That's because what CoolEdit is showing you between the "dots" is a hermite interpolation. Sample rate conversion, when properly coded, smoothly interpolates rather than just dropping samples or repeating the values to fill in.Christian Schüler wrote: Audition/Cooledit has is very good demonstration of this effect. You can zoom into the wave until you can see the sample points, plus you can see the corresponding continuous wave resulting from the sample points (be surprised!).
If you want the ultimate kick, try converting sample rates to 96 kHz or even down to 22 kHz and observe that the continuous wave hasn't changed shape(as long as no signal energy was lost).
Any use of going beyond 44100?
- KVRAF
- 8088 posts since 9 Jan, 2003 from Saint Louis MO
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fishbowl.tucson.az fishbowl.tucson.az https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=45536
- KVRist
- 415 posts since 23 Oct, 2004
If your only intention is to work in a processor-friendly "draft mode", why stop at 44.1k? For draft work, it's going to be tough to even distinguish it from 22050, half again the processing. Yeah, I know, we're getting into audible frequencies now, even for 40 year olds. I prefer to lock everything in at 48kHZ, and I'm far more concerned with dynamic range than with frequency response, but that's just old-school, classical composin' me.contrast wrote:Work at 44.1k to keep the load on the computer to a minimum, bump it to 96k and render to get the best sound out of the plugs.UvRayz wrote:Kuniklo: Could you elaborate on the logic behind that procedure?
Why 48kHz and not 44.1? I have a master clock that works at 48kHz, and I'm very sure that sampling is in sync. And since my recording stuff can handle twice that rate, I like to delude myself into beleiving it's in a midpoint of some margin of headroom. In other words, what it can do in a flat out run, it ought to be able to do half that in its sleep. I certainly think this is true for the bottleneck where the bits have to go through the disk controller's buffer, and anything at all that has to do a bitwise operation on the sample.
48K is a nice happy medium, and happens to be an even number, which appeals to my sense of symmetry.
Now, as to whether 24 bits of dynamic range are better than 16, I definitely say yes, and politely allow anyone who disagrees to do it their way. Most people seem to be struggling to keep everything in the top 4 bits anyway.
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- KVRist
- 487 posts since 19 Jan, 2003
I actually think it's best to work and render at the same sample rate. I was just explaining the thought process behind not doing so since someone asked.
I would definitely work at 24 bits if at all possible, not much reason not to unless your gear just doesn't support it.
I would definitely work at 24 bits if at all possible, not much reason not to unless your gear just doesn't support it.
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- KVRist
- 86 posts since 11 May, 2003
22050 means a maximum frequency limit of 11025hz, which is below maximum frequency perception limit. This will make a big difference in many cases (human perception goes up to about 17-18khz). Imagine not being able to turn the cutoff knob of a filter above 11025hz, it would only allow for pretty dull sounds...why stop at 44.1k? For draft work, it's going to be tough to even distinguish it from 22050, half again the processing.
- KVRist
- 352 posts since 31 Mar, 2002 from Europe
Well I haven't read the complete thread, but I hopefully can add something new and share my limited knowledge...
In most cases it's sufficient if the Synth is doing oversampling, working with n * sampling rate, internaly. The advantage is, that the calculated overtones above the audiable frequency can be filtered before downsampling to 44.1khz. If you don't filter them, even overtones in non audiable frequencies can create audiable aliasiang when inclompletely calculated.
I hope this brings a little light into things...
Summa
In most cases it's sufficient if the Synth is doing oversampling, working with n * sampling rate, internaly. The advantage is, that the calculated overtones above the audiable frequency can be filtered before downsampling to 44.1khz. If you don't filter them, even overtones in non audiable frequencies can create audiable aliasiang when inclompletely calculated.
I hope this brings a little light into things...
Summa
A formant is a preferred resonating frequency of any acoustical system.
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- KVRian
- 911 posts since 1 Dec, 2003 from tejas
As if the issue wasn't convoluted enough I would like to offer this ...
For a given sample rate there is a specific number of sample points that must be used to produce a sinc interpolated 2X over-sample. This number is a function of the required flatness and stop-band attenuation. For an attenuation of -126.42dB and a response dip of no greater than 0.1 db here is some interesting data:
@44.1K 60 pts -0.09dB 2.646 MFlops
@48K 34 pts -0.08dB 1.632 MFlops
@96K 12 pts -0.014dB 1.152 MFlops
It is interesting that the number of operations for the 96K case is actually less than half that required for the 44.1K case.
For a given sample rate there is a specific number of sample points that must be used to produce a sinc interpolated 2X over-sample. This number is a function of the required flatness and stop-band attenuation. For an attenuation of -126.42dB and a response dip of no greater than 0.1 db here is some interesting data:
@44.1K 60 pts -0.09dB 2.646 MFlops
@48K 34 pts -0.08dB 1.632 MFlops
@96K 12 pts -0.014dB 1.152 MFlops
It is interesting that the number of operations for the 96K case is actually less than half that required for the 44.1K case.
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- KVRist
- 111 posts since 18 Dec, 2004
Thanks. If I understood correctly, you suggest that if a synth is well crafted, there shouldn't be much advantages in having project samplerate above 44.1 kHz.
Summa wrote:Well I haven't read the complete thread, but I hopefully can add something new and share my limited knowledge...
In most cases it's sufficient if the Synth is doing oversampling, working with n * sampling rate, internaly. The advantage is, that the calculated overtones above the audiable frequency can be filtered before downsampling to 44.1khz. If you don't filter them, even overtones in non audiable frequencies can create audiable aliasiang when inclompletely calculated.
I hope this brings a little light into things...
Summa
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- KVRAF
- 12977 posts since 29 Sep, 2003 from Ottawa, Canada
I dunno. I guess it depends on what you're used to and what equipment you're using, but I'm a pretty much 'normal' dude with a normal stereo, like the majority of the population. I could easily run a low-pass filter at even 9,000 kHz and usually not notice the difference.AntiPro wrote:22050 means a maximum frequency limit of 11025hz, which is below maximum frequency perception limit. This will make a big difference in many cases (human perception goes up to about 17-18khz). Imagine not being able to turn the cutoff knob of a filter above 11025hz, it would only allow for pretty dull sounds...why stop at 44.1k? For draft work, it's going to be tough to even distinguish it from 22050, half again the processing.
Sometimes I'd even CHOOSE to do so, to get rid of any offensive hissy leftover crap way up there.
Don't get me wrong, I'd never work in 22,050 anyways, just saying that there's a whole lot of frequencies up there that I could do without and not feel too much pain.
Greg
- KVRian
- 1202 posts since 8 May, 2003 from Munich
wtf? are you lot all deaf or what? 
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- KVRAF
- 12977 posts since 29 Sep, 2003 from Ottawa, Canada
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- KVRAF
- 13446 posts since 14 Nov, 2000 from Hannover / Germany
Even my old ears noticed a difference in some cases between 44.1 and 96 - but I hardly ever noticed much of a difference between 44.1 and 48.
So, for me, working at 96 *would* make sense if (my) computers were faster.
48 would *perhaps* make sense too on rare occasions, but the hassle to convert each and every f**king sample coming from a CD or so from 44.1 to 48 is just annoying me too much for the little outcome I'd gain from this. Let alone that around 90% of my drumsamples are in 44.1, so if I was working at 48, no preview (using DR-008, Battery or the EXS) would sound correct (they all only convert the samples to the project's sample rate once you actually load them).
So I just stick with 24bit/44.1kHz.
So, for me, working at 96 *would* make sense if (my) computers were faster.
48 would *perhaps* make sense too on rare occasions, but the hassle to convert each and every f**king sample coming from a CD or so from 44.1 to 48 is just annoying me too much for the little outcome I'd gain from this. Let alone that around 90% of my drumsamples are in 44.1, so if I was working at 48, no preview (using DR-008, Battery or the EXS) would sound correct (they all only convert the samples to the project's sample rate once you actually load them).
So I just stick with 24bit/44.1kHz.
There are 3 kinds of people:
Those who can do maths and those who can't.
Those who can do maths and those who can't.
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fishbowl.tucson.az fishbowl.tucson.az https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=45536
- KVRist
- 415 posts since 23 Oct, 2004
Been playing music for just almost 40 years. Hell I remember when they were still pressing 78's. Granted, it was the tail end of the era, but still...xRAVENx wrote:wtf? are you lot all deaf or what?
- KVRist
- 352 posts since 31 Mar, 2002 from Europe
You can check that with Reaktor, since it can change the engine frequency indenpendently from the cards sampling rate...sc_a wrote:Thanks. If I understood correctly, you suggest that if a synth is well crafted, there shouldn't be much advantages in having project samplerate above 44.1 kHz.
...Summa
A formant is a preferred resonating frequency of any acoustical system.
- KVRist
- 352 posts since 31 Mar, 2002 from Europe
Doubled posting, KVR behaves a bit strange today...
A formant is a preferred resonating frequency of any acoustical system.

