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2
3rd Party Plugins - (fb2k) / Re: foo_convolve - Impulse response convolver
Last post by sveakul -
I assume you mean foo_convolve not "foo_convolver."  What do you mean by "won't install correctly?"  What do you do, and then what happens?

It is 32-bit only so requires 32-bit Foobar;  two other choices that can use 32 or 64 bit are foo_dsp_stereoconv and foo_dsp_convolver.

BTW, I just installed foo_convolve 0.3.3 on Foobar 1.6.18 and everything went fine:

https://www.foobar2000.org/getcomponent/efd071729d99e8c14cae0368b8a8ecaf/Convolver.fb2k-component
3
Vinyl / Re: What's the ideal way to fix the pitch of a vinyl recording?
Last post by fooball -
I still say resample in Audacity (or whatever audio editor you like), it will be simpler in the long run.  Keeping to a standard sample rate will be compatible with more players, and the resampling will not be audible.  There are far greater distortions in the extraction of audio from vinyl.

What you might not realise is that unless you take special measures, the player will resample to its default sample rate anyway, that's how multiple audio sources can be mixed together to get (say) a system alert with a music source and/or a video track simultaneously.  Resampling is all over the place whether you like it or not.
4
3rd Party Plugins - (fb2k) / Re: foo_convolve - Impulse response convolver
Last post by orphictrench -
Hey guys/gals!

I became a great fan of foo_convolver over the years and I go to great lengths to create the best possible filters with Acourate software. I don't see myself using another librarian, but recently I changed computers and the convolver won't install correctly.

I tried downgrading to 32-bit foobar2000 and it doesn't work either. Could anyone help? What's going on? What am I doing wrong?

I simply cannot live without this extra component.
5
AAC - Tech / Re: AAC frame lengths
Last post by includemeout -
Quote from: Klymins on
Quote from: includemeout on
Dun, dun, dun!

And that based on which psychoacoustic model??

Sorry, I couldn't understand you, can you explain a little more @includemeout ?
What I meant is that, as you know, MDCT/sub band-based lossy codecs are wild beasts built from the ground up around specific psychoacoustic models.

In fact, this community abounds with cases where even tweaking their settings excessively has been proven to wreak havoc on audio quality and end up causing more problems than the initial ones they were meant to solve - and I'm as guilty as the next guy for having tried a couple dozens of them - ugh! including -r3mix for MP3 and -mnt for customizing MPC's noise shaping back in 2001/02 and such and such.

So, I'm not trying to be disrespectful and doubt your programming skills and what-not, but what makes you think such changes won't be, for starters, an uphill task and that they will end up improving a codec as established as AAC?

6
3rd Party Plugins - (fb2k) / Re: Playlist-Manager-SMP
Last post by regor -
It's a SMP bug with specific angles on rounded corners, which I simply handle as popups. It's funny I have had them many times and no one noticed. Just ignore it or just change the panel size if that happens. I will probably disable it for regular users.
8
3rd Party Plugins - (fb2k) / Re: Biography Discussion
Last post by marc2k3 -
Bug in helpers.js: success / sucess

Code: [Select]
	save(fn, text, bom) {
		try {
			const success = utils.WriteTextFile(fn, text, bom);
			if (!sucess && fn.length >= 256) {
				fb.ShowPopupMessage('Script is trying to save a file in a path containing more than 256 chars which leads to problems on Windows systems.\n\nPath:\n' + fn + '\n\nTo avoid this problem, install your foobar portable installation at another path (with less nesting) or change the the cache folders at \'biography.cfg\'.')
			}
		} catch (e) {
			this.trace('error saving: ' + fn);
		}
	}

10
Vinyl / Re: What's the ideal way to fix the pitch of a vinyl recording?
Last post by drumliner -
First a big thanks to everyone for responding, now on to your replies...

Quote from: Glenn Gundlach on
I believe you can do this in Audacity. I KNOW you can do this in Adobe Audition. The result is exactly the same as turning the pitch control on the turntable. BTW, what is "lossless vinyl"?

Many audio editors will let you do it and I am asking about the best way to go about this in order to maintain audio quality as much as possible, ie not unnecessarily introduce anything unwanted in the process. As for your question I said "lossless vinyl recordings", meaning recordings of vinyl stored in a lossless format to point out these aren't lossy files editing of which would lead to loss of quality anyway, making my question about retaining quality largely moot. Apologies for the confusion.


Quote from: fooball on
If you're saying the original vinyl was spinning at the wrong rate...

Yes that's what I meant. I'm familiar with the audio editing processes and I was trying to avoid confusion with my choice of words, but I guess I wasn't all that successful at that. I'm used to "time stretching"  as the process of changing tempo without altering the pitch and "pitch shifting" as the process of changing pitch without altering the tempo and I know these two will mess with audio quality, although they have certainly progressed a fair bit compared to the early days. In any case I'm not trying to do either of these, all I need is to simply slow down the recordings to avoid re-rerecording the vinyls at the right speed (I determined all the problematic vinyls have been recorded at +8%). I used "pitch" in the thread's title because that's what the control for changing the speed at which the vinyl is spinning is usually labeled on turntables like the famous Technics and I thought people would be most familiar with this terminology when talking about vinyl. Again sorry for the confusion.

Quote from: magicgoose on
to be really sure you're doing this the right way, better stick to using only resampling.

it requires a step where you change the sample rate without resampling (to actually change the speed), this is rarely implemented but when it's in WAV/PCM format, it has a fairly simple structure so you can edit the sample rate number in the header with a hex editor.

resampling is even optional, if you don't care that the resulting sample rate will be something wildly non-standard. Most players will resample on the fly. And it's the most lossless method, if all you do is modify one number in the file header without even changing the waveform data.

so, for example: let's say you have a record in 44.1 kHz and you need to make it play twice as fast (and therefore also an octave higher). Then you can edit the sample rate info in the header to be 88.2 kHz, and that's it.

This is exactly what I'm trying to do. The recording was done at 96khz with the "pitch" setting on the turntable at +8%, so I need to slow down by roughly 7.41%, which if my calculations are still correct, would be the same as setting the sample rate to 88.886hz (no resampling as you say). My audio editor allows me set the sample rate directly (has a "set sample rate only, no resampling" option), so I don't even have to hex edit the header manually. But I do think I'd rather have a standard sample rate, so I guess I can then resample from 88.886hz to the standard 44.1khz. Would this be fine or am I messing up somewhere?