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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<DeHackEd>
jafa: normally you shouldn't need to specify the number of channels...but when the input has multiple streams of the same type, i think ffmpeg only picks one... did you map them properly?
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<jafa>
mapped correctly - both channels are reported and both channels work if I specify the channel count
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<noobaroo>
Is the automatic muxer that gets chosen when a output file ends with .webm just the same matroska one with `-dash true` ?
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<aaabbb>
noobaroo: webm is just the matroska muxer but limited to only certain features and codecs
<aaabbb>
and with a slightly different header
<noobaroo>
Yes, so its the same exact matroska muxer but with `-dash true` ?
<noobaroo>
Or it has its own?
<JEEB>
it's in the same file, libavformat/matroskaenc.c , and I don't think the dash option is related? since that is a very specific thing and not even related to base webm spec
<aaabbb>
yeah it has nothing to do with dash
<aaabbb>
you can do webm with dash or without
<aaabbb>
noobaroo: both matroska and webm are just a type of format called ebml
<JEEB>
anyways, if you want to look at the differences, check how the mode gets set and CRCs get disabled in mkv_init, and afterwards there's a bunch of IS_WEBM macro usages
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<noobaroo>
Im looking at https://ffmpeg.org/ffmpeg-all.html#matroska where it says "Create a WebM file conforming to WebM DASH specification. By default it is set to false." and the parameter to do this is `-dash <bool>`
<JEEB>
yes, that is specific to WebM DASH, not WebM itself
<noobaroo>
aaabbb I also thought it had nothing to do with dash. JEEB Oh, okay, so then ffmpeg to .webm has a completely separate webm muxer ?
<aaabbb>
no it's the matroska muxer
<JEEB>
there is a `is_dash` variable in the muxer if you want to see the difference in writing for that :P
<aaabbb>
.webm is literally matroska with a few limitations
<noobaroo>
I know what it is, im asking about ffmpeg libraries
<aaabbb>
noobaroo: the webm muxer in ffmpeg is the matroska muxer with some limitations
<aaabbb>
how come the matroska muxer doesn't support lacing anyway? it's kinda wasteful
<JEEB>
> put_ebml_uint(pb, MATROSKA_ID_TRACKFLAGLACING, 0); // no lacing (yet)
<JEEB>
in other words most likely not added initially and then nobody added it
<JEEB>
watches pelcom
<JEEB>
*pelcome
<aaabbb>
JEEB: yeah i read the code before and saw that. i've been thinking of porting over the ebml and xiph lacing from libmatroska i think it was (or whatever mkvtoolsnix uses)
<noobaroo>
aaabbb I did notice mkvmerge makes smaller files than ffmpeg, the lacing is why? i thought the mkvmerge docs said without lacing the file size would be huge, guess that was just my impression and it just said 'bigger'
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<noobaroo>
I feel like it specifically used the term "very large" though...
<JEEB>
you'd really have to diff the results structure wise to tell why A is bigger than B
<noobaroo>
But does that mean mp4 doesnt do lacing? Because ffmpeg's mkv files without lacing are still smaller than ffmpeg mp4 files
<aaabbb>
JEEB: it is for that reason, lack of lacing
<aaabbb>
noobaroo: lacking is an ebml thing. mp4 is not ebml, it's another type of format called isobmff
<aaabbb>
s/lacking/lacing/
<aaabbb>
JEEB: mkvinfo -a is a good way to diff them, shows that mkvmerge does ebml lacing by default, ffmpeg matroska muxer does no lacing
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<aaabbb>
noobaroo: if you want to optimize ffmpeg's mkv/webm files, just remux with mkvmerge
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<noobaroo>
i have a file right now that i put 'FFTStoNIXtoFFMKV' in the filename, because its catted from multiple files, and through trial and error i had to use ffmpeg to make them mpegts first, and then nix to concat them, and then ffmpeg to move cues to front
<aaabbb>
you could have done that all with mkvmerge ;)
<aaabbb>
which is its raison d'etra
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<noobaroo>
JEEB btw i saw your name in a forum from 2012, i didnt know you are this old, you are probably like 50+ huh? in 2012 i was in middle school
<JEEB>
pffft
<JEEB>
I'm internet old, but not that old
<noobaroo>
How? I havent figured out how to move cues to front with mkvmerge yet
<aaabbb>
noobaroo: i think you can do that with a debug option?
<aaabbb>
JEEB: i've always wondered actually, are you responsible for -tune touhou? :D
<noobaroo>
And also appending the direct files with mkvmerge makes the video short by 20 seconds, i think its because its VFR
<aaabbb>
i saw your avatar ages ago and figured might be
<JEEB>
aaabbb: lol, no, I am not. the "optimizing for töhö" thing was a really old thing
<JEEB>
the maintainers of x264 liked töhö, and liked how the gameplay area was fitting in 16x16 macroblocks
<noobaroo>
But mkvmerge does handle concatting mp4 files directly, way better than ffmpeg does, when i tried with ffmpeg the video was 11+hours long and all broken
<JEEB>
the bullet hell was really nice hard thing to optimize, too
<noobaroo>
Still, both seem to require concatting from mpegts to do it without bugs
<aaabbb>
JEEB: just thought since you had a moku avatar :D
<aaabbb>
noobaroo: are they encoded with the same settings?
<noobaroo>
aaabbb How can i check for lacing with mkvinfo -a ?
<noobaroo>
All it did was spam my terminal with the clusters
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<noobaroo>
Probably but not definitely, its from the same livestream within 10 minutes of each other
<aaabbb>
noobaroo: you don't actually need -a
<aaabbb>
just do mkvinfo blah.mkv | grep Lacing
<aaabbb>
that won't guarantee it... because iirc the flag can be zero but lacing is still there
<noobaroo>
And the files are downloaded with streamlink, i dont think they are remuxed at all, or if so, its absolute minimum modifications possible. The times start at 37:00 and 44:00 (example, not actual, i didnt look)
<noobaroo>
Offset from whenever the stream started. And the files are from m3u8 link but they are not mpegts, they are .mp4 files but they show up to `file` command as "Data" and they arent playable or readable by ffprobe
<aaabbb>
btw lacing won't fix your compatibility issue, that's just about packing data more efficiently
<noobaroo>
I got that, all of them show 0. I thought mkvmerge does lacing by default?
<aaabbb>
noobaroo: then do -a and scroll down to all the "simple block" and see if there are multiple "frame with size X" under each simple block, or if there is only one per
<noobaroo>
Without lacing, there is only one per?
<aaabbb>
right
<aaabbb>
lacing is just a way to not have to repeat the overhead of a bunch of headers
<noobaroo>
I see, well remember my "FFTStoNIXtoFFMKV" file i mentioned above? it has one per, but the source mkvmerge has many per, but both are 88.1MB, probably some kB difference but seems like lacing doesnt affect much
<aaabbb>
it's not huge huge overhead no
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<bryanodonoghue>
hello ffmpeg
<bryanodonoghue>
ffmpeg -hwaccels
<bryanodonoghue>
vdpau
<bryanodonoghue>
vaapi
<bryanodonoghue>
drm
<bryanodonoghue>
vulkan
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<bryanodonoghue>
on my aarch64 qualcomm laptop
<bryanodonoghue>
doesn't list v4l2m2m
<bryanodonoghue>
instead I need to say something like
<bryanodonoghue>
taking zoneminder as an example use-case
<bryanodonoghue>
it zoneminder has a libffmpeg integration
<bryanodonoghue>
where you specify one of the hwaccel methods via a web dialog box
<bryanodonoghue>
"vaapi"
<bryanodonoghue>
or "cuda"
<bryanodonoghue>
whatever ffmpeg -hwaccels throws up
<bryanodonoghue>
is there a value in a v4l2m2m wrapper ?
<bryanodonoghue>
and if so should that wrapper be specific to ffmpeg or a library that ffmpeg talks to
<bryanodonoghue>
...
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<another|>
there is no libffmpeg
<another|>
v4l2m2m is AFAIK not implemented as a classic hwaccel for reasons that are beyond me.
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<HGStyle>
hello, can i ask for help with ffmpeg? i am using it to convert a mp4 h264 video to mp4 h265, and it says "conversion failed" "error muxing a packet" "not yet implemented in ffmpeg, patches welcome". it seems weird to me because the video plays just fine in vlc by example, which, from my knowledge, uses ffmpeg internally. (i can post the full error
<HGStyle>
i am using a dev build from yesterday, if i remember
<JEEB>
HGStyle: the error is when writing so it's not the source mp4 but somehow the packet received from libx265 wrapper
<JEEB>
and VLC uses its own mp4 reader
<JEEB>
mpv is one thing that utilizes FFmpeg's one
<HGStyle>
oh okay
<JEEB>
(also you can make vlc use the FFmpeg one, but that requires arguments to override demuxer)
<JEEB>
anyways, can you post the full log with `-v verbose` to a pastebin site of your choice and link here?
<HGStyle>
so if i export it to a mkv file instead of exporting it to mp4 it might work?
<HGStyle>
okay
<JEEB>
also that mp4 pts/dts error happens in the following case: > (uint64_t)pkt->dts - pkt->pts != (int32_t)((uint64_t)pkt->dts - pkt->pts)
<JEEB>
which sounds like you got some really funky timestamps :D (thus `-debug_ts` might become useful in addition, although that will spam a lot more, `2> funky_log.log` or so might come in handy instead of trying to read it directly from the terminal)
<HGStyle>
it is a file i recorded on my phone using some google play app called xrecorder, and everytime i use a video recorded with that app inside ffmpeg i get warning/info messages but no error
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<llyyr>
what's your ffmpeg version
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<HGStyle>
ffmpeg version N-117740-g7f51cf75c6-20241110
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<noobaroo>
hey guys, im playing around with the bit depths my sound card supports (which is an ordinary one), and possibly the sample rates it supports (not sure if this is correct terminology at the hardware level or if its only applicable to the alsa/pipewire/wireplumber software layers)
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<noobaroo>
Anyway, I havent really made any successful changes, im looking at /proc/asound/card0/pcm0p/sub0/hw_params and i cant get it to say anything other than S32_LE and 48000 even though im playing in audacious and i have output bit depth at 24 and when i play in mpv cli it says "AO: [pipewire] 44100Hz stereo 2ch floatp"
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<noobaroo>
But, in my troubleshooting/research i have discovered an old friend at https://www.alsa-project.org/main/index.php/Asoundrc , I used to use pure alsa + dmix before pipewire was a thing because I heard pulseaudio caused terrible latency and audio degradation
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<noobaroo>
I'm wondering if today in 2024 is pure alsa + dmix still relevant? I think its not on by default, it would also be redundant to have 2 layers of mixing running all the time since everyone uses sound servers by default, pipewire or if not, pulseaudio is probably running, so dmix feature would be stupid to have on by default
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<JEEB>
I think it's telling that for various audio use cases for low latency etc, JACK has been utilized often :P
<JEEB>
so you have /something/ there instead of just pure alsa
<noobaroo>
So alsa's dmix is probably not very good then
<JEEB>
and pulse and/or pipewire latency in a lot of stuff had been up to how much buffers you give things. since xruns (buffer under or overruns) is not something you want to happen in general desktop use
<noobaroo>
Is jack better than pipewire? its older i think
<JEEB>
I think the arch linux wiki article for Professional Audio isn't completely bullshit in the choosing a sound server point, so feel free to read through that
<noobaroo>
are you in #pipewire by any chance? if not, do you think you could join and take a look at my question please
<JEEB>
thanks but no thanks
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