Any use of going beyond 44100?

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wtf? are you lot all deaf or what?
No. As mentioned several times already...the end result is going to be at 44.1/16 so quite a few of us see no reason for soaking up HD space with ridiculously long files and flogging a PC to death for no audible gain at the end mix. I don't dispute that some VSTi might sound marginally better at higher rates, but it's all going to be truncated and reduced at the end.

And I have another very practical reason for not using high sample rates. Since I went up to 24bit, I find that one complete song generally takes up a complete CD for storage. Most of my tracks are around 6-7 minutes each - I also do alot of rendering to audio rather than running everything live. The last 4 or 5 projects have all consumed between 600-700MB each. If I then go to 88.2KHz I'm going to need to use 2CDs to store each project - what's the point in that, when I can't hear the difference on the final mix. Just because a facility is available, it doesn't mean you have to use it. In the same way I could get latency on my soundcard down to under 1mS probably - but why would I need to? It would cripple my system.

And personally I think aliasing is one of those things that some people go on about for no valid reason. When was the last time any of you regularly played VSTi riffs at the top of your keyboard at G7 or G8....unless you do, then aliasing is really not going to be an issue. Even with harmonic enhancers, most harmonics are still going to be within the 22KHz limit.

Not that aliasing can't be a problem...but the majority of musos into electronica just won't be playing high enough for it to bother them, and the average patches they use aren't bright enough to alias. :roll:

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edit: double post.

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I have a Scope Board and I find the Creamware Vectron sounds much smoother at 96KHz. Some of the other Creamware synths sound smoother and silkier and slightly less rough. However, they could sound too CLEAN ???

I think once the bog standard CPU is at 5GHz and we had tons of storage we will all be working at 96 KHz and having the same discussion about 192 KHz -- although I can't run my Scope board at 96 KHz as the DSPs get overused. I like my 44.1 KHz - it sound good enough for me.

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while i can beleive their is little point to recording at 96 khz and downsampling to 44.1/16 bit their is a clear and easily heard difference between some audio recorded at 96kHz and that recorded at 44.1

i own a handfull of SACD/DVD-A's most of which are cds i already owned - they sound better - not even a hint of maybe - 100 % confident of this

Now i will concede that my stereo at 4 grand or so is not an i-pod but the difference is obvious - and its not that i can hear frequencies for dogs - its better seperation of ALL instruments, much clearer soundstaging and a better sense of space - and most importantly high frequency instruments (horns, voices) sound scarily realistic

So if you plan to record for these formats (and more and more people do) you will benefit from recording at 96 Khz - how this affects vst is dependant on how they are programmed

PS - for all the join the dots haters - explain how a 44.1 hz system differentiates between a 20 Khz sine wave and a 20 khz sqaure wave - both types have one negative and one positive point

the answer is they don't - any postfiltering when converting to analogue will make the signal sound like a sine wave whether it was one or not - at SACD 2400(!) khz sampling frequency i think you might just be able tell them apart

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ericj23 wrote: PS - for all the join the dots haters - explain how a 44.1 hz system differentiates between a 20 Khz sine wave and a 20 khz sqaure wave - both types have one negative and one positive point
Well, talking about a 22040hz or so sine vs square in theory, they will both come out as sines because what makes a square wave a square wave is that it contains frequencies above the fundamental. In fact you can construct this square wave with a 22040hz sine wave combined with an infinite (in theory obviously) number of additional sines of successively higher frequency and lesser amplitude. If you put a square wave into an additive synth (eg. chameleon and whatnot), this is exactly what it will do in fact (using however many sines it can generate).

In other words, ideally there would be no difference because there is not supposed to be, at 44.1KHz frequencies from 22050 on up cannot be accurately reproduced. You've just restated that 96KHz can reproduce higher frequencies than 44.1KHz, which I don't think anyone questions.
at SACD 2400(!) khz sampling frequency i think you might just be able tell them apart
Assuming you can hear them in the first place and further that your speakers/headphones can reproduce them with any accuracy.

But as I said in my first post in this thread, I don't doubt that higher sampling rates, up to a point, sound better. In fact, everything I have read leads to me to believe that is the case.
Last edited by contrast on Fri Dec 31, 2004 2:37 am, edited 1 time in total.

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And they're not 16 bit so you're not just comparing differing sample rates.
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kritikon wrote:I don't dispute that some VSTi might sound marginally better at higher rates, but it's all going to be truncated and reduced at the end.
Some VSTi sound quite a bit better at higher bitrates, and the differences fall well within the range of spectrum that will remain after downsampling. This is why some devs implement 2x oversampling within the instrument itself.

You could just avoid using any such synths but I've found that quite a few of them fall into this category.

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firstly the notion that both a square wave and a sine wave at 22.1khz are both sine waves are both correct if you use the everything can be made out if sines model - alas in the real world it is totally irrelevant becuase the DAC in your audio equipment does not use such an approach - now it may be oversampled and use all sorts of filtering but the bottom line is it cannot 100% accurately reproduce the wave forms put into the upper frequency range becuase it does not have enough data to follow the output - which lets face it is a constantly eveolving stream of wave data mixed in

and lets see my cd has a linear response to 40 khz - as does my amp - and my speakers -3db is at 25 khz - with -6db up at 50 khz

i can hear it Ok thanks

my point wasn't to say that recording vsts at higher rates has particular benefits but that recording audio at higher frequencies does - and once half your project is at a higher frequency then......

but i defintely dont have the knowledge to assert that the difference once it is downsampled is audible - i'm not even that convinced that recorindg at 24 bit and then truncating to 16 for cd is particulalry audible

what i am convinced about is that your dac plays join the dots - it may check a few dots ahead to make sure it is not producing nonsense but if there arent a whole lot of dots (ie at high frequencies ) it loses information

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If half your song is above 10,025 Hz, then I have serious doubts about your abilities as a composer. ;)
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ericj23 wrote:while i can beleive their is little point to recording at 96 khz and downsampling to 44.1/16 bit their is a clear and easily heard difference between some audio recorded at 96kHz and that recorded at 44.1
No there isn't. Only if the ADC or DAC works bad at 44.1KHz.
ericj23 wrote:PS - for all the join the dots haters - explain how a 44.1 hz system differentiates between a 20 Khz sine wave and a 20 khz sqaure wave - both types have one negative and one positive point

the answer is they don't - any postfiltering when converting to analogue will make the signal sound like a sine wave whether it was one or not - at SACD 2400(!) khz sampling frequency i think you might just be able tell them apart
Again, wrong, a square wave at 20KHz will sound exactly like a sine wave to any normal human. To even hear any difference at all, you'd have to be able to hear 40KHz signals. And you don't.

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The fact that SACD or DVDA sounds better is subjective - If I were the mastering engineer, I bet my orders would have been to "sweeten up" the mixes on those formats or else no one will buy them. Does that mean the mix is closer to the original? Another variable is that when someone does hear a difference at other than 44.1 - it may be that their converters are performing differently, not necesarily more accurately.

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I thought not to comment on this thread further because I feel I can only loose... but I can't help.


(1)
The fact that some of the SACD/DVD-recordings sound subjectively better may be due to anything besides the sample rate. This includes bit depth, sound engineering, other equipment used, better DACs, or just flatly, different EQ used during recording...

The fact is, a 96 kHz recording does not contain any information relevant to your listening experience, that wouldn't have been reproducible with 44.1.

(2)
There is no need to distinguish between different waveforms at pitches >9 kHz. As the pitch of a tone goes higher, fewer and fewer harmonics are within audible range. Thus, the higher the pitch, the less you can tell different waveforms. All tones >9 kHz should be sines to the ear, because only the fundamental is within audible range.

(3)
Again, synths and VSTi sounding smoother at 96 kHz is due to programming artifacts. They should have sounded as smooth at 44.1 in the first place. (In an ideal world). When this is not possible, a VSTi should use internal oversampling, or give the user the choice of doing so, to avoid having to hog the system with a global 96 kHz sample rate.

Generally, this is where you can tell the cheap synths and samplers with crappy wavetable interpolation and filters from the good ones.

Also again, try alias-free synths that use the BLIT or MinBLEP algorithm, which need no oversampling at all, and should sound the same no matter what sample rate used (as long as it's >44.1)

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... and happy new year to every1. :)

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I thought I killed this horse a while ago ;)

There are at least three measurable aspects of an audio recording that may be compromised by an inadequate signal chain. These are (1) amplitude irregularities (2) phase anomalies and (3) intermodulation distortion (and this was often completely ignored in the old days). Sadly, not much has changed for the masses …

If we restrict our discussion to steady-state, single sine wave recording and playback then the first two aspects are all that matter – in this case it is possible to provide a nearly perfect rendition of an extremely boring tune - even at 44.1K sampling rate ;)

The moment more than one sine wave is recorded the fun starts and intermodulation distortion rears its ugly head. Anytime two tones sound together there are two side bands created – the sum and difference. In air these new frequencies don’t do anything really bad – in electronic circuits they are amplified non-linearly and can be quite audible by some people (and measurable by anyone with the right equipment).

I have no quarrel with people who want to quote chapter and verse of the Nyquist-Shannon sampling theory. But it is just a theory and as I indicated here and here there are limits to what can be done in the real world.

Anti-aliasing filters that have to deal with incredibly small transition bands are very hard to get really close to "perfect". They suffer in all three of the above areas.

The truth: higher sampling rates provide a benefit at a cost. If you can afford the cost (in higher CPU usage) it is well worth making the jump to at least 48K.

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Unless in case of hard low passed sounds, there's * always * a variable amount of aliasing, with higher sampling rates you allow those uninvited guests to unfold into non hearing regions then you can get rid of the hay when downsampling, there's a variable mix of dull white noise and frequency patterns in this aliasing so it's not pure hay.
I became so picky that at many many times it's 96K or bust because of CPU/DSP drained ! :cry:
In some cases though, I prefer 44.1 or lower so just test BUT in the long term, in many synths * and * presets, a pattern should emerge normally ! :wink:
Just play with the aliasing or not ...

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