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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<KillerWasp> don't exist AV_SAMPLE_FMT_U8P?
<johnjaye> i'll have to come back to this in a bit. for some reason the zoompan is resetting itself after a few seconds but i can't see why exactly
<johnjaye> is the z expression in the zoompan filter evaluated on every frame?
<KillerWasp> Specified sample format u8p is invalid or not supported
<KillerWasp> like AAC don't support in 8 bits.
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<KillerWasp> i try set ->frame_size=44100 for AVCodecContext, but avcodec_open2 change it to 1024, then instead of change 1 second of audio, i only can change 43.06640625 times in 1 second, which mean i must send each 0.0232199546485260770975056689342~~~~ second. Very imprecise. I must use av_interleaved_write_frame in a mandatory pts/dts juggle?
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<nasso> hello! i'm trying to use `hwupload` with the lavfi input virtual device. however, command-line options like `-init_hw_device` and `-filter_hw_device` don't seem to apply when using it? only when using `-vf`?
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<nasso> actually let's get straight to the point: how can i know if ffmpeg is doing the equivalent of a `hwdownload` between my filter graph and my encoder?
<nasso> i have a hardware encoder (h264_nvenc) and i also use hardware accelerated filters, and i'd like to make sure ffmpeg is keeping the frames in hardware between the two
<nasso> `-dumpgraph` doesn't seem to show my `-vf` graph, and the debug logs don't seem to mention any `auto_scale`. however, i also don't see an auto_scale when i use libx264, so i guess im just not looking for the right thing?
<welder> I checked the pipewire audio recording patch that has been submitted last year. The reviewer suggested to implement it as lavfi and not as device (-f pipewire), because apparently the second option is a hack, even thought pulseaudio is recorded this way. What's wrong with the approach to implement it as -f flag?
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<kepstin> lavfi (libavfilter) doesn't make sense; sure they didn't mean libavformat?
<furq> i assume because pipewire can carry video as well as audio
<kepstin> hmm. requiring that folks use libavfilter for something that they expect to be a device source sounds like it would be quite confusing.
<kepstin> I mean, i suppose "-f lavfi" works, but still :/
<furq> there are also already devices that carry video and audio
<furq> but i don't know anything about lavd so maybe it really sucks
<JEEB> kepstin: avfilter works without inputs ;) (just `-filter_complex STUFF`, no need for `-f lavfi -i INPUT`)
<kepstin> sure, but -f lavfi makes it easier to drop into existing workflows that already have other filters.
<JEEB> also the reason why capture sources are starting to not be devices is because avdevice/avformat limitations
<furq> is this to do with outputting hardware frames or more general limitations
<JEEB> hardware frames is one thing, but in general the API is not meant for receiving frames
<johnjaye> can ffmpeg refer to the end time of the video?
<johnjaye> i want to fade out 7 seconds before the video end. is there a symbolic way to do it?
<johnjaye> or do i need absolute time stamps
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<kepstin> johnjaye: ffmpeg filters do not have access to information about the length of the stream they're processing, so if you're doing a fade-out via filter, you'll have to provide absolute time stamps.
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<noobaroo> how can I remove empty timestamps in middle of video?
<kepstin> noobaroo: you'll have to better explain what you mean by that.
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<Dan97> I'm trying to segment a wav file exactly at specific timestamps or frames but when I check the result with ffprobe the files are close but not exactly what I want
<Dan97> fmpeg -i input.wav -c copy -f segment -segment_times 1000ms,10000ms output%d.wav
<Dan97> ffmpeg -i input.wav -c copy -f segment -segment_frames 50,500 output%d.wav
<Dan97> shouldn't it give me a first segment with exactly 50 audio frames and 1 second?
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<sonicrules1234> JEEB: For the results of the lossless compression I was doing the other day, for file size: libaom-av1 < x264 < svt-av1 < ffv1 . The clip was 30 seconds with very high movement, changes
<sonicrules1234> *movement/changes
<noobaroo> kepstin When i save livestreams with yt-dlp then often times there is lots of small several second periods of timestamps with no video or audio, and also when i network stream with mpv and then use dump-cache ~/video.mkv then the entire timestamps will be empty except for the areas that i watched.
<sonicrules1234> Also, the clip was from a blu-ray
<noobaroo> I keep switching between various mpv --video-sync options like display-sync, adrop, and now most recently desync, which is supposed to play at constant rate according to system clock and only meant for testing. But in mpv docs it says the entire flag of --display-sync (and all of its possible values) doesn't work on VFR for typical containers (including mkv)
<sonicrules1234> noobaroo: I've had problems with mpv for youtube livestreams too recently. I end up having to use streamlink instead of yt-dlp
<JEEB> sonicrules1234: and that was with similar parameters? GOP size etc
<sonicrules1234> Yeah, gop_sizes were all defaults
<JEEB> the defaults are not the same :D
<sonicrules1234> > .>
<sonicrules1234> Any other parameters I should keep the same?
<JEEB> x264 defaults to 250 frames, was it?
<JEEB> (that's the max GOP duration of course, if the encoder thinks it's done a long enough GOP and that this would be a Good Spot then it will of course finish the GOP earlier
<furq> Dan97: the first one should work, but -segment_frames refers to video frames
<furq> maybe try the asegment filter
<JEEB> it's kinda hard to figure these things out, which is why back when I did encoder testing I just attempted to max things out. although even there I utilized the fact that one encoder had literally a standardized config file that was utilized by the format developers, and others had presets
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<furq> how many orders of magnitude faster was x264 than aom
<Dan97> furq: thanks. I can't find info about asegment filter in the docs
<sonicrules1234> I didn't time them :P
<Dan97> thanks
<sonicrules1234> But a lot longer
<sonicrules1234> aom's output was 15 megabytes smaller :P
<furq> what's that as a percentage
<Dan97> furq: what is the asegment filter syntax? This gives me an error
<Dan97> ffmpeg -i input.wav -c copy -f asegment=timestamps="60|150" output%d.wav
<furq> it's much more annoying than that
<Dan97> Requested output format 'asegment=timestamps=60|150' is not a suitable output format
<sonicrules1234> About 5% smaller
<furq> ffmpeg -i input.wav -c copy -lavfi asegment=timestamps="60|150[x][y][z]" -map "[x]" out01.wav -map "[y]" out02.wav -map "[z]" out03.wav
<furq> except without -c copy
<furq> also you probably want timestamps=1|10
<Dan97> 1|10 was just a test.
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<Dan97> Thanks your command with -c copy seems to work and produce accurate times
<Dan97> -lavfi is a dummy device? why do I need it?
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<JEEB> it's not a dummy, it's a "device" that calls libavfilter internally
<JEEB> ahhh, actual `-lavfi`
<JEEB> that is alias for filter complex
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<JEEB> a filter chain that is not limited to a single output basically
<Dan97> thanks. verified that -filter_complex gives me the same result
<JEEB> not sure if either of those names is slanted for removal, so just feel free to utilize the shorter one
<noobaroo> sonicrules1234 I'm also using streamlink since about 1month ago
<Dan97> JEEB: removal and replaced with something else?
<JEEB> I'm saying that I don't know of either being changed or removed
<JEEB> so just feel free to utilize the shorter
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<Dan97> thanks
<johnjaye> something i've been wondering for awhile
<johnjaye> if i didn't specify audio in the filtergraph how do i get it back
<johnjaye> when I try -map "[0:a]" I get errors about the filtergraph not having it or something
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<johnjaye> example: https://dpaste.com/B4MD74S5J
<johnjaye> i tried changing [0:v] to [0] but that didn't get the audio in either
<JEEB> [thing] are definitions from within the filter graph when utilized in maps
<JEEB> so if you actually just want stuff from input, you do a normal map
<JEEB> without the []
<johnjaye> oh thanks that works. i had ome up with a complicated way of putting [1:a] into the filtergraph and splitting it and then sending one output to the anullsink
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<fairway> i understand the point of higer resolution 4k vs 1080p vs 480p, what is the point of using 60fps vs 30fps
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<sonicrules1234> fairway: Can you check your monitor's settings? Do you have a refresh rate of 120hz or higher available?
<fairway> no, i am using 75hz
<sonicrules1234> Ok, switch to 60hz or 30hz. Then move your mouse around and switch back. Move your mouse around again
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<sonicrules1234> There's a noticeable difference
<fairway> there is no such thing as 30hz
<fairway> only 60hz or 75hz
<sonicrules1234> Try 60hz then
<sonicrules1234> It won't be as noticable
<fairway> and no, i don't see the difference between 60hz an d75hz
<sonicrules1234> If you had something like 120hz available, you would see the difference
<sonicrules1234> It's way smoother
<fairway> sonicrules1234 then how come you don't see 120 fps videos
<fairway> you will not find a video that is using 120 fps
<sonicrules1234> I say 120hz because the difference is way more noticable
<fairway> answer my question
<sonicrules1234> 120hz would double the size of the video?
<sonicrules1234> 120fps*
<fairway> by size you mean file-size?
<sonicrules1234> Yes
<fairway> yes it would, it would actuallyy quaduple the size of 30fps video
<fairway> assumingg such video exist
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<another|> not exactly double. inter compression can shave some of that off
<fairway> another| 30fps ---> 120fps that's more than double
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<another|> oh you mean from 30? well, depends on the content and the codec
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<fairway> another| do you see the point of using 60fps instead of 30fps
<another|> yes of course
<fairway> then how come we don't see 120 fps videos
<another|> diminishing returns
<fairway> another| so do you think there is a number (fps) where you say, it's enough
<fairway> there is not point of higher fps
<fairway> not = no
<another|> sure.
<fairway> okay what is the number
<another|> depends on the content
<fairway> let's say sports
<fairway> let' say content = sports
<another|> maybe 120 ?
<fairway> okay fair enough
<fairway> another| what about 96khz vs 48khz audio
<another|> what about it?
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<fairway> do you think 96hz is pointless
<another|> there use cases for both
<fairway> another| what about people that says human's eye cannot see more than 30 fps
<fairway> or human ears cannot hear more than 48khz
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<another|> whoever says that about eyes has no idea what they're talking about
<another|> re: ears: that's just plain wrong