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6
3rd Party Plugins - (fb2k) / Re: Playlist-Manager-SMP
Last post by regor -
You can not have 2 playlists with same name. There are multiple warnings via popups when that happens. Also if it's "same name/different type", then its referencing itself, since a reference is matched by name.

I need the file, not a screenshot. Text is larger than the window. You can export the properties at that same panel.
7
Uploads / FLAC (subset) file that ffmpeg (7.0, Windows) cannot decode
Last post by Porcus -
sox -b 24 -c 6 -r 96000 -n whitenoise.wav synth 1 whitenoise
flac whitenoise.wav --channel-map=none -b16384

Code: [Select]
ffmpeg version 7.0-essentials_build-www.gyan.dev Copyright (c) 2000-2024 the FFmpeg developers
  built with gcc 13.2.0 (Rev5, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      59.  8.100 / 59.  8.100
  libavcodec     61.  3.100 / 61.  3.100
  libavformat    61.  1.100 / 61.  1.100
  libavdevice    61.  1.100 / 61.  1.100
  libavfilter    10.  1.100 / 10.  1.100
  libswscale      8.  1.100 /  8.  1.100
  libswresample   5.  1.100 /  5.  1.100
  libpostproc    58.  1.100 / 58.  1.100
Input #0, flac, from 'whitenoise.flac':
  Duration: 00:00:01.00, bitrate: 13891 kb/s
  Stream #0:0: Audio: flac, 96000 Hz, 5.1, s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[aost#0:0/pcm_s16le @ 00000209af0e2c80] No filtered frames for output stream, trying to initialize anyway.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf61.1.100
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, 5.1, s16, 9216 kb/s
      Metadata:
        encoder         : Lavc61.3.100 pcm_s16le
[out#0/wav @ 00000209af0e8e40] video:0KiB audio:0KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: unknown
[out#0/wav @ 00000209af0e8e40] Output file is empty, nothing was encoded(check -ss / -t / -frames parameters if used)
size=       0KiB time=N/A bitrate=N/A speed=N/A
9
General Audio / Re: Can lossy high resolution outperform lossless at standard resolution?
Last post by .halverhahn -
Quote from: guruboolez on
If I summarize:
At the moment, the possible formats of choice are:
  • FLAC: resampled to 48000 and 20 bit = 1000 kbps. Not bad. A few improvements over Red Book (+4000 Hz, +4bit) and decent bitrate. No fear of artifact or noise
  • FLAC: resampled to 54000 and 19 bit = 1000 kbps. Not bad either (+10.000Hz and +3bit over RedBook).
  • FLAC: resampled to 54000 and 18 bit = 900 kbps. (+10.000Hz and +2bit over RedBook).
Slight misscalculation in your table: 54kHz Samplingrate = 27kHz bandwidth = +5000 Hz over RedBook

I woulnd't use odd non standard samplingrates. There is a quite big chance that your sound-hardware isn't supporting it.
To test it for support, set foobar output to exclusive mode.
To playback odd rates additional resampling is required by the player or OS.

Quote from: guruboolez
I tried then 60000 Hz, 58000 Hz, 56000 Hz, 52000 Hz => output file is always 192.000 Khz. Is it a bug or a feature? Probably a bug.
By default Foobar is set to Windows "Primary Sound Driver", the "Primary Sound Driver" is resampling it to the standard samplingrate you're set up in your Windows sound settings.

Quote
  • Keeping 192.000 Hz is a total waste of space: no information but noise on all recordings I tried (and most of them are modern and state-of-the-art recordings from the last three years
  • 96.000 Hz is from an perceptual point of view the same waste, but from a technical point of view the upper frequencies contains real information (but of course not audible)
As you paid a Premium for your 24/192 Files, I would go to 24/96 or 20/96 to save space and to be still in the area of peace of mind.  ;)