michaelni changed the topic of #ffmpeg-devel to: Welcome to the FFmpeg development channel | Questions about using FFmpeg or developing with libav* libs should be asked in #ffmpeg | This channel is publicly logged | FFmpeg 7.1 has been released! | Please read ffmpeg.org/developer.html#Code-of-conduct
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<mkver> Clang-ASAN reports a stack buffer overread in checkasm-sw_rgb here (in check_yuv2packed1, yuv2rgb_1_c_template).
<fflogger> [newticket] steven1: Ticket #11407 ([undetermined] heap-buffer-overflow vulnerability find in in function mov_read_trun at /ffmpeg/libavformat/mov.c:5944) created https://trac.ffmpeg.org/ticket/11407
<mkver> Seems like I sent a patch twice; sorry for that.
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* Compn facepalms
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<fflogger> [editedticket] MasterQuestionable: Ticket #9412 ([avformat] HEVC fails to seek on keyframe) updated https://trac.ffmpeg.org/ticket/9412#comment:18
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<fflogger> [editedticket] quinkblack: Ticket #9412 ([avformat] HEVC fails to seek on keyframe) updated https://trac.ffmpeg.org/ticket/9412#comment:19
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<mkver> haasn: Your check_yuv2packed1 added in c601bb8df5ae has a stack-buffer-overflow.
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<mkver> If one uses (int16_t *)src_u + i*INPUT_SIZE, the stack-buffer overflow is fixed; but I don't get why you use int32_t for data that is only treated as int16_t later (this is a effective-type violation and therefore UB).
<mkver> Is it because it simplified initialization?
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<fflogger> [editedticket] Balling: Ticket #9412 ([avformat] HEVC fails to seek on keyframe) updated https://trac.ffmpeg.org/ticket/9412#comment:20
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<meego> Quick call for freelancer: a project at my company could really use a faster loudnorm filter (see https://ffmpeg.org/ffmpeg-filters.html#loudnorm). Current implementation is capped at x≈20 faster than realtime, but some other implementations go as high as x230. We pay well and fast, and will offer bonuses if x100 and x200 milestones are achieved.
<Compn> ooo
<Compn> meego, wait around here for a devel or drop your contact info if you like
<Compn> (though this channel is publicly logged)
<Compn> we also have trac.ffmpeg.org if you'd prefer putting your bounty/job in a more solid place than irc. also can post to ffmpeg-devel mailing list if you wish
<Compn> the ffmpeg-devel mailing list is the preferred and most read place for developers in this project.
<Compn> send a mail to ffmpeg-devel@ffmpeg.org and i can just approve it for you. i think it should work
<Compn> (we're not all on irc)
<meego> Compn: thanks for the advice. I'm reachable at tech@spectre-music.com but i'll stick around here. I wasn't sure if the ML was the right place for this.
<Compn> no problem :)
<meego> I'll fill a bug on trac and an email on ffmpeg-devel
<Compn> its all just options, whichever you like :)
* Compn afk
<fflogger> [editedticket] plondino: Ticket #11338 ([ffmpeg] Error transcoding from dv source - fps 60000 is too large) updated https://trac.ffmpeg.org/ticket/11338#comment:2
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<fflogger> [editedticket] jun66j5: Ticket #11386 ([avformat] Subtitles and MOV are not working anymore) updated https://trac.ffmpeg.org/ticket/11386#comment:3
<fflogger> [newticket] guillaumekh: Ticket #11408 ([avfilter] loudnorm performance improvement) created https://trac.ffmpeg.org/ticket/11408
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<another|> NanananaLeader: seen !11408 ?
<NanananaLeader> loudnorm implementation is more than 100x realtime in librempeg, the dynamic mode
<another|> there was a bug bounty offered
<NanananaLeader> scanning + volume pass is also > 300x faster (true peak) or > 690x (sample peak) (>1000x none peak), so for one pass just use ebur128 filter + volume filter, but this needs manual intervention
<NanananaLeader> or scripting
<NanananaLeader> i doubt the bounty is about dynamic loudnorm mode
<NanananaLeader> so user is probably unaware of ebur128 + volume  way
<another|> <meego> Quick call for freelancer: a project at my company could really use a faster loudnorm filter (see https://ffmpeg.org/ffmpeg-filters.html#loudnorm). Current implementation is capped at x≈20 faster than realtime, but some other implementations go as high as x230. We pay well and fast, and will offer bonuses if x100 and x200 milestones are achieved.
<Marth64> i always found the manual intervention for second pass cumbersome
<NanananaLeader> the biggest slowdown in loudnorm is truepeak (aka resampling to 192000 sample rate and that is done two times, for both input and output samples)
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<NanananaLeader> Marth64: its not about manual, one could easily code something/script to extract values from ebur128 scan and apply it in 2nd pass with volume filter
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<NanananaLeader> because it need to scan all audio from start to the end there is not single line way to accomplish the task
<NanananaLeader> but maybe bounty is really about dynamic mode in loudnorm, need to ask user...
<Compn> the real issue is that film production companies are mixing these films with mumbling whispering dialog and 120db explosions. its insanity
<Compn> and also tv speakers have gotten worse
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<NanananaLeader> arent dialogs mixed into C channel?
<NanananaLeader> aka front center
<nevcairiel> usually yes
<Compn> sometimes
<Compn> but also things are mixed into that area so you cant just blanket increase gain there
<Compn> and better speakers wont help because you're still stuck with 120db noises. but now you're throwing 120db through some powerful speakers and its shaking the windows.
<NanananaLeader> toggle voice activity detector
<Compn> going to a movie theater is like an assault on your ears :(
<Marth64> NanananaLeader: I have such python script but my meaning is that its a redundant task, which it can be handled within filter itself (for example output to file then read from file as options)
<Marth64> ideally
<NanananaLeader> Marth64: yea, even scripting might not be necessary, just pass some options into another pass but within ffmpeg - could be coded like this filtering the encoded streams...
<NanananaLeader> but as ffmpeg works with streams and not only files its problematic
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<Compn> NanananaLeader, next gen ai word detection filter, only adds gain per word
<Compn> now theres a filter we need.
<NanananaLeader> Compn: tried dialoguenhance filter yet?
<Compn> doesnt use ai though
<NanananaLeader> ai is overrated
<NanananaLeader> and fake
<Compn> is dialoguenhance 200x realtime though
<Compn> ehe
<NanananaLeader> its more than than, arround 600x for float input
<NanananaLeader> dunno why you force double floating processing in your usage
<Compn> lots of floats
<jamrial> i paid for 64 bit registers, and i'm going to use all 64 bits
<Compn> send patch for filter in asm ?
<NanananaLeader> why 96? bit registers are not supported ?
<jamrial> x87 has eighty something bits intermediates
<NanananaLeader> Compn: that filter uses fft/rdft and that already have asm SIMD for power 2 size in ffmpeg
<NanananaLeader> long double sinl()
<NanananaLeader> sinl() are part of posix
<NanananaLeader> that means we must add long double sample format to libavutil/avfilter
<NanananaLeader> and add support for such sample formats in *all* audio filters
<NanananaLeader> patch welcome
<NanananaLeader> jamrial: its 88 bits iirc
<NanananaLeader> Compn: why you are unilateral supporter of leader?
<Compn> dont support anyone with unilateral
<Compn> thanks for asking instead of assuming
<Marth64> lets leave people alone and stick to chat about ff tech, its more intellectual and interesting
<NanananaLeader> lets talk about SDR then?
<NanananaLeader> or why not moved to github/gitlab?
<Marth64> yes i think many people want to move
<NanananaLeader> or about other problems, like subtitles support/API etc...
<Marth64> subtitle problems are real, I am working on a powerpoint to suggest improvements
<Marth64> will share with you for opinions
<Marth64> current subtitle support sucks
<NanananaLeader> just make subtitles stream-lined aware
<NanananaLeader> currently all subtitles are read in one go
<Marth64> yes and I want to add filtering finally
<Marth64> also closed captions need work. i am almost done with my improvement proposal. i am curious to your input
<Marth64> for CC
<Marth64> (eia608)
<NanananaLeader> yes, readeia608 should be ultimately V->S filter not what it is now
<Marth64> i will have something material to share by sunday for closed captions (powerpoint 1)
<Marth64> will definitely loop you
<NanananaLeader> what you (gonna)change with CC ?
<NanananaLeader> if there is something very complex or new media type or what not its less likely to over be done/merged
<Marth64> I want to propose we have a CBS at the heart of parsing the byte stream and abstract bespoke logic from various areas
<Marth64> I will send you a draft now actually
<Marth64> for the deck
<NanananaLeader> also if you gonna invent new bitstream inplace of ASS that will be problematic
<Marth64> no not at all, i think that we leave ass alone for now
<NanananaLeader> also, are you aware of bunch of similar subtitle filtering related patches from some guy that are iirc still on github fork?
<Marth64> yes, developer did not address feedback. i asked them if they would continue the work last April and they said yes. but i have seen nothing so I plan to tackle it this year
<Marth64> afk for a bit. have a good day all
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<fflogger> [newticket] root670: Ticket #11409 ([ffmpeg] Using -af and -vf with non-exhistant filter succeeds if input lacks required stream type) created https://trac.ffmpeg.org/ticket/11409
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