24/96khz

[Uncle E]

I don't know what I should say about this. I thought 96kHz is the holy grail?

If you prefer any of your plug-ins at more than 2x oversampling, you prefer beyond 96kHz.

Another thing that bothers me... why 44kHz?
Oversampled (at least twice) results in 88kHz. Wouldn't 48kHz be more sufficient and less CPU intensive considering that a lot(!) of hardware modules run with an internal clock of 48kHz?

[IMO]
As far as recording audio is concerned, 48kHz already gets you most of the improvement that people are looking for from higher sample rates. However, if what you're going for is improvement in your plug-ins, 88.2kHz is a massive improvement over 48kHz and actually takes a lot less math to get to 44.1kHz than 48kHz does.

Anyway, to directly address your question, yes, the standard should be 48kHz, not 44.1kHz.
[/IMO]

[hibidy]

A man of reason. But you should consult.......oh wait, n/m........

[Compyfox]

Hey Compyfox,

This is what I do -

Step 1: Open my DAW

Step 2: Make music

Step 3: Enjoy

Peace y'all!

So this was more or less a piss-take on the topic? Hm...

But you're right - less talking, more music.

If you prefer any of your plug-ins at more than 2x oversampling, you prefer beyond 96kHz.

Well but we're talking about "Internal Oversampling". It's IMO essential for distortion devices (compressors!) and one or another FX section.

We can call it cheating to use OS (which gives us a short-cut to use higher samplerates for certain tools, without creating higher samplerate content ourself), but the load on HDDs with 44/48kHz is just not as much as with 96kHz. Then again, I remember the "Upmix your production" thread where the general opinion was, that rendering at a higher samplerate doesn't add anything to the equation.

And, we have another problem... let's say we do(!) work at higher sampling rates right off the bat, do internal OS matrixes still work?

[IMO]As far as recording audio is concerned, 48kHz already gets you most of the improvement that people are looking for from higher sample rates. However, if what you're going for is improvement in your plug-ins, 88.2kHz is a massive improvement over 48kHz and actually takes a lot less math to get to 44.1kHz than 48kHz does.

But 88.2kHz (which is twice of 44.1kHz) is an odd number to 48kHz (twice as much would be 96kHz). So there is still a lot of math involved. Same with going from 48kHz to 44.1kHz. Thankfully there are some really good standalone SRC's that can handle that, but still.

Anyway, to directly address your question, yes, the standard should be 48kHz, not 44.1kHz.
[/IMO]

If you want to be flexible in terms of workareas and being somewhat future-ready - then yes I agree. Made the switch years ago due to lower latency as well. Never looked back, never got any complaints either.

Only Wavelab is a b'tch if it comes to CD authoring (it only likes 44kHz in this case - so mastering in a montage in 96kHz is moot).

[digidennis]

Double-rate DSD ( 128 x oversampling ) @ 1 bit is the new black, sounds reeeeally analog ( http://en.wikipedia.org/wiki/Direct_Stream_Digital )

[whyterabbyt]

everytime you run a soft synth at a higher sampling rate, it will sound better

everytime you run any analog modeled plugin that has any nonlinearities (and that is most "analog modeled" plugins since like 2003) at a higher sampling rate, it will sound better

its been proven by the scientists already

as soon as you provide the scientific definition they're using for 'sounds better' i'll believe you. however, i'm pretty sure there's no actual scientific correlation between the conflation of 'higher frequency content' and 'sounds better' that you're making here.

as for 'analog modelled', im not entirely sure what aspect of that you're referring to, but if you're talking about higher modulation rates etc, then i'd be interested in comments from the respective DSP gurus round here as to whether higher sampling rates actually provide significant improvements in accuracy, result, comparison to oversampling the respective stages in the signal chain. and im aware that you've said that you consider oversampling has issues because of the downsampling stage, but Im also not yet convinced that this is results in provably 'worse' (ie not just a provably more band-limited) sound than a higher sample rate. comments from the respective DSP gurus would be welcome.

As for nonlinearities, I'd wonder if, for frequency content above the Nyquist limit for 44.1Khz/48Khz sample rates, most normal monitors and rooms contribute enough audible nonlinearities to render the point moot for most human's hearing.

In short, Im glad you're happy with your choice. It would be nice if you tempered the presentation of yourself as the only one in the room smart enough to have 'discovered' a new silver bullet, though.

dude. heh.

load up a fabfilter synth. Go to the highest octave and play sawtooth wave, pitch bend up and down.

Do this at 44.1khz

Then repeat it at 96khz

are you having a problem with the concept of scientific proof?

[Compyfox]

Double-rate DSD ( 128 x oversampling ) @ 1 bit is the new black, sounds reeeeally analog ( http://en.wikipedia.org/wiki/Direct_Stream_Digital )

Several problems with that system:
1) only Sony and KORG used that system so far, IIRC, Yamaha also tried to go into that area and failed
2) no backing support from the industry (SACD's failed, which brought up the technology in the first place, though Blu-Ray as medium succeeded - and the players are backwards compatible)
3) tons of heated debates in whether or not DSD is superior to PCM


I'd support such a system, if it would be backwards compatible to existing (say) ADAT ADC/DAC devices, or if they'd offer an update to these devices (meaning I can still record with my famous ADC but in 1bit instead of 24bit). Else I need to rebuy all of my gear, or hope for certain updates. I do trust RME in that section, but it won't be cheap if it'll ever happen.

I think development and building cost are also a major factor in actually not porting that. Hosts also need to implement and support it. Not to mention CPU usage on playback - unless the load will be handled on DSPs "in" the recording module.

[jupiter8]

Jitter may cause a sample to be misinterpreted and cause pops and crackle.

Jitter doesn't cause pops and crackles. Whatever gave you that idea ? At most it will give the sound a kind of "haze" but it is inaudible in all modern soundcards. Jitter is a nonissue.

Pops are the same thing to me whether caused by buffer underrun or misinterpreted sample.

Well great but jitter doesn't cause pops. At least if you have a soundcard that is less than a 100 years old.

Well, 48k and 96k is not the same sample rate - is it?

Wow really ?

So the PLL has different datarate to handle, right?

The chips CS8412 and CS8416 in the DACs I built differentiate on that.

And the obvious - it got to be double accurate not to misinterpret at 96k.

Well the thing that you obviously missed is that 48 and 96 runs from the same clock. A crystal that is running at like 128x48 kHz or somewhere around there. It is clocked from the same clock. To the crystal there's only 2 sample rates: multiples of 44.1 and multiples of 48.

[eidenk]

Jitter may cause a sample to be misinterpreted and cause pops and crackle.

I'm not giving in on that.

Jitter is minute variations (at the nanosecond or picosecond scale) of the audio card clock rate, which means samples are not played back or recorded perfectly on time and the lowest the jitter the better the clarity/definition of the sound. It doesn't generate pop or crackles. Pop/crackles are due to buffer underruns when, at low latency settings, the cpu gets monopolized by an hardware interrupt and the sound card buffers are emptied before the cpu is freed and can fill them up again.

[eidenk]

I think at this point the vast majority of the plugins I use are already internally oversampled.

Well perhaps but this means that two samplerate conversions per plugin are taking place in your audio processing chain and this isn't too good IMO as each of those conversions use extra CPU cycles in addition to degradate the sound quality because they are lossy.

[@midnight]

everytime you run a soft synth at a higher sampling rate, it will sound better

everytime you run any analog modeled plugin that has any nonlinearities (and that is most "analog modeled" plugins since like 2003) at a higher sampling rate, it will sound better

its been proven by the scientists already

as soon as you provide the scientific definition they're using for 'sounds better' i'll believe you. however, i'm pretty sure there's no actual scientific correlation between the conflation of 'higher frequency content' and 'sounds better' that you're making here.

as for 'analog modelled', im not entirely sure what aspect of that you're referring to, but if you're talking about higher modulation rates etc, then i'd be interested in comments from the respective DSP gurus round here as to whether higher sampling rates actually provide significant improvements in accuracy, result, comparison to oversampling the respective stages in the signal chain. and im aware that you've said that you consider oversampling has issues because of the downsampling stage, but Im also not yet convinced that this is results in provably 'worse' (ie not just a provably more band-limited) sound than a higher sample rate. comments from the respective DSP gurus would be welcome.

As for nonlinearities, I'd wonder if, for frequency content above the Nyquist limit for 44.1Khz/48Khz sample rates, most normal monitors and rooms contribute enough audible nonlinearities to render the point moot for most human's hearing.

In short, Im glad you're happy with your choice. It would be nice if you tempered the presentation of yourself as the only one in the room smart enough to have 'discovered' a new silver bullet, though.

dude. heh.

load up a fabfilter synth. Go to the highest octave and play sawtooth wave, pitch bend up and down.

Do this at 44.1khz

Then repeat it at 96khz

are you having a problem with the concept of scientific proof?

Anybody else want to chime in here? I am having trouble understanding what Whyterabbyt is getting at.

[Mushy Mushy]

Anybody else want to chime in here? I am having trouble understanding what Whyterabbyt is getting at.

http://en.wikipedia.org/wiki/Opinion

vs

http://en.wikipedia.org/wiki/Fact

as substantiated by

http://en.wikipedia.org/wiki/Evidence

[@midnight]

Whyterabbit, FWIW, i am mostly referring to the huge "Lets test plugins" thread on Gearslutz, where they tested a lot of plugins and found the aliasing to be far reduced at the higher sampling rates.

That is what I mean by "sounds better"

Basically the character of the tone is the same, but you get less aliasing artifacts and usually a more open sound, that is why I use the words "sounds better"

[G.hostLA]

Frequencies above nyquist(half the samplerate) are reflected back. 30khz would read 10khz at 40khzSampleRate.
So increasing the sampleRate can help keep a mix sounding clean.

SoftSynths and recordings can have un-needed high frequencies, a true saw on a synth(which isn't band limited to the nyquist frequency) will produce these frequencies reflected back.
double the frequency range will mean that one can cut frequencies above ~20k and produce a cleaner sound.

Samples at differing sampleRates may have some troubles but I'm sure most DAWs won't screw your sounds up too much.

[whyterabbyt]

Whyterabbit, FWIW, i am mostly referring to the huge "Lets test plugins" thread on Gearslutz, where they tested a lot of plugins and found the aliasing to be far reduced at the higher sampling rates.

That is what I mean by "sounds better"

Basically the character of the tone is the same, but you get less aliasing artifacts and usually a more open sound, that is why I use the words "sounds better"

So in other words instead of "scientists" we actually have "people on a forum", instead of "proved" we have "measured" and instead of "always sounds better" we have "had less aliasing".

ok. that kinda speaks for itself. the less aliasing thing would have been kind of obvious though, Nyquist and all that. proper ABX testing to determine whether the aliasing artefacts presents are perceptable and/or affect a subjective quality assessment of the sound would be the kind of direction you'd actually want to go in to start proving that kind of hypothesis, though. scientific proof has to be kinda rigorous that way...

[Shabdahbriah]

As of today I am now in the land of 96khz. Already everything is sounding more open and airy, crisp analog round fat gooey goodness.

See: Placebo Response.

Lol! You guys are probably stuck at 44.1 and have no idea what you are missing.

Sorry... having skimmed this thread a bit, I decided to start at the beginning, then only got this (that ^^^ up there) far into it, before:

Seriously?!!!



A lot of very good (and accurate) commentary here, that clearly illustrates otherwise.