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Ways to improve 24 khz or lower sound files using audacity/audition by Sephirothkefka at 2:08 AM EDT on April 29, 2018
Is there an easy way to improve music that is noticeably lower quality than normal (like Super paper mario, kid Icarus uprising, etc.) im fine with 32khz files most of the time especially if they're synth in origin (as most synths sample at that rate anyway). I haven't found sn easy way to do so.
by kode54 at 3:14 AM EDT on April 29, 2018
Not really possible. What was lost stays lost. The only feasible way is to remake all the music and target a higher sample rate.
by simonmkwii at 8:56 AM EDT on April 29, 2018
@kode54 - I disagree.
You can even make 16KHz 4-bit ADPCM recordings sound like a CD-quality track.
For example, you can use an exciter, which interpolates the low-to-mid range frequencies into the high range spectrum, or simply resample it with no interpolation, and manually filter out the aliased frequency bands.
by MoldyPond at 10:30 AM EDT on April 29, 2018
But that involves processing and alteration, which leads to the argument of "If it's coming from a low-quality source, is it really HQ?".

Personally I believe kode54 is right. As someone who obesses over having as highest quality as possible it really bugs me to side with kode, but I'd rather have the original low-quality version rather than knowing it's been processed (because who's to say their version is 100% accurate?)
by simonmkwii at 10:36 AM EDT on April 29, 2018
Considering digital audio isn't 100% accurate anyway, it's kinda a moot point to be honest.
That's not just me defending my objectively brilliant remasters at all, nope, not at all 😁
by MoldyPond at 12:35 PM EDT on April 29, 2018

Honestly the only part that bugs me about remastering is when it is labeled as such with no mention of it being a fan-made thing.
by nothingtosay at 2:58 AM EDT on April 30, 2018
simonmkwii is actually hitting on something that's not as widely known or understood as it should be. Game consoles resample audio for mixing and apply interpolation that generates high frequencies through aliasing.

A great demonstration of this is with a game like Goldeneye 007, where the USF files play at about 22 kHz, yet if you play the game on a real Nintendo 64 it sounds better and has frequencies well into the ultrasonic range. It doesn't sound as good as if the music were natively at a higher sampling rate, of course, but it's still an improvement.

I find that if you use the MultiResampler DSP in foobar and set it to cubic interpolation, you get a good approximation of N64, Wii (I assume GameCube as well but haven't specifically tested), and 3DS resampling. For PS2 games with 22 or 24 kHz streams, they come out a little more muffled on real hardware than with any of MultiResampler's interpolation options because the PS2 uses an interpolation method not offered.

Note that the Wii and 3DS appear to use cubic interpolation to get everything to 32 kHz for their software mixing stages, and then evidently output with sinc interpolation applied, which allows alias frequencies until about 18 kHz for both systems. This is true even for Wii games played on a Wii U and captured digitally from the HDMI output, so it has to be done in software rather than with a filter in the audio output circuit. But when you're listening to ripped files, you might as well just set MultiResampler to do one pass with cubic interpolation to your target output rate.
by simonmkwii at 4:32 AM EDT on April 30, 2018
@nothingtosay - Don't forget the fact that the GBA and DS weren't capable of interpolation of any kind, so PC music rips using a decent interpolation method can actually sound significantly better than on actual hardware.

Lanczos resampling tends to sound (and look) the best, but if you want those high frequencies back, you'll need to give up all interpolation altogether.

Most people don't understand much about the fourier domain, so I won't bore people with the details.
by MoldyPond at 7:13 AM EDT on April 30, 2018
I've found that (at least for the N64 games with 22kHz) setting the output to 48kHz also sounds close enough.
by nothingtosay at 10:20 AM EDT on April 30, 2018
@simonkwii: Yeah, adding interpolation does a lot to clear up the nastiness of those systems. Audiophiles sometimes criticize digital audio for supposedly reducing smooth, round sound waves to stair-step shapes, which is not what happens because oversampling combined with interpolation/low-pass filtering restore the smooth shape. Well, that criticism is actually basically valid in the case of the GBA and DS!

Lanczos interpolation is a form of sinc interpolation and can tend to make things like drum and cymbal samples on those systems too muffled. If you want high frequencies that sound closer to intended without distortion, cubic interpolation is a good idea. Linear interpolation is only subtly different as well.

@MoldyPond: Indeed. But I have a pedantic note about that. kode54 added that output option to foobar's USF decoder after I discussed this stuff with him a while back. He said at the time it applies the same cubic interpolation scheme that is used in the N64's mixing stage when dealing with samples, but I found that the spectogram isn't quite as similar to that of a real N64 when compared to using MultiResampler's cubic interpolation. ¯\_(ツ)_/¯

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