[mort96]

And actually, why do we have both 48kHz and 44.1kHz anyway? If all "consumer grade high quality audio" was in 44.1kHz (or 48kHz) we probably could've avoided resampling in almost all circumstances other than professional audio contexts (or for already low quality audio like 8kHz files). What benefit do we get out of having both 44.1 and 48 that outweighs all the resampling it causes?

[CharlesW]

> And actually, why do we have both 48kHz and 44.1kHz anyway?

Those two examples emerged independently, like rail standards or any number of other standards one can cite. That's really just the top of the rabbit-hole, since there are 8-20 "standard" audio sample rates, depending how how you count.

This isn't really a drawback, and it does provide flexibility when making tradeoffs for low bitrates (e.g. 8 kHz narrowband voice is fine for most use cases) and for other authoring/editing vs. distribution choices.

[Joeboy]

> This isn't really a drawback

But, that's only true because people freely resample between them all the time and nobody knows or cares about it.

[pixelpoet]

The nice thing about standards is, there are so many from which to choose! :)

[brudgers]

Perhaps the benefit we get is access to existing recordings?

44.1khz exists because it was the lowest technically practical speed and was an optimization for processing speed and storage space.

48khz exists because it syncs with video easily — I’ve also heard it allows for more tolerance in the anti-aliasing filter.

[cormorant]

> 48khz exists because it syncs with video easily

I guess meaning 24fps video? Because 44100 is already a multiple of 25, 30, 50, and 60.

[brudgers]

24fps is where the money is because 24fps is the standard for film.

[adgjlsfhk1]

48k also supports 24, 48, 120, and 240 which are all nice to haves.

[zamadatix]

As far as I understood, both rates ultimately come from trying to map to video standards of the time. 44.1 kHz mapped great to reusing analog tape of the time, 48 kHz mapped better to digital clocking and integer multiples of video standards while also having a slightly wider margin on oversampling the high frequency.

44.1 kHz never really went away because CDs continued using it, allowing them to take any existing 44.1 kHz content as well as to fit slightly more audio per disc.

At the end of the day, the resampling between the two doesn't really matter and is more of a minor inconvenience than anything. There are also lots of other sampling rates which were in use for other things too.

[rerdavies]

> Why do we have both 48kHz and 44.1kHz anyway

Because of greed.

Early audio manufacturers (SONY notably) used 48kHz for profession-grade audio equipment, that would be used in studios or TV stations, and degraded 44.1khz audio for consumer devices. Typically you would pay an order of magnitude more for the 48kHz version of the hardware.

48khz is better for creating and mixing audio. You cannot practically mix audio at 44.1khz without doing very slight damage to audible high frequencies. But enough to make a difference. If you were creating for consumer devices, you would mix at 48Khz, and then downsample to 44.1khz during final mastering, since conversion from 48kHz to 44.1kHz can be done theoretically (and practically) perfectly. (Opinions of the OP notwithstanding).

I think it's safe to say that the 44.1kHz sampling rate was maliciously selected specifically because it is just low enough that perfect playback is still possible, but perfect mixing is practically not possible. And obviously maliciously chosen to be a rate with no convenient greatest common denominator with 48Khz, which would have allowed easy and cheap perfect realtime resampling. Had Sony chose 44.0kHz, it would be trivially easy to do sample rate conversion to 48Khz in realtime even with primitive hardware available in the late 1970s. That extra .1kHz is transparently obvious malice and greed in plain sight.

Presumably SONY would sell you the software or hardware to perform perfect non-realtime conversion of audio from 48khz to 44.1khz for a few tens of thousands of dollars. Not remotely subtle how greedy all of this was.

There has been no serious reason to use 44.1kHz instead of 48kHz for about 50 years, at least from a technology point of view. (And no real reason to EVER use 44.1khz instead of 48kHz other than GREED).

[ianburrell]

The Wikipedia page explains it as coming from PCM adaptors that put digital audio on video tapes. The constraints of recording on videotape led to 44.1kHz being best option. It sounds like there wasn't enough capacity for 48kHz.

Then Sony used the frequency on CDs.

[brendyn]

Are you able to share evidence for this?

[rerdavies]

What would you consider evidence? Emails between standards committee members agreeing to collude in order to screw pro-audio customers?

The evidence is: why on earth would anyone on a standards committee choose 44.1kHz, instead of 44.0kHz? The answer: 44.1kHz was transparently obviously chosen to make it impossible to perform on-the-fly rate conversions.

The mathematics of polyphase rate converters was perfectly well understood at the time these standards were created.

[brewmarche]

Someone else wrote that it was chosen to best match PAL and NTSC. IIRC there is also a Technology Connections video about those early PCM adaptor devices that would record to VHS tape.

<https://en.wikipedia.org/w/index.php?title=44,100_Hz&oldid=1...>

Take it with a grain of salt, I’m not really knowledgeable about this.

E: also note the section about prime number squares below

[rerdavies]

4800kHz and 44100kHz devices appeared at roughly the same time. Sony's first 44100kHz device was shipped in 1979. Phillips wanted to use 44.0kHz.

If you can do 44.1khz on an NSTC recording device, you can do 44.0khz too. Neither NTSC digital format uses the fully available space in the horizontal blanking intervals on an NTSC VHS device, so using less really isn't a problem.

Why is 44Khz better? There's a very easy way to do excellent sample rate conversions from 44.0Khz to 48Khz, you upsample the audio by 12 (by inserting 11 zeros between each sample), apply a 22Khz low-pass filter, and then decimate by 11 (by keeping only every 11th sample. To go in the other direction, upsample by 11, filter, and decimate by 12. Plausibly implementable on 1979 tech. And trivially implementable on modern tech.

To perform the same conversion from 44.1kHz to 48kHz, you would have to upsample by 160, filter at at a sample rate of 160x44.1kHz, and then decimate by 147. Or upsample by 147, filter, and decimate by 160. Impossible with ancient tech, and challenging even on modern tech. (I would imagine modern solutions would use polyphase filters instead, with tables sizes that would be impractical on 1979 VLSI). Polyphase filter tables for 44.0kHz/48.0kHz conversion are massively smaller too.

As for the prime factors... factors of 7 (twice) of 44100 really aren't useful for anything. More useful would be factors of two (five times), which would in increase the greatest common divisor from 300 to 4,000!

[sneak]

CDs used 44.1, DAT and DVDs used 48. That’s it.

Most stuff on the internet ripped from CD is 44.1. 48 is getting more common. We’re like smack in the middle of the 75 year transition period to 48kHz.

For new projects, I use 48, because my mics are 32bit (float!)/48kHz.

[PunchyHamster]

technically we could use 40kHz and just upsample, the extra frequency over 40kHz is basically leeway to make analog part possible/cheap, but it is not technically needed in the signal

the first CD player didn't had compute power to upsample perfectly but modern devices certainly do.

[mort96]

AFAIU, 40kHz exactly wouldn't really work, if your goal is to represent 0Hz-20kHz: in order to avoid aliasing, you need a low pass filter to remove all frequency content above half your sample rate, and no filter is infinitely hard (and you generally want to give the filter a decent range of frequencies to work with). If you want to start your low pass filter at 20kHz, you want it to end (i.e reach practically -∞dB) at a few kHz above 20kHz. If you used a sample rate of exactly 40kHz, you would need your low pass filter to reach -∞dB at 20kHz, meaning it'd have to start somewhere in the audible region.

Though this is just my understanding. Maybe I'm wrong.