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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<zayd>
Any idea what I'm doing wrong here? https://bpa.st/MVMQ I'm trying to transcode DTS-HD to Opus but it fails with something about channel layout. FLAC works but not Opus for some reason.
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<aaabbb>
what situations is libx264's aq-mode=2 worse than aq-mode=1? for libx265 2 is the default
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<clever>
ive been messing with the background recording feature in steam, and its generating a session.mpd that looks like normal DASH streaming stuff, if i host the directory over http, i can then playback the recording normally
<clever>
how can i convince ffmpeg to accept the same thing without the http in the middle?
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<noobaroo>
did you guys know the wikipedia pic of vincent van vogh's painting is over 200MB?
<noobaroo>
lximage-qt refuses to open it but swayimg handles it fairly smoothly. Its somewhat choppy with antialiasing turned on, but swayimg always is. Oh actually nvm they added multithreaded antialiasing very recently, non-200MB sized pics arent choppy
<noobaroo>
A 200MB .jpg file
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<aaabbb>
noobaroo: there are jpegs that are hundreds of gigabytse out there
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<noobaroo>
The biggest jpeg i ever seen before this was probably 20MB or so
<noobaroo>
Almost identical, there is a hair of difference in the upper bandwidth area on one spot, except for this one microscopic spot, everything else is identical at least to my naked eye
<JEEB>
I have no idea about what extensions you're possibly talking about, but at the point of 2016 the decoder was rewritten by a nice person called foo86, and it supported everything defined at that point to my knowledge.
<JEEB>
the reason why the object based stuff is atm ignored is because we have no means of parsing that decoded result or doing object based rendering :P
<noobaroo>
So is going from DTS-HD to wav/flac or even high bitrate opus, the same as going from the bsf:a dca_core of it to wav/flac/opus ?
<JEEB>
no?
<JEEB>
as I said, the lossless etc extensions are supported. the full shebang gets decoded unless there are new extensions after 2016
<JEEB>
but what is called "DTS-HD" is just usually the lossless extension
<noobaroo>
So audacity visualization is just a general sense and not very accurate then
<noobaroo>
Because decoded, the two appear 99.9% identical
<noobaroo>
At least in this one 8 second area
<JEEB>
which of course is possible
<JEEB>
if the lossy result is that close to the lossy one
<JEEB>
argh, one of those is *lossless
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<noobaroo>
JEEB do you know if its possible to add this with ffmpeg or is this something only mkvtoolnix can do at the moment? "Side data: Frame cropping: 0/0/0/358" <-- shows up like this in ffprobe
<noobaroo>
h264_metadata bsf crop is a lifesaver but none of the open source codecs have this bitstream option, so the only way is using mkvtoolnix
<JEEB>
currently side data override support is only for the display matrix stuff on CLI. the problem is mostly that there's no centralized text -> side data logic. if there was, it'd already work
<JEEB>
in other words, the structured metadata (side data) exists, and there is code that logs it either in ffprobe or elsewhere, but not something that takes in a structure and creates side data from it
<welder>
I've got a file with 2 streams (1 audio 1 video), av_read_frame returns sometimes first, sometimes second. Can I ask just to only read the audio stream? I see av_seek_frame for instance takes stream id as argument, does av_read_frame have similar equivalent?
<welder>
Not a big deal tho, I can just if(...) continue to skip the video packets
<JEEB>
yea, all packets are read from the container according to interleaving (and some internal interleaving logic if the container interleaving is bork)
<JEEB>
ahh, there might be AVStream->discard |= AVDISCARD_ALL
<JEEB>
that might make the demux logic drop it before it gets passed to you from reading
<welder>
Noted
<JEEB>
some modules have custom logic for discard all, which means that packets might get discarded even earlier in the process
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<realies>
kepstin, this was a stupid user error; i was doing -ar 44.1khz as part of the second pass loudnorm and this introduced new peaks from the resampling...
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<kepstin>
i honestly wouldn't have expected resampling down to increase the peaks that much, tbh :/
<realies>
me neither , but maybe it doesn't use high precision resampling by default
<realies>
i'll try replacing -ar 44100 with aresample=44100:resampler=soxr:precision=33:osf=dbl:filter_size=1024
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<furq>
filter_size is only for swresample
<realies>
thanks
<realies>
now i wonder how to retain the integrated loudness and true peak values when converting to mp3 lol
<kepstin>
integrated loudness shouldn't have any significant changes; true peak will change somewhat as a result of lossy encoding, nothing you can do