Marth64 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 7.1 is released
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<jokoon>
I have this filter: -filter_complex "[0:v]setpts=1.25*PTS[v];[0:a]atempo=0.8[a]" -map "[v]" -map "[a]"
<jokoon>
I want to use that filter, but I don't want it to adapt and keep the same pitch
<jokoon>
I want something similar to audacity's "change pitch AND tempo"
<jokoon>
"[0:v]setpts=0.8*PTS[v];[0:a]asetrate=38400[a]" if the input is 48000Hz, this doesn't seem to work well
<vlt>
jokoon: Maybe the rubberband filer can help here.
<vlt>
*filter
<jokoon>
vvlt thanks
<jokoon>
vlt -filter_complex "[0:v]setpts=1.25*PTS[v];[0:a]rubberband=pitch=0.8, rubberband=tempo=0.8[a]" this seems about correct, but maybe a bit imprecise
<vlt>
jokoon: Is there a reason you run two rubberband instances in sequence instead of letting one of them do the job?
<jokoon>
vlt, because using just one did not do the job
<vlt>
jokoon: Seems weird but maybe :D
<jokoon>
Like I said, I want want to change pitch AND tempo, so I want the audio to be slower AND to have less pitch
<jokoon>
which is what audacity change pitch and tempo does
<jokoon>
yes it's weird, I know
<jokoon>
basically, it's just stretching the audio data without changing the signal
<jokoon>
tried to use audacity instead and import the video with avidemux, not synced
<jokoon>
I am bad at this
<vlt>
jokoon: Does "rubberband=pitch=0.8:tempo=0.8" do something other than you need?
<jokoon>
don't you mean "atempo"?
<jokoon>
"[0:v]setpts=1.25*PTS[v];[0:a]atempo=0.8,rubberband=pitch=0.8[a]" like this then
<jokoon>
apparently yes it's better, although it seems slightly out of sync at the end, but Im not sure
<jokoon>
I can't share the test file I guess
<jokoon>
No sorry the source file is the problem here
<jokoon>
so thanks, that works
<vlt>
jokoon: No, rubberband’s parameter name is – as you have demonstrated in the second instance you used – "tempo". "atempo" is an entirely different filter (that might do exactly what is needed in your case).
<jokoon>
ok but not with a complex filter then
<vlt>
jokoon: Why not?
<jokoon>
vlt, [AVFilterGraph @ 000002149d129cc0] No such filter: 'tempo'
<vlt>
jokoon: The name of the filter is "atempo".
<vlt>
jokoon: "tempo" is one of rubberband’s parameters.
<jokoon>
oooh yes, I thought that ":" was a typo
<jokoon>
sorry
<jokoon>
yeah that seems better indeed
<furq>
rubberband seems like overkill for that
<furq>
asetrate=38400,aresample=osr=48000 is much faster and sounds identical from the five seconds of testing i just did
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<jokoon>
thanks for the suggestion
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<EmberCrest>
hey this is a little off-topic -- hopefully not too much though -- I have a WebRTC stream on my website, which works GREAT! But most devices don't support re-streaming/casting WebRTC. Even VLC can't use WHEP as input.
<EmberCrest>
Looking to find apps/methods (that aren't "screen-mirroring"/Miracast) which will allow me & others to do this -- or point me to a chatroom/forum which I can pose this to..
<BtbN>
You could open it in a browser source in OBS maybe?
<EmberCrest>
also thanks for the other day, the latest ffmpegs with the newer x265 versions are humming along beautifully without segfaults :-) excited for the release into mainline.
<EmberCrest>
BtbN: oh that's not a bad thought! what kinds of casting does OBS support?
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<EmberCrest>
ah.. the browser source option omits the video element. won't do. well, maybe I'll go look for WebRTC forums and bug them. thx again.
<noobaroo>
I have a youtube dancing compilation that is only a few mins in total length, I want it to play it in a loop during 20 min of music
<noobaroo>
thus far i have used concat: filter to make like 45min of the video, and then I combined it with audio and used the -shortest option so it automatically cuts the video to the audio's length
<noobaroo>
This is really inefficient though, i cant imagine if it was a 4k video (its not) like 5min becomes 10x the file size ? How can i loop the same tiny file without using concat+streamcopy to duplicate the data again and again and blowing up file size to huge proportions ?
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<andrewrk>
hello friends
<andrewrk>
when calling av_buffersink_get_frame, is there a way to find out the corresponding frame that was added with av_buffersrc_add_frame?
<andrewrk>
like can you assign userdata to the input and get it back in the output
<andrewrk>
is that what the void *opaque field of AVFrame is for?
<andrewrk>
ok yeah that seems to do exactly what I want. thanks everyone
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<theprof86>
Hi, what is the difference between hev1 and hvc1 for HEVC 265? All my videos are in hev1 and they do not play on iPhones (or any apple device actually). They only play if I tag them hvc1. Which is better? It seems HEV1 is the way to go and modern meanwhile hvc1 is proprietary and a apple thing.
<BtbN>
Just apple being special and calling it something else than everyone else, and refusing to play what the rest of the world calls it.
<theprof86>
I agree.....will they eventually fold and go with hev1? It seems nobody cares about hvc1 other than apple....will they be forced to go the hev1 route
<BtbN>
no
<BtbN>
hev1 has been written into the standard, but I don't see Apple caring
<BtbN>
Most non-apple things happily play both, since it's literally just two different bytes in the header. But Apple makes no signs of caring about the standard.
<JEEB>
hvc1 I think says "no parameter sets outside of the codec initialization data"
<JEEB>
while hev1 allows it to be in packets, too
<JEEB>
with AVC apple allowed both options in their parser, but with HEVC they suddenly became really strict regarding only wanting to support out-of-band parameter sets
<JEEB>
funny enough if you look at the git history of FFmpeg at one point hvc1 was the default for HEVC in mp4. then it got changed.
<theprof86>
i see
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