How much of a difference does a high-end audio interface really make?

[whassup]

Good video on the topic:

[aMUSEd]

If you are on Mac, perhaps very little, unless you are using hardware or need to use better wired external speakers. Those are the only reasons I use my Fireface UC now - the inbuilt CoreAudio is almost as good in terms of latency and has the advantage of also natively supporting immersive sound, which my display speakers (which are also good enough for day to day use) support. Sometimes I want to mix using my Genelec's, or use some hardware - then I switch on the UC, but most of the time these days I’m just using the built in audio.

[eLawnMust]

It makes absolutely no difference if you don't give a shit about it>>>

[SybilSztorm]

It's actually a pretty good question. I recently acquired a uad 2192 to master my apollox6. How much of a difference does it make? Not much in the context of mixing, a bit more in the context of monitoring but where it really starts being worth it in my opinion is when all your tracks are running in and out of it on their way to your hardware, it does add up. But you need to have this workflow of sending all your tracks out. You have to work hybrid or reasonably out of the box to really benefit. In such conditions I would say it changes your final mix of something like 20%. It does add realism, 3dness. But if you plan to get a high end interface simply to mix entirely in the box, you will most likely only experience a 5% difference and that will probably come from monitoring only.

[Sonic Illusions]

Except recording sensitive mics and precise monitoring.. nothing

[stoopicus]

No impact on:
- Sound output
- Recording instrument inputs (beyond audio rate)

Some impact on:
- Interface latency
- Audio rate choices (higher can be better for ITB workflows)

Definite impact on:
- Recording sensitive microphones (i.e. condensers, not SM-58s)
- Interface quality of life features (monitor control, direct monitoring blend, input level control and feedback, etc)
- Number and type of inputs
- Onboard compression, EQ, effect builtins, effect inserts, etc - i.e. mixers with integrated interfaces have these, also stuff like the Apollo or multieffects boxes
- Specific workflow targeting (mixers again, or stuff like the Bitwig Connect or NI Maschine)

[mixyguy2]

Some impact on:
- Interface latency
- Audio rate choices (higher can be better for ITB workflows)

Nah. Anything above 24/48 is a waste of time.

Definite impact on:
- Recording sensitive microphones (i.e. condensers, not SM-58s)

And again, nah.

The only advantage of more expensive units is more abilities (more inputs/outputs, etc)...so the overall answer in terms of quality of recordings is "little to none." But people love to throw money at something and assume it will help. Don't be one of those people unless you love wasting money (and can afford to do so).

[stoopicus]

Some impact on:
- Interface latency
- Audio rate choices (higher can be better for ITB workflows)

Nah. Anything above 24/48 is a waste of time.

While this is correct in terms of raw audio quality, it’s incorrect in terms of how DSP algorithms work for things like filters, due to basic sampling theory and algorithm behavior as you approach the Nyquist frequency (which is at half the sampling rate.)

As one specific example, an analog low pass filter might have its filter zero at infinity, while a biquadratic DSP algorithm would have the zero at Nyquist. So, the higher the sampling rate, the higher the Nyquist frequency, and the more accurate the modeling.

This is a common example in DSP programming texts. It’s a little contrived as there are other ways around it, but the general principle holds that the farther the Nyquist frequency is above the audible range, the better the audible performance of some algorithms will be. One common consequence of not doing so is aliasing.

This is a major reason so many plugins support oversampling - it has the same effect as increasing the audio rate.

[whyterabbyt]

Nah. Anything above 24/48 is a waste of time.

That depends entirely on what you're actually doing. For sound design (sound design as in film, game, TV sound eg Ben Burtt, not 'I made a preset for a synth') then working with much higher sample rate recordings (96Khz and 192Khz) is pretty common because of the ubiquity of pitchshifting audio down by multiple octaves as a core technique. Working with actual signal above 20Khz is required.

[koalaboy]

The only advantage of more expensive units is more abilities (more inputs/outputs, etc)...so the overall answer in terms of quality of recordings is "little to none." But people love to throw money at something and assume it will help. Don't be one of those people unless you love wasting money (and can afford to do so).

That's not true at all. The difference between my previous RME BabyFace and my more recent RME Fireface UFX 2+ is audible to me, who is not an audiophile or producer in any way. And that's not something I expected - I just expected more I/O and some other features, but it turns out (no surprise) that the huge difference in price does actually mean a clearer sound.

Now is it worth that much money, for a bedroom producer, such that listeners of the end result would notice ? Probably not. That's true for most audio, especially as it's often streamed and compressed.

But when listening even to YouTube videos, I have more clarity and a different (better) soundstage, through exactly the same headphones I've always used. The extra feature are also valuable to me, but there *is* an audible difference. Perhaps it's just unit variance or other failing electronics. But I would buy another UFX (or equivalent level) in a heartbeat if this one failed.

[stoopicus]

Nah. Anything above 24/48 is a waste of time.

That depends entirely on what you're actually doing. For sound design (sound design as in film, game, TV sound eg Ben Burtt, not 'I made a preset for a synth') then working with much higher sample rate recordings (96Khz and 192Khz) is pretty common because of the ubiquity of pitchshifting audio down by multiple octaves as a core technique. Working with actual signal above 20Khz is required.

Yeah there's this too, which actually combines with the reason I mentioned for work ITB.

24 bit is enough. But the idea that anything above 44/48kHz is useless is only true in the most straightforward use cases for recording. Unless you're just using a DAW as a glorified multitrack tape recorder, there are legit reasons to want a higher audio rate.

[VELLTONE MUSIC]

24bit/48khz is a standard from 90's ...
Entire decade people listened 16bit/41khz crappy cd's and in the end somebody say 'Hey this is low quality audio' and cd business start to melt like spring snow.
Then music go to dvd cinema standard 24/48 and that was ok for 2000-2010 maybe,but now is 2025...
If somebody think than 24bit/48khz is same as 32bit/96khz probably have to check again more carefully.
The difference is audible immediately,but most common hardware devices are 24/48, so can't enjoy the real beauty of 32bit/96 khz and the size of 32bit float audio record is 500 mb vs 20mb under 24bit.
I am in search of new sound card with 32bit/192khz internal processing,but not many affordable options available.
Not mentioned often,but 32bit float audio is the format for true audiophiles :):)
Cheers

[Ploki]

24bit/48khz is a standard from 90's ...
Entire decade people listened 16bit/41khz crappy cd's and in the end somebody say 'Hey this is low quality audio' and cd business start to melt like spring snow.
Then music go to dvd cinema standard 24/48 and that was ok for 2000-2010 maybe,but now is 2025...
If somebody think than 24bit/48khz is same as 32bit/96khz probably have to check again more carefully.
The difference is audible immediately,but most common hardware devices are 24/48, so can't enjoy the real beauty of 32bit/96 khz and the size of 32bit float audio record is 500 mb vs 20mb under 24bit.
I am in search of new sound card with 32bit/192khz internal processing,but not many affordable options available.
Not mentioned often,but 32bit float audio is the format for true audiophiles :):)
Cheers

Not sure if you're serious, but 24bit is fine for studio.
it has 144dB of dynamic range, and analog paths are limited by physics to about 120dB, for now.
Recording 32bit would mean you would 48dB of thermal noise to recordings.

32bit floating point is effectively 23bit mantissa, 1bit sign, and 8bit exponent. Effective resolution is 24bit, scaled. Single sample cannot have more than 24bits of resolution with 32bit FP.

As far as 48k vs 96k goes, i won't even go into that.

Yeah there's this too, which actually combines with the reason I mentioned for work ITB.

24 bit is enough. But the idea that anything above 44/48kHz is useless is only true in the most straightforward use cases for recording. Unless you're just using a DAW as a glorified multitrack tape recorder, there are legit reasons to want a higher audio rate.

which are these legit reasons?
I can name one practical reason, but you go first.

The only advantage of more expensive units is more abilities (more inputs/outputs, etc)...so the overall answer in terms of quality of recordings is "little to none." But people love to throw money at something and assume it will help. Don't be one of those people unless you love wasting money (and can afford to do so).

That's not true at all. The difference between my previous RME BabyFace and my more recent RME Fireface UFX 2+ is audible to me, who is not an audiophile or producer in any way. And that's not something I expected - I just expected more I/O and some other features, but it turns out (no surprise) that the huge difference in price does actually mean a clearer sound.

Now is it worth that much money, for a bedroom producer, such that listeners of the end result would notice ? Probably not. That's true for most audio, especially as it's often streamed and compressed.

But when listening even to YouTube videos, I have more clarity and a different (better) soundstage, through exactly the same headphones I've always used. The extra feature are also valuable to me, but there *is* an audible difference. Perhaps it's just unit variance or other failing electronics. But I would buy another UFX (or equivalent level) in a heartbeat if this one failed.

err, there is no UFX 2+. There's UFX I, II, III and UFX+.
I have UFX+ here, and i have an ancient Fireface800 next to it, piped in via ADAT.
(Also have an old Fireface400 in the other room, "Sound" is not why it's going to get an upgrade)

Surprise surprise, there is no surprise. There's no elusive clarity when watching compress YT garbage.
You're hearing money you spent.

This is a common example in DSP programming texts. It’s a little contrived as there are other ways around it, but the general principle holds that the farther the Nyquist frequency is above the audible range, the better the audible performance of some algorithms will be. One common consequence of not doing so is aliasing.

This is a major reason so many plugins support oversampling - it has the same effect as increasing the audio rate.

No it does not. Oversampling is nearly always done in conjunction with an antialiasing filter. Running at higher sampling rates will produce more aliasing than running @48k with antialiasing enabled for that very reason. If you just run at 96k, shit will get mirrored over Nyquist.
If you run at 96k internally, then filter out everything above nyquist, and THEN downsample, you won't get aliasing.

There's a case for higher sampling rates and audio quality and that is, removing AA filter further from audible range, but it's obviously non-consequential enough we're still doing 48k instead of years ago proposed 60k.

[koalaboy]

err, there is no UFX 2+. There's UFX I, II, III and UFX+.

I use the + to mean the more recent UFXII hardware that had similar upgrades to the UFX+ and UFXIII from what I recall. You're correct that it's not an official SKU, but there are hardware differences.

[stoopicus]

Yeah there's this too, which actually combines with the reason I mentioned for work ITB.

24 bit is enough. But the idea that anything above 44/48kHz is useless is only true in the most straightforward use cases for recording. Unless you're just using a DAW as a glorified multitrack tape recorder, there are legit reasons to want a higher audio rate.

which are these legit reasons?
I can name one practical reason, but you go first.

You quote them below

This is a common example in DSP programming texts. It’s a little contrived as there are other ways around it, but the general principle holds that the farther the Nyquist frequency is above the audible range, the better the audible performance of some algorithms will be. One common consequence of not doing so is aliasing.

This is a major reason so many plugins support oversampling - it has the same effect as increasing the audio rate.

No it does not. Oversampling is nearly always done in conjunction with an antialiasing filter. Running at higher sampling rates will produce more aliasing than running @48k with antialiasing enabled for that very reason. If you just run at 96k, shit will get mirrored over Nyquist.
If you run at 96k internally, then filter out everything above nyquist, and THEN downsample, you won't get aliasing.

There's a case for higher sampling rates and audio quality and that is, removing AA filter further from audible range, but it's obviously non-consequential enough we're still doing 48k instead of years ago proposed 60k.

Yes, of course you need to use an antialiasing (usually brick wall) filter up near Nyquist as well. But the fact remains that filtering being equivalent, 96kHz sampling rate is equal to (and technically superior to, as there's no interpolation) 2x oversampling at 48kHz.

But the other issue is the behavior of the various filter algorithms themselves near Nyquist. If you can keep Nyquist farther and farther above the audio range, the behavior of the digital filters transfer functions inside the audio range will more closely match that of the analog ones. This is simply a fact for some algorithms.

(And yes, I am aware and recognize that you do have more experience than I do at DSP programming - I am more of a student of it, though I am working on changing that.)