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by MoldyPond at 7:20 AM EDT on May 14, 2018
I enjoy the GBA's quality as is too, but only because we don't really have any other option. I mean, if the GBA DID have an option for high-quality sound, would you be complaining? (Also why is this thread suddenly so toxic?)
by simonmkwii at 11:50 AM EDT on May 14, 2018
SimonMKWii - Today at 12:13 AM
@bxaimc My god, the "Ways to improve 24 khz or lower sound files using audacity/audition" thread is a total nightmare!
It's basically just people flaming each other

bxaimc - Today at 12:18 AM
You expected something different?
I didn’t
everyone that responded in that thread has their own quirky ego

This thread has become a meme
by VIRGIN KLM at 12:42 PM EDT on May 14, 2018
@derselbst This is not linear interpolation. There is either some fault or there is no pre-filtering applied. I don't know from where you got the samples but there is somekind of fault in the implementation. Try instead the linear interpolation method and sinc interpolation option in NDS tracks. This implementation is done correctly.

@MoldyPond There are ways to render GBA audio in higher quality that GBA's DSP. One is to convert tracks to BASS midi and the other is this: Agbplay

Also for the record here's what happens with and without Linear interpolation. Any other than Linear interpolation will introduce a level of the artifacts shown on the right side of the Spectogram since they all rely on creating those fake metalic sounding harmonics:

edited 12:49 PM EDT May 14, 2018
by simonmkwii at 1:21 PM EDT on May 14, 2018
@virgin klm - the middle spek looks correct to me
by derselbst at 2:11 PM EDT on May 14, 2018
@VIRGIN KLM: I've made it myself using fluidsynth. I can reproduce the same behaviour with libsamplerate. Which is totally comprehensible, because linear interpolation is not band-limited, while sinc interpolation is. Reconstructing a sample point without introducing artifacts cannot be done by only taking two samples into account as linear does. In fact all existing samples contribute to the sample being reconstructed. In other words: you must use a sinc interpolation in time domain to get a rectangular response in frequency domain that you want, rather than a sinc^2 frequency response like linear interpolation yields. You're mixing up sinc and linear interpolation.
by SmartOne at 10:22 PM EDT on May 14, 2018
I think you're confusing "ego" with "autism."

The cool kids use chat.

Talking past each other is fun. As was my sarcasm. Right kode54?

But yes, kode54 did get it right in post #2.

There's nothing objective about subjectively enhancing source material. There's nothing wrong with it, but it's up to personal taste, and shouldn't be advertised as something more right. There's already enough confusion without that (see this thread).
by nothingtosay at 11:20 PM EDT on May 14, 2018
I have a quirky ego? Meh, I'll take it. There are certainly worse things to have. Surely a natural consequence of being very interested in/passionate about almost anything, but especially things as niche as video game music and hardware behavior.

By the way, I've gone all this time without saying yo bx. It's been a while, but I still remember you as a kickass dude. I noticed arbingordon also popped in here some pages back. I enjoyed chatting with both of you and hcs back half a decade ago or something, hope y'all are doing well.
by kode54 at 11:53 PM EDT on May 21, 2018
You're going to love the pending QSound emulation that's coming to MAME and VGMPlay and is already in libVGM in a fast HLE form. It's hard clocked to 24038Hz, or thereabout. And there's no way to change that. It's HLE code based on a 60MHz DSP16A, which runs specific code, and has specific FIR filter coefficients, and specific delay line lengths in game specified sample counts. Making it produce a higher sample rate is not really feasible without in fact harming the sound quality.
by simonmkwii at 3:26 AM EDT on May 22, 2018
@kode54 - Can you at least give an option to ZOH it to 48076Hz during post?
by kode54 at 4:25 AM EDT on May 22, 2018
How disgusting, mirrored aliasing. When it's in foo_input_qsf, you can do what you want, but when it'll be in foo_gep or foo_input_vgm, it will use the plugin's own linear interpolation, with no other option.

Also, 24kHz isn't much lower than my threshold of hearing anyway. 12kHz vs 14.5kHz.

edited 4:27 AM EDT May 22, 2018

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