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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<uprprc777> Hello everyone,
<uprprc777> I’m a a novice FFMPEG user and need some help. I want to convert a FLAC audio file to AAC using the AudioToolbox encoder with CVBR at 320kbps.
<uprprc777> I’ve done some research and read some documents myself, but I’m still unsure if I’ve got it right. Is the following command correct?
<uprprc777> ffmpeg -i input.flac -c:a aac_at -b:a 320k -aac_at_mode output.m4a
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<uprprc777> I found that the previous one was definitely wrong and made some update myself. Is the following new one good?
<uprprc777> ffmpeg -i input.flac -c:v copy -c:a aac_at -b:a 320k -aac_at_mode cvbr output.m4a
<uprprc777> I also want to transfer all tags including the cover art to output aac
<uprprc777> I tried this one on some audio files. The output looks fine though without detailed examination. The audio plays fine and seems all metadata including the cover is transfered to the new file.
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<uprprc777> But I have a huge audio collection and I need convert all of them. Those files may be encoded in some different settings. Therefore I am still worrying that if that would work for all file in my collection.
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<Marth64> By default ffmpeg will carry the metadata of the first input file (in your case input.flac)
<galad> ffmpeg doesn't support all the iTunes-style metadata that can be found in mp4/m4a, but probably it's good enough for your files if your source are flacs
<uprprc777> they are all flac
<uprprc777> the output looks fine on my computer so far but I am going to use them on some music player app on my phone and some dedicated mobile music player where compatibility is usually limited
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<alien_lappy> does anyone know; how can i use multiple -i with audio files and to have each of those start at a specific time and skip some seconds and have a specific duration; in one line to merge it into one audio file?
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<alien_lappy> something like "ffmpeg -i file1 -ss 15 -t 60 -i file2 -ss 20 -t 40 -wait 40 output.mp3" but more files; this case only uses the first input and does nothing with the second file
<alien_lappy> that'd have a merging of 2 audio and would last 80 total
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<alien_lappy> oh, adelay and then amerge, maybe?
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<alien_lappy> i tried this filter, but i'm failing to grasp how it works, i think: "[0:0]atrim=10:88[a];[a]apad=80[a2];[1:0]adelay=90[b];[a2][b]amerge"
<alien_lappy> the adelay doesn't happen, at the very least, but i think something is wrong with outputs back to inputs
<kepstin> maybe you're misunderstanding what the amerge filter does?
<kepstin> if both the audio streams are stereo, for example, amerge combines them into a single 4-channel audio stream.
<alien_lappy> ah, so i need amix instead? or do i need pan
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<alien_lappy> i figured out what adelay didn't work, i need s suffix for seconds
<kepstin> note that adelay works per-channel in audio, you probably need to set the all=1 option
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<kepstin> are there any sections where you want silence - i.e. no input audio file mixed in?
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<kepstin> hmm, the adelay should take care of that by inserting silence, should be fine.
<alien_lappy> kepstin: probably no need for silence
<alien_lappy> need the all=1 though
<kepstin> yeah, using the "amix" filter would make sense here. Note that you will probably need to set normalize=0 to get the desired result.
<kepstin> the other alternative would be to delay only the first audio input, then add silence to the end of each input file corresponding to the gap between them using the apad filter. Then concatenate the result.
<alien_lappy> they overlap often, so can't really concatenate
<alien_lappy> hmm, strange, the amerge seems to work? it does produce a stereo? how odd
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<alien_lappy> kepstin: i got fairly good results with "[0:0]atrim=14.96:86[a];[1:0]atrim=4[b];[b]adelay=delays=66s:all=1[b2];[a][b2]amix=duration=longest:normalize=0[final];[final]volume=0.4" and i see that i can just add more sources and atrim and adelay (amix will need the inputs=num)
<alien_lappy> thx
<alien_lappy> btw: do you know of a way to play something muted (not by volume; but like it's played behind a veil/wall) ? is there something like that?
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<daily> hello everyone :) i've been messing with DVDs again and noticed that my 24/1001 FPS DVD isn't playing correctly in Media Player Classic on Windows XP
<daily> it thinks it's 30/1001 FPS
<daily> is there any way to add flags to a VOB file with ffmpeg, so that it tells the player to apply "inverse telecine"
<daily> hopefully, and i pray, without reencoding, because i no longer have the original videos, only the VOBs made with ffmpeg -target film-dvd
<daily> and yeah i realized that ffmpeg was doing things correctly, the hardware player wouldn't play it because it was VBR
<daily> ffmpeg actually does the job fine, at least as far as that hardware player is concerned...
<daily> but now the issue is with MPC on XP
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<daily> anyone?
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<another|> 24/1001 fps sounds very wrong
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<daily> another|: why?
<daily> oh, oops, i meant 24000/1001
<daily> so like i have a 24000/1001 FPS DVD but it plays at 30000/1001 in MPC on XP because apparently it can only do NTSC or PAL and not "FILM"
<daily> but FILM is better clearly as it saves space
<daily> it seems to work in my DVD player..
<iive> daily, what is the issue you are having. Sorry, I've joined after your initial question.
<daily> iive: i made a DVD with ffmpeg and dvdauthor, but its FPS is 24000/1001 (film-dvd), when i play it in Media Player Classic on Windows XP it plays at 30000/1001 which is too fast
<daily> it is NTSC in every other way (720x480)
<iive> i see...
<daily> iive: i found out there's apparently something called TFF and RFF that one can set to play 24000/1001 as 30000/1001
<daily> but then i also have to fix all the PTS
<daily> as well
<daily> TFF, RFF, and then the framerate would be set to 30000/1001 even though it's really 24000/1001 or something
<iive> yess, top-field-first and repeat-first-field
<iive> since ntsc/pal are interlaced formats and work on half-frames called fields.
<daily> this is all progressive though
<daily> iive: i don't know, i guess it works in my DVD Player, hehe
<iive> the encoder could mark some fields to be repeated on playback, instead of having them hard-coded
<furq> i don't think you would need to modify the timestamps
<daily> are you sure? the apparent framerate would change
<daily> what about audio sync?
<iive> technically, the encoder does encode 24 progressive frames, but some frames are shown for 150% of the regular time.
<daily> it only fails in Media Player Classic on XP
<iive> to be fair, I'm not sure if the ffmpeg encoder uses them on its own.
<furq> sounds like their bug
<daily> furq: yes but does it mean my DVD isn't according to spec?
<furq> if it works on a dvd player then it's fine
<daily> i just tried it
<daily> it does seem to be the correct timing
<daily> i guess there's no need to worry, ffmpeg did fine then
<furq> if you have another soft telecine dvd you could test mpc-hc with that i guess
<furq> it's not that uncommon for film
<daily> it seems like great for saving space if there's no need for 30 FPS
<daily> furq: it's just that i'm not sure this is soft telecine, i think it's just set to 24000/1001 in the MPEG-2
<daily> and maybe some DVD players wouldn't play it right
<daily> but this one that i have does
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<daily> then again this DVD is just for me, so...
<daily> as long as i don't get a different DVD player it should be fine
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<daily> i guess it's probably hard to make actual soft telecine
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<daily> well, thanks everyone :)
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