[FFmpeg-user] VBR encoding not supported ???

[Aditya Dandavate]

Also, according to my latest tests, HE-AAC v2 also supports VBR encoding
with `-vbr 5` as well.

Also, till now, I used 96khz as sampling rate for vbr 5. I did many encodes
with these settings, but it was all flawless!

I guess, these bugs related to VBR and other things that you mentioned are
all fixed, but I guess the Fraunhofer people forget to remove the warning
it seems.


On Sun, 28 Apr, 2024, 9:55 pm Ferdi Scholten, <ferdi at sttc-nlp.nl> wrote:

> On 28-04-2024 12:44, Aditya Dandavate wrote:
> > I was converting flac audio to aac_lc with libfdk_aac with VBR encoding.
> > But I saw a message about VBR encoding.
> >
> > The command and output of the command is as follows :
> >
> > Command :
> >
> > ffmpeg.exe -i input.flac -c:a libfdk_aac -afterburner 1 -frame_length
> 1024
> > -cutoff 0 -vbr 5 -profile:a aac_low output.m4a°
> >
> > Output :
> >
> > F:\DSDA-Doom>ffmpeg.exe -i input.flac -c:a libfdk_aac -afterburner 1
> > -cutoff 0 -frame_length 1024 -profile:a aac_low -vbr 5 output.m4a
> > ffmpeg version N-115010-ga2cdb2f5be Copyright (c) 2000-2024 the FFmpeg
> > developers
> >    built with gcc 13.2.0 (Rev6, Built by MSYS2 project)
> >    configuration:  --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache
> > g++' --ld='ccache g++' --extra-cxxflags=-fpermissive
> > --extra-cflags=-Wno-int-conversion --disable-autodetect
> > --enable-libopus --enable-libvorbis --enable-libx264 --enable-libx265
> > --enable-libaom --disable-debug --enable-libfdk-aac --enable-schannel
> > --enable-gpl --enable-nonfree
> >    libavutil      59. 16.101 / 59. 16.101
> >    libavcodec     61.  5.103 / 61.  5.103
> >    libavformat    61.  3.103 / 61.  3.103
> >    libavdevice    61.  2.100 / 61.  2.100
> >    libavfilter    10.  2.101 / 10.  2.101
> >    libswscale      8.  2.100 /  8.  2.100
> >    libswresample   5.  2.100 /  5.  2.100
> >    libpostproc    58.  2.100 / 58.  2.100
> > Input #0, flac, from 'input.flac':
> >    Metadata:
> >      encoder         : Lavf61.3.103
> >    Duration: 02:05:52.13, start: 0.000000, bitrate: 621 kb/s
> >    Stream #0:0: Audio: flac, 48000 Hz, stereo, s16
> > Stream mapping:
> >    Stream #0:0 -> #0:0 (flac (native) -> aac (libfdk_aac))
> > Press [q] to stop, [?] for help
> > [libfdk_aac @ 000001b9623d7400] *Note, the VBR setting is unsupported
> > and only works with some parameter combinations*
> > Output #0, ipod, to 'output.m4a':
> >    Metadata:
> >      encoder         : Lavf61.3.103
> >    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo,
> s16
> >        Metadata:
> >          encoder         : Lavc61.5.103 libfdk_aac
> > [out#0/ipod @ 000001b962402280] video:0KiB audio:229238KiB
> > subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead:
> > 0.605099%
> > size=  230625KiB time=02:05:52.13 bitrate= 250.2kbits/s speed=18.1x
> >
> > F:\DSDA-Doom>pause
> > Press any key to continue . . .
> >
> >
> > So, may I know what's wrong with VBR to the point its unsupported ? Is
> > it going to be removed in the future from the encoder ? Is it
> > considered experimental or stable ?
> >
> >
> > Please tell me about this 🙏
> >
> >
> > Thanks
> >
> Well, the warning is not from ffmpeg but from libfdk_aac and this is the
> reason:
>
>      "As of FDK version 3.4.12, not all combinations of audio object
> types, bitrate modes, channel layouts, and sample rates can be used
> together, due to a limited table of pre-computed values used by the
> encoder. For example, using 96kHz stereo input with the AAC-LC audio
> object type and bitrate mode 5 (VBR 96-112kbps/channel) will result in
> catastrophic failure"
>
> There has been a patch to (partly) resolve this, but it still has
> limits, hence the warning.
>
> Also, VBR 1-4 can only be used with specific (low) sampling rates, only
> 5 can be used for samplerates >15 kHz
>
> Fraunhofer does not recommend using VBR with this codec.
>
> Generally, if your device supports the Opus codec, use that, it has
> (much) better quality at the same bitrate as AAC, or much smaller files
> with the same quality as AAC
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