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Recent Posts
1
General Audio / Re: Vorbis better than opus?
Last post by sld -
How many times have you been pointed at this forum's terms of service?

You have not "actually tested both" for being any "better".

I have to agree with you then and withdraw my outrageously wrong statements.
It's kind of sad for us forum lurkers to see yet another one succumb to confirmation bias instead of performing some really basic scientifically valid audio comparison testing.
2
General Audio / Re: Good requantizers?
Last post by saratoga -
Do you want PCM requantization or compression?  If the former, Sox is a good choice, but there are many other options too.  If you want compression, FLAC or MP3 have better compression than DPCM. 
I want requantization, not compression. Especially reduction to 8-14 bits.

I know about sox but it mostly does dithering (with noise shaping), I'm interested in other methods with as little noise as possible (or at least having adaptive requantization and noise shaping like lossyWAV, but I'm unable to output 8 bit PCM with it).
DPCM sounds very impressive but it decodes to 16bit PCM and after decoding I see more bits in use than selected.

The reason encoding to DPCM sounds good is that it is a form of lossy compression.  If you don't want compression, then your options are much more limited since there is a fixed amount of noise that must be present in an uncompressed bitstream of a given size.  You can use things like noise shaping to move that noise around (usually to higher frequencies where it is less noticeable), but you can't get rid of it without either introducing compression or increasing the bitrate. 
3
3rd Party Plugins - (fb2k) / Re: foo_upnp
Last post by typecrazy789 -
I realize support for this plug-in is non-existent now but I'm wondering if anybody has the answer to this one...I'm using the Logitech Media Server UPnP/DLNA bridge to connect it to Foobar2k using foo_upnp, and it works flawlessly to crossfade until I get a short track less than 10 seconds or so, and it won't crossfade. The log on the LMS side normally shows gapless but with a very short length it returns "gapped" instead, which I presume is triggered by foo_upnp. Is there some buffer setting that might cause this behavior? The same tracks played directly in Foobar2k work perfectly, so it definitely seems to be a bridge issue.
4
3rd Party Plugins - (fb2k) / Re: foo_youtube
Last post by meemosafaji -
meemosafaji, playback should not stop.A warning window should pop up(which can be disabled in the advanced settings)and switch to the next track while continuing to play.It is advisable to use the latest versions of ALL components involved in the process of playing videos from YouTube, and it is desirable to get an API key.
i didnt say that my playback stops, just my game because the warning pops out the game but now that u said that i can turn off the warning on advanced settings i get it thx, i already had the newer version and api key
5
FLAC / Re: New FLAC compression improvement
Last post by Wombat -
You can use my old compile. If you want to use the encoder from the recent CUEtools download you need to copy the additional file Newtonsoft.Json.dll
Not to sure about padding. With foobars Optimize file layout + minimize i get this for a single album:
Total file size decreased by 29953 bytes for the CUEtools -8 version and Total file size decreased by 62721 bytes for the flac-native -9 version. I doubt this is much.
7
Support - (fb2k) / txtp files from mario kart live: home circuit (nintendo switch) won't play
Last post by ARV -
hi, this is my first time posting here because i have an issue regarding txtp files in foobar2000. i have downloaded an audio dump of mario kart live: home circuit for nintendo switch from hcs64 and when i try to play only some txtp files in foobar i get an error and idk why. i mainly use vgmstream as a plugin for video game music but idk why this happens since it has worked for me before
8
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by Porcus -
Perhaps SoX was born in the era that CPUs' built-in FPUs were slow or didn't even had built-in FPUs?
If I understand correctly, the ability to handle 32-bits at all was introduced much later, and if so, "born" is not an excuse.

Also a bit off-topic, those so-called 32-bit DACs nowadays only use 24-bit filter coefficients, people are being scammed! :))

32-bit means:
 * is able to receive and immediately trash the last eight bits
 * will naturally dither away a few more in the conversion (measurements from Archimago)
 * so in the end you can take pleasure in the sound of your lightbulb while you are playing at pain threshold ... and not to mention, revel in the placebo effect over the eleven or twelve more bits that are printed on the outside of the device (and take up most of the hard drive space)

OH(DCD) JOY!
9
General Audio / Does removing padding remove data or does it just remove what’s not required
Last post by stevehero -
If say I use metaflac or ID3 to remove the padding for FLAC and MP3s'.

ID3.exe command:
Code: [Select]
ID3 -2 -s 0 "file"

metaflac.exe  command:
Code: [Select]
metaflac --preserve-modtime --dont-use-padding --remove --block-type=PADDING "file"

If I use the commands for both to remove padding, does it affect any existing tags (remove them) or does it just remove what's not needed with regards to padding?
10
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by bennetng -
Gaaaah ... someone had a bright idea back in the day then.

(sox is thirty years old. I used it quite a lot in my Linux-on-desktop days, so, mildly disappointed over an old acquaintance being that stupid. Allegedly it supports 32-bit integer and 64-bit float ....)
Perhaps SoX was born in the era that CPUs' built-in FPUs were slow or didn't even had built-in FPUs?

Anyway... Reaper supports lossless import and export up to 64-bit float, Audacity is lossy beyond 32-bit float but able to handle int32 and float64 files in lossy ways like foobar. Audition is also lossy beyond 32-bit float, don't know about the latest Adobe CC versions, but at least it is true in the older versions I have. For these float32 DAWs exporting to int32 is actually lossy! Only special DAWs like the old (early 2000s), fixed-point hardware-based Pro Tools TDM used integer as internal format: 24-bit for external link and 48-bit processing precision with 8 guard bits for clipping prevention, so 56 fixed bits for a measly 24-bit data chain (source). The newer Pro Tools HDX uses 32/64-bit float already (source). So for those who strive for the "industry standard" in this era, 64-bit float is the way to go, even WavPack doesn't support it. Also a bit off-topic, those so-called 32-bit DACs nowadays only use 24-bit filter coefficients, people are being scammed! :))
X

For foobar, I care more about plugin compatibility, what if kode54 don't update all of his game music plugins? I lost foo_midi already, can't afford more loss.
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