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

[stoopicus]

With oversampling - my understanding anyway, I haven't actually implemented it myself, but I have studied it (Pirkle, Oppenheim & Schafer, etc) - basically you either pad the samples out by alternating with zeros or interpolated values for the added samples, apply the effect and the steep antialiasing filter, and then downsample back.

With a higher sample rate, you have actual samples there and not interpolated ones and get to skip the additional filtering as well (and resulting phase effects, etc).

Like I said, it's a slight difference.

you only get to skip filtering if the delivery format is 88.2/96k, else, you will, at one point have to apply filtering.
I'm a proponent of DAW-level oversampling so you only need to filter once per channel strip for example, not every plugin consecutively. But then again, you only ever oversample compressors and saturators so a lot of overhead for little benefit

So, if "actual samples" are more or less noise, since there's rarely any useful signal above 20k, what's the effective difference between interpolated and recorded samples?

That's actually a good point and also the reason padding with zeros is even an option, or at least how I understand it. Again, like I said, it's a small difference.

[Ploki]

With oversampling - my understanding anyway, I haven't actually implemented it myself, but I have studied it (Pirkle, Oppenheim & Schafer, etc) - basically you either pad the samples out by alternating with zeros or interpolated values for the added samples, apply the effect and the steep antialiasing filter, and then downsample back.

With a higher sample rate, you have actual samples there and not interpolated ones and get to skip the additional filtering as well (and resulting phase effects, etc).

Like I said, it's a slight difference.

you only get to skip filtering if the delivery format is 88.2/96k, else, you will, at one point have to apply filtering.
I'm a proponent of DAW-level oversampling so you only need to filter once per channel strip for example, not every plugin consecutively. But then again, you only ever oversample compressors and saturators so a lot of overhead for little benefit

So, if "actual samples" are more or less noise, since there's rarely any useful signal above 20k, what's the effective difference between interpolated and recorded samples?

That's actually a good point and also the reason padding with zeros is even an option, or at least how I understand it. Again, like I said, it's a small difference.

I tested some plugins a while ago with oversampling vs native @2x resolution and for some reason, native had more aliasing. Go figure. Need to do some more with newer plugins

[stoopicus]

With oversampling - my understanding anyway, I haven't actually implemented it myself, but I have studied it (Pirkle, Oppenheim & Schafer, etc) - basically you either pad the samples out by alternating with zeros or interpolated values for the added samples, apply the effect and the steep antialiasing filter, and then downsample back.

With a higher sample rate, you have actual samples there and not interpolated ones and get to skip the additional filtering as well (and resulting phase effects, etc).

Like I said, it's a slight difference.

you only get to skip filtering if the delivery format is 88.2/96k, else, you will, at one point have to apply filtering.
I'm a proponent of DAW-level oversampling so you only need to filter once per channel strip for example, not every plugin consecutively. But then again, you only ever oversample compressors and saturators so a lot of overhead for little benefit

So, if "actual samples" are more or less noise, since there's rarely any useful signal above 20k, what's the effective difference between interpolated and recorded samples?

That's actually a good point and also the reason padding with zeros is even an option, or at least how I understand it. Again, like I said, it's a small difference.

I tested some plugins a while ago with oversampling vs native @2x resolution and for some reason, native had more aliasing. Go figure. Need to do some more with newer plugins

Yeah could be due to the brick wall added for the oversampling having other beneficial effects.

[VELLTONE MUSIC]

I have basic internet education and understanding of digital stuff,

then maybe listen to the people with actual qualifications and jobs in the field

I am listening the sound,not people,which has nothing to say.

[VELLTONE MUSIC]

So,after few guys,including me,which actually have heard how 32 float sounds ,compared to 24 bit,which most probably the opposition on topic here doesn't - the bigger size and more information for 'reading' in such file,after the internet info about differences so on, the topic will continue with conclusion that studio sound doesn't need or benefit anything better than 24/48 ...hahahah same people,same conversation here ...funny
Cheers

[koalaboy]

If there is no use for 32bit in 'the studio', why do DAWs mix at 64bit FP ?

IMHO:

Record at the highest (with the best) rate you can. If you are capturing noise with a tiny Dynamic Range, then sure use 24bit, but you may have a much higher requirement or simply not want to worry about gain-staging everything accurately, 32bit can help in those situations - that's the benefit of technology. You're still allowed to do things the 'old' way if you prefer.

In the box, use the most flexible setup you can. For me, that is 32bit Float for all audio. It's all digital and binary maths, so best to keep the most headroom you can. Plugins, etc. all make a difference. If you are using any outboard in the processing, this will complicate things massively, and perhaps that is the best reason *not* to use 32bit if you are doing so, as you will likely lose data in the truncation each time. If you're completely "In The Box" this isn't an issue.

Final mixdown is the most important, as you need to reduce the 'box' to the real world, which is likely 24bit or 16bit. So this needs (in the DAW) to bring all of the audio into an 'expected' range, which may require dithering as well, to get whatever huge DR down to the output. This is also why you need to listen to the mix on different speakers/etc. as they all reproduce in different ways. You're effectively 'sculpting' the audio to playback well on the most common devices people will use.

That's it. Oh, and have fun doing it whatever way you choose.

To bring it back on topic, why do high-end interfaces matter ? Because the convertors effect the sound, and aspects such as Jitter affect the conversion to/from digital to analog. Some are more accurate, some are more colouful. Most are easily 'good enough' but it comes down to the sound reproduction that you prefer. Higher-end interfaces generally have more accurate conversion and transparency or more pleasing colouring when driven or saturated. Subtle and nuanced maybe, but that's more personal choice... just like music.

[Ploki]

If there is no use for 32bit in 'the studio', why do DAWs mix at 64bit FP ?

IMHO:

Record at the highest (with the best) rate you can. If you are capturing noise with a tiny Dynamic Range, then sure use 24bit, but you may have a much higher requirement or simply not want to worry about gain-staging everything accurately, 32bit can help in those situations - that's the benefit of technology. You're still allowed to do things the 'old' way if you prefer.

In the box, use the most flexible setup you can. For me, that is 32bit Float for all audio. It's all digital and binary maths, so best to keep the most headroom you can. Plugins, etc. all make a difference. If you are using any outboard in the processing, this will complicate things massively, and perhaps that is the best reason *not* to use 32bit if you are doing so, as you will likely lose data in the truncation each time. If you're completely "In The Box" this isn't an issue.

Final mixdown is the most important, as you need to reduce the 'box' to the real world, which is likely 24bit or 16bit. So this needs (in the DAW) to bring all of the audio into an 'expected' range, which may require dithering as well, to get whatever huge DR down to the output. This is also why you need to listen to the mix on different speakers/etc. as they all reproduce in different ways. You're effectively 'sculpting' the audio to playback well on the most common devices people will use.

That's it. Oh, and have fun doing it whatever way you choose.

To bring it back on topic, why do high-end interfaces matter ? Because the convertors effect the sound, and aspects such as Jitter affect the conversion to/from digital to analog. Some are more accurate, some are more colouful. Most are easily 'good enough' but it comes down to the sound reproduction that you prefer. Higher-end interfaces generally have more accurate conversion and transparency or more pleasing colouring when driven or saturated. Subtle and nuanced maybe, but that's more personal choice... just like music.

For recording. Not for mixing.
f**ks sake we’re talking about interfaces not internal processing of software.

Audio recorded at 24bits is still processed as 32/64bit fp by software. Software doesn’t work with audiofile bit depth, it works with whatever it’s coded with.

32bit FP “recording” is more or less 24bit recording wrapped in 32bit fp format (unless dedicated recorders like sound devices, where i already explained how they worked)

Well treated and designed rooms have more speaker pairs but 99% of the time, one pair is used by the engineer.
Speaker hopping is done when you have poor monitoring conditions.

Go read a book on digital audio

[koalaboy]

Audio recorded at 24bits is still processed as 32/64bit fp by software. Software doesn’t work with audiofile bit depth, it works with whatever it’s coded with.

Right... so people never bounce down to audio in a DAW.

[whyterabbyt]

You mistaken 32bit fixed point and float point i guess.Fixed point is small size.

Float point takes huge amount of space -

A fixed-point 24-bit value takes up 24 bits of space.
A floating-point 32-bit value takes up 32 bits of space.
A fixed-point 32-bit value takes up 32 bits of space.

3-4 min audio is around 400-500 mb.

A raw file of 4 minutes of stereo 32-bit 96K audio takes
4 (minutes) x 60 (seconds) x 96000 (samples) x 2 (channels) x 4 (bytes per word) = 184,320,000 bytes = 180Mb

A raw file of 4 minutes of stereo 24-bit 48K audio takes
4 (minutes) x 60 (seconds) x 48000 (samples) x 2 (channels) x 3 (bytes per word) = 69,120,000 bytes = 67.5Mb

That's how maths, and computers work. If a RAW export of audio, without any metadata, is taking more space than that, then you're either exporting wrong, or misunderstanding what you're doing, or your DAW is broken.

feel free to prove that the maths is wrong though.

Export something and you'll see.

I did, and I do see. I see the exact same file sizes as my calculations above predicted.

Screenshot 2025-08-23 114401.png

I have basic internet education and understanding of digital stuff

Hmmm. Maybe too basic, and more misunderstanding than you realise.

edit : to fix quote block

[Ploki]

Audio recorded at 24bits is still processed as 32/64bit fp by software. Software doesn’t work with audiofile bit depth, it works with whatever it’s coded with.

Right... so people never bounce down to audio in a DAW.

Bouncing down is still software…

This is a thread about interfaces.

[koalaboy]

Audio recorded at 24bits is still processed as 32/64bit fp by software. Software doesn’t work with audiofile bit depth, it works with whatever it’s coded with.

Right... so people never bounce down to audio in a DAW.

Bouncing down is still software…

This is a thread about interfaces.

Right, and what's the point of using a high-end interface if you're going to bounce down in a way that will lose data later. That's the point - you want the highest quality input *and to maintain that* throughout the DAW process until the final mixdown. Otherwise, you've lost the point of the 'high-end' that you paid so much money for.

A high-end interface is of no benefit if the rest of your chain is discarding that information.

[Ploki]

Audio recorded at 24bits is still processed as 32/64bit fp by software. Software doesn’t work with audiofile bit depth, it works with whatever it’s coded with.

Right... so people never bounce down to audio in a DAW.

Bouncing down is still software…

This is a thread about interfaces.

Right, and what's the point of using a high-end interface if you're going to bounce down in a way that will lose data later. That's the point - you want the highest quality input *and to maintain that* throughout the DAW process until the final mixdown. Otherwise, you've lost the point of the 'high-end' that you paid so much money for.

A high-end interface is of no benefit if the rest of your chain is discarding that information.

Bruh, what are you even talking about?
I never talked about bouncing down, you don’t even need to have a dedicated interface to bounce down or process audio or use a DAW. (Actually you don’t need it at all, because it’s maths.)

Interfaces have at very f**king best 120dB SNR, if you record at -20dB peak, 24bit is still enough to avoid 24bit rounding error. Preamps with 120dB of SNR dont even exist.

[whyterabbyt]

Right, and what's the point of using a high-end interface if you're going to bounce down in a way that will lose data later.

what data do you think you lose?

i mean, i presume you understand that a sample of audio is a value that represents amplitude of a signal at that point?

do you think that all the potential differences in amplitude that can be stored in 24-bit or 32-bit or even 64-bit samples are perceivable by the human ear?

do you think that they are transmissable to your ears at the same degree of accuracy, through the electronic and electromagnetic device chains that exist on either side of the ADC/DAC in the interface?

[VELLTONE MUSIC]

This topic turns into a fight between musicians and mathematicians hahahah :):)
It cross my mind to record a video in details and to export in real time,so to see the answer of calculating people and to ''prove'' my point,but don't have time and desire to continue a conversation,when one side ,self promoted ''experts'' continue not to understand a single word i am saying,even a simple thing, that fixed and float are different animals so on...
Could some calculator give an advise on something?
I am about to download Dolby Atmos Composer Essential (free version).
Do i need to waste my time with it or to stick with 24/48 stereo format....rhetoric question hahahah :):)
Every time the same,self announced specialists start a war with typical for their cast aggressive and insulting behavior (which is enough to me to understand what kind of people are on the other side),just to prove something,instead of discuss it and to expand their point of view...
Not interested to waste time discussing with non musicians a music related stuff...
Calculate and prove whatever you like,go even 8bit if you like and claim that this is the best studio format...the topic is interested,but turns into fight,for what...everybody is free to understand...or not ...whatever ...

[Etienne1973]

No analog gear goes beyond what can be captured by a 20 bit integer.

This is the issue on the input side before the AD conversion I think.

do you think that all the potential differences in amplitude that can be stored in 24-bit or 32-bit or even 64-bit samples are perceivable by the human ear?

That is the issue on the output side after the DA conversion I think.