measurement first using the ebur128 filter to do measurements, then a second pass applying any needed gain adjustment using the volume filter.
<realies> i think it would be better not to dynamically change the loudness over time, but that sounds like tweaking the dynamic range
<realies> i should have mentioned, this is a two pass loudnorm where the first one takes measurements and the second one applies them to the file
<kepstin> the two-pass loudnorm will, depending on the measurements and settings automatically pick between either applying a single gain adjustment or do the second pass in dynamic mode.
<kepstin> mostly if the measured dynamic range is larger than your settings, it'll run in dynamic mode to reduce dynamic range.
<realies> yeah, i tried linear=true/false but the result was identical
<realies> i'm not sure if that what you refer to
<realies> that's*
<realies> just for a sanity check, this is how i've implemented the two-pass run: https://bin.reali.es/ditu/slim
<kepstin> linear=true (the default) will cause it to do a linear adjustment _if possible_. in your example file it's apparently not possible to do linear mode while fulfilling your requested parameters, so it's running the second pass in dynamic mode.
<realies> (i've also tried without setting linear= explicitly)
<kepstin> setting linear=false forces it to always do dynamic mode even when linear is possible. The default is to use linear when possible.
<realies> i understand, but the output is identical with linear=false and linear=true
<kepstin> yes, because with your input file and chosen settings, linear mode is not possible.
<kepstin> so it always uses dynamic (non-linear) mode
<realies> isn't dynamic mode expected to reduce the dynamic range instead of clip the output?
<realies> these are the measurements and normalisation details for this specific file: https://bin.reali.es/PDgS/slim
<kepstin> it's not perfect :/ sometimes get some weirdness on specific audio.
<realies> this is actually weird
<realies> "output_tp": -1, "output_lra": 3.8,
<kepstin> if you don't need dynamic range reduction, i continue to recommend using ebur128 (or loudnorm first pass) to get measurements, then apply gain adjustment with the volume filter.
<realies> using the ebur128=peak=true filter on the output file tells me Integrated loudness: I: -16.0 LUFS Threshold: -26.5 LUFS
<realies> Loudness range: LRA: 4.1 LU Threshold: -36.6 LUFS LRA low: -18.8 LUFS LRA high: -14.7 LUFS True peak: Peak: 3.5 dBFS
<realies> why would ebur128 say TP 3.5 dBFS but loudnorm "output_tp": -1?
<kepstin> it looks like the issue is that your target integrated loudness is -16 but the file is measured as -18, so it has to increase the volume to meet the loudness. But the true peak of 0.01 exceed the target true peak, even more so after increasing volume, so it has to run in dynamic mode in order to apply the peak limiter to bring peaks down to -1
* realies goes checking the way things are parsed
<kepstin> output tp is tp _after_ the peak limiter is applied to bring peaks down to your requested -1dB
<realies> why would the ebur128 filter report TP 3.5 dBFS on the output file when loudnorm has a peak limiter?
<kepstin> so it successfully did what you asked - raised loudness from -18.93LUFS to -16.02LUFS while reducing true peak from 0.01dBFS to -1dBFS
<kepstin> i dunno, bugs in the loudnorm peak limiter? :/
<realies> fair enough
<realies> it has been getting better over the years but it's still not universally great it seems
<realies> do you know of any other cli tools that can do ebur128 normalisation?
<realies> was just going through this https://github.com/sdroege/ebur128
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<kepstin> if you just want to do loudness normalization, ffmpeg's ebur128 and volume filters are fine. If you want to do loudness normalization + peak limiting, then use the volume filter followed by the alimiter filter (if you need to limit true peak, then upsample before the alimiter filter)
<kepstin> you only need something fancier if you specifically want to do dynamic range reduction
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<realies> as i said, i need to conform to i: -16, tp: -1, lra: 6
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