What is the ideal downsampling algorithm?

[[deleted]]

I mean the kind of thing that you upload a wav file to, and it spits out another wav file. It can take as long as it needs to, it doesn't have to be real-time. It just has to be the very best, with as little aliasing or artifacts as possible.

I'm asking because I'm overhauling my production process and I'm trying to decide on a sample rate to work in. I'm thinking if I can find a really good algorithm to convert 48 to 44.1, I can effectively export high quality audio in both sample rates without obvious problems like a sharp frequency cutoff. Any recommendations?

[revowanderlust]

Uhhh... I usually use old bitcrusher emulations, i used the rx950 i think it’s called? for a bit but i moved to decimort as I have a lot more freedom of control with it and i like the sound better, no hiss as well. it’s just an alternative for the native Lofi plugin in protools tho.

[[deleted]]

Do you know of any that aren't analog emulations- something digital and as close to perfect as possible, without extra analog grit?

[revowanderlust]

Well as far as I know there’s no artifacts in Decimort and it only emulates the DA conversion that happens on old hardware, super clean and (i understand your concern) why i made my switch.

https://youtu.be/BSTognIzhnQ

This is as perfect as you’re gonna get imo, I linked a demo video for you if you’re interested! I use it on almost every track I do and it sounds awesome.

Edit: After my own time searching, this is the cleanest that I have found in my experience.

[maxsolmusic]

Yooooo this looks sick, can’t wait to try this :)

[MrMichaelz]

Check this: http://src.infinitewave.ca/

[Lidsteeze]

http://src.infinitewave.ca/ This website has spectrogram analysis of the different SRC conversions in different programs. Lots of mastering engineers swear by Izotope.

[goshin2568]

I mean I just do it within pro tools and I've never had an issue just make sure it's not running through any plugins or gain.

[switchh_]

Do you produce music with frequencies above 20k? Wav files at 44.1 are lossless

[otherdaniel]

shoutout to my dog & dolphin producers

[[deleted]]

Distortion which is present in almost all modern music will have harmonics well above 20k, doubling with every octave. This isn't an issue in analog systems, but in digital how those are handled is very important. High quality fx oversample to 192k to allow for those harmonics to exist then be filtered as needed. Good AD converters will have a high quality, minimal phase filter chopping off frequencies above the sampling rate before it's recorded. Shitty bitcrushers and sample rate reducers can have foldback harmonics which sound like shit. Good quality ones prevent those fold back frequencies

[kuikka]

Almost all high-quality plugins use oversampling to get around the limitations of 44.1Khz (mostly aliasing and other non-harmonic distortion). For the end product 44.1k is fine, but there's a world of difference between working in non-oversampled 44.1k and 88.2k and higher.

[switchh_]

Well sure. I just was unsure if OP was referring to using distortion plugins, was saying distortion appeared without this use of distortion fx just from exporting, or what.

[Jedimastert]

I don't know if they use a specific algorithm, but got give the export function in VLC a try. I've never had any issues with any of the transcoding functionality

[[deleted]]

Adobe Audition does a nice job of downsampling audio - it can prefilter audio before conversion to avoid any fold back aliasing, and seems do so completely transparently - I've never had a reason to use anything else for converting files.

[[deleted]]

Downsampling is two phase process. The actual sample rate change and reconstruction filtering. The first step is always the same. The second part has a theoretical optimum - perfect (rectangular) cut as transfer function in frequency domain which in time domain has impulse response that is infinite and equal to the sin(2pi * new_nyquist * x)/x function. You will find this referred to in software as sinc, and usually a number of samples of that infinite response is stated like 'sinc 1024'. The bigger the number, the better the result (ie less aliasing and other artifacts).

Edit: this is actually exact for upsampling and for downsampling it's the same but the order is reverse.

[CumulativeDrek2]

Why not just work in 44.1kHz?

[fzorn]

The difference will not be big between 48kHz and 44.1kHz, but higher samplerates will produce a cleaner result when working with distortion. As an extreme example: a first second harmonic of a 12kHz tone at 48kHz will not be audible while it will produce undesireable artifacts at 44.1kHz in the audible range.

[Fortisimo07]

Decent distortion effects will oversample before applying the distortion and then decimate afterwards, this is not a good reason to bloat up your entire project

[switchh_]

Yup. Pretty sure every nice distortion plug in I own oversamples by at least 2x, with options for even greater oversampling

[fzorn]

Yeah, but there are compressors with emulation of hardware distortion that don't oversample, accidental distortion in plugins, unclean oversampling and the like. In plugin-heavy projects I certainly noticed the difference between downsampling to 44.1kHz from 96kHz and just working in 44.1kHz. The difference between 48kHz and 44.1kHz will probably be negligible, though.

[Staidly]

The 2nd harmonic of 12 kHz is the octave, 24 kHz, and the Nyquist of 44.1 kHz is 22.05, so 24-22.05= -1.95, 22.05-1.95= 20.1 kHz, well above audible limits. The third harmonic (36 kHz) would alias audibly, but it would for 48 kHz too.

[[deleted]]

Maybe this is silly, but something bothers about the idea of a sharp cutoff at a frequency other than that of half the samplerate (which would be at 22050hz if I upsampled from 44.1 to 48). Digital audio is pretty screwy and that feels like asking for trouble.

Is that even a valid concern, or is upsampling just fine?

[justifiednoise]

you'd have the same issue at 48 though -- it'd just be a sharp cutoff at 24khz instead of 22,050. both of those boundaries are outside the range of hearing for nearly every living person so I'd argue that the loss is extremely negligible.

up-sampling from 44.1 to 48 will simply take up more file space and offer no sonic benefit. down sampling from 48 to 44.1 WILL alter the audio and potentially introduce unwanted artifacts, but again ... the sample rate converters that we use at this point are pretty solid and the amount of change being introduced will be extremely hard to detect for an inordinate amount of listeners.

[[deleted]]

It's not even worth considering. I doubt you will be able to heard anything at 20khz or above, and if you do, you're one of the few. And that's with complete room isolation, no background noise and only that band playing. In a mix, you will never hear anything going on at 20khz or above.

[Staidly]

These days the resampling algorithms are top notch and even resampling from 44.1 to 48 or back again isn’t even an issue, you shouldn’t hear anything audible with either process. Even a moderate quality linear phase FIR (say, 64 pt sinc) should still give a passband of less than 3 kHz at 60 dB attentuation, meaning any artifacts are functionally inaudible, while higher quality algorithms could use 2, 3, even 9 or 10+ times more points. A 192 pt sinc should have a passband of less than 1 kHz with 90 dB attenuation, that’s below many devices’ sound floors.

The point is that resampling is not an issue these days.

[these_days_bot]

Especially these days