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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<Alex1f341>
Good day peeps, could someone guide me through this please?
<Alex1f341>
I'm trying to create a test sample with the ffmpeg from the commandline. I aim to create a 10-bit video sample with a specific color on each frame. The problem is that the `color` filter on libav only supports colors in 0xXXXXXX hex format (24 bits), but 10-bit color needs more bits than that. Is there an alternative way to represent colors with this filter?
<Alex1f341>
Appreciate it :D
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<noobaroo>
I converted a EAC3+Atmos 768kbps 5.1 stream to AAC and also to Opus, because my TV won't play the original stream (I guess because of the atmos part)
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<noobaroo>
But I'm a little weirded out because under Audacity, the opus and the original audio look basically identical, and the aac one looks significantly warped.
<noobaroo>
I don't know how the ffmpeg decoding of eac3+atmos work. But I'm wondering if it's possible if libfdk_aac is somehow encoding the atmos data into the stream, and maybe opus is ignoring it ?
<noobaroo>
That would also mean that audacity ignores the atmos stuff when looking at the original stream though.
<noobaroo>
According to mediainfo: Format : E-AC-3 JOC, Format/Info: Enhanced AC-3 with Joint Object Coding, Commercial name: Dolby Digital Plus with Dolby Atmos, Codec ID : A_EAC3
<noobaroo>
I tried to use bsf:a eac3_core to get rid of the atmos stuff, but didn't work and apparently there is a 6 year old ticket about this with no resolution: https://trac.ffmpeg.org/ticket/7574
<galad>
atmos is totally ignored
<galad>
libfdk_aac might apply a cutoff
<noobaroo>
What do you mean? and thanks for replying :)
<furq>
fdk usually applies a lowpass filter
<noobaroo>
I'm lost but I will Google cutoff and lowpass filter
<furq>
opus does as well but it's almost always at 20khz
<furq>
so it will be a lot less obvious in spectrals
<noobaroo>
Thanks. While you're here do you maybe know why he_aac doesn't work on Kubuntu? I even built it from source from https://sourceforge.net/projects/opencore-amr/ And tried the fbdkaac standalone binary, it says unsupported profile
<furq>
no idea
<furq>
he-aac sounds awful in my experience
<noobaroo>
Oh okay
<noobaroo>
How do I set AACENC_BANDWIDTH value?
<furq>
i believe it's mapped in ffmpeg as -cutoff
<noobaroo>
Thanks
<noobaroo>
I think -cutoff worked, no errors and the filesize is larger, but in Audacity it looks identical (still warped)
<noobaroo>
I also tried vbr 5 and all 3 fdkaac encodes all look the same, all warped
<noobaroo>
I guess I will probably go with opus then.
<noobaroo>
What do you recommend I set cutoff to ?
<furq>
i recommend you don't
<noobaroo>
Okay.. does that mean for opus I should set it to less than 20khz then?
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<furq>
no it means you should just let the codec do whatever it thinks is best
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<noobaroo>
If I encode from dts-hd to something else, is it the same as encoding from bsf dca_core? Is it ignoring the dts-hd part?
<noobaroo>
Audacity does show different stuff for the dts-hd versus the dca_core so it isn't ignoring the hd extensions
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