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Joined: Sep 2013
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I finally got my VPC1 last friday and I couldn't be happier; although a stunning instrument, doesn't make me feel nostalgic about the CS7 I had in Belgium. I'll make a thread with pictures and details about the setting in the next few days in case it'll be useful to other members of the blog. In the meanwhile, I'm fooling around with the Kontakt 5 settings and have some doubts about the best "sample rate" for the Galaxy Vintage D. My Macbook Pro (ssd, so it's quite fast) can handle no prob the 64 Samples latency, but I'm not sure if I should keep the Sample Rate to 44100, 48000, 88200 or 96000 ... Any heads up?
Thanx in advance :-)
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Emiliano, congrats on your VPC1 (and move from Belgium to England...).
Regarding your query, I doubt there's a major difference in audio quality between the different sample rates (although latency may be affected). Assuming Galaxy Vintage D utilises 48 kHz sample data, I would set Kontakt's output to match.
Having said that, perhaps Kontakt's effects/reverb processing would benefit slightly from the higher sample rate?
Cheers, James x
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Thank You James, I'm SO excited....I'm here fooling around with "Autumn in New York" as I type :-) I heard that the "Sample Rate" has to match the sample data of the Galaxy, but how do I know? 44 or 48?
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I'm afraid I don't know, although it may be possible to load the samples into a wave editor to check their properties.
Please note that with modern technology it may be fine to use different sample rates for the source wave data and audio output...
Cheers, James x
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Well, there is always the "play and listen" test.
If something sounds better then use that, otherwise ... flip a coin.
Don
Kawai MP7SE, On Stage KS7350 keyboard stand, KRK Classic 5 powered monitors, SennHeiser HD 559 Headphones
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Hi Emiliano
I run the Galaxy Vintage D at 44.1 Khz on my Motu Ultralite Hybrid and sounds just fine, I tried also at 48Khz but I don't hear any sensible improvement so I prefer the lower rate to lessen the stress on the CPU & disk (though I also have an i5 and a SSD so not really a problem)
PS: Galaxy Vintage D samples are 48Khz 24bit
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Well, there is always the "play and listen" test.
If something sounds better then use that, otherwise ... flip a coin. thanx dmd, pianistically I'm not seasoned enough to tell the difference, so I guess it's a bit of a toss up btw the two parameters....
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Hi Emiliano
I run the Galaxy Vintage D at 44.1 Khz on my Motu Ultralite Hybrid and sounds just fine, I tried also at 48Khz but I don't hear any sensible improvement so I prefer the lower rate to lessen the stress on the CPU & disk (though I also have an i5 and a SSD so not really a problem) Max, do you have any idea about the manufacturer's sampling data for the Vintage D? I heard is better to avoid any mismatch btw Kotakt's "sampling rate" and the sound engine's sampling data to avoid the so-called "resampling". Do you have any clue? :-)
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Hi Emiliano
I run the Galaxy Vintage D at 44.1 Khz on my Motu Ultralite Hybrid and sounds just fine, I tried also at 48Khz but I don't hear any sensible improvement so I prefer the lower rate to lessen the stress on the CPU & disk (though I also have an i5 and a SSD so not really a problem) Max, do you have any idea about the manufacturer's sampling data for the Vintage D? I heard is better to avoid any mismatch btw Kotakt's "sampling rate" and the sound engine's sampling data to avoid the so-called "resampling". Do you have any clue? :-) I was able to find the info, 48Khz 24bit (I may reconsider my settings... )
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Nice catch Max, I sifted through the web the whole eve yesterday without success .... you rock!!
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by the way, I just discovered this: keeping the latency lever set to "64 samples" and changing the "sample rate" from 44100, 48000 to the highest 96000, I noticed that there's a bottom right indicator that DECREASES its value from 11.5 ms (with 44100), 11.3 ms (with 48000), all the way to 10.7 ms (with 96000). Is the "ms" value the actual latency? If that's the case it reduces going up with the 'sample rate'.... I need an aspirin :-P
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ms is milliseconds, yes.
If you increase the sample rate, the computer is having to work faster, of course. So, all other things remaining the same, the 'latency' value is going decrease because the whole operation is speeded up.
Therefore, it is a 'win/win' situation, except that, if you make the computer work too hard (by decreasing sample buffer size and/or increasing sample speed (and possibly bit breadth) too much, the signal will begin to crack up.
Roland HP 302 / Samson Graphite 49 / Akai EWI
Reaper / Native Instruments K9 ult / ESQL MOR2 Symph Orchestra & Choirs / Lucato & Parravicini , trumpets & saxes / Garritan CFX lite / Production Voices C7 & Steinway D compact
Focusrite Saffire 24 / W7, i7 4770, 16GB / MXL V67g / Yamaha HS7s / HD598
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Yes of course it reduces by increasing the sample rate, because the number of samples in the buffer is the same (the one you set with the cursor) but each sample duration decreases when the sample rate changes (duration is 1/samplerate): surprised?
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oh boy.....should I set it to 96000 to get the lowest latency then? Better to stick with the 48000?
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PS - total latency could be significantly greater than 10 - 11.5ms because the reading given there may only be dealing with delays inside that particular sampling program - there are other things causing latency like MIDI and the D/A converters.
But if it sounds ok and you can play comfortably with that latency value, probably best forget about it. Otherwise, you can get into latency heck* - especially with the PC, not particularly because it's worse than Apple Mac, but because you can endlessly tinker around with drivers and switching on and off background routine operations, and many other geeky things.
But for this specific case, just try 96k, 48k and 44k, and see if they sound ok in terms of overall quality, on the one hand, or cracking up problems on the other. If you are experiencing a perfect coherent signal with 64 sample buffer, 24 bit and anything over 44k sample rate, then you are pretty well set up! The 96k level may not improve the sound (since the sample seems to be at 48, but you get a slightly lower latency, as you've said).
* unfortunately, the auto moderator doesn't allow the Queen's English so better say 'inferno'
Last edited by toddy; 10/24/14 08:15 AM.
Roland HP 302 / Samson Graphite 49 / Akai EWI
Reaper / Native Instruments K9 ult / ESQL MOR2 Symph Orchestra & Choirs / Lucato & Parravicini , trumpets & saxes / Garritan CFX lite / Production Voices C7 & Steinway D compact
Focusrite Saffire 24 / W7, i7 4770, 16GB / MXL V67g / Yamaha HS7s / HD598
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That 0.6 ms latency difference is completely un-noticeable, and even if it weren't, it corresponds to the time taken for sound to travel 20 cm, so you could cancel it out by moving your head slightly.
It's also physically impossible for humans to hear above 20 kHz, and the maximum sampling rate required to capture such sounds is double that, 40 kHz. Any sampling or playback rate above that is only useful if you are an animal with ultrasonic hearing (e.g. a bat or porpoise), and an audiophile one at that.
So there's no advantage in setting it above 48 kHz. If it's sampled at 48 kHz then setting playback to 44.1 kHz will not affect the audible quality at all, but will require CPU time.
Kawai CA95 / Steinberg UR22 / Sony MDR-7506 / Pianoteq Stage + Grotrian, Bluethner / Galaxy Vintage D / CFX Lite In the loft: Roland FP3 / Tannoy Reveal Active / K&M 18810
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lolatu, I would certainly agreed with everything you said - it is well known that we can't hear above 20kHz, and in the case of adults (especially musicians) probably not much above 8KHz. So 96KHz appears to be totally unnecessary.
But I find it hard to believe they are only making these high rates for fun, or to impress spec-heads. There must be some utility in it - maybe because we are not only duplicating the sounds (as if listening to a CD), but also processing them. I don't know - just guessing.
lolato said:
So there's no advantage in setting it above 48 kHz. Setting it to 44.1 kHz will not affect the audible quality at all, but will require CPU time.
What exactly do you mean here?
Roland HP 302 / Samson Graphite 49 / Akai EWI
Reaper / Native Instruments K9 ult / ESQL MOR2 Symph Orchestra & Choirs / Lucato & Parravicini , trumpets & saxes / Garritan CFX lite / Production Voices C7 & Steinway D compact
Focusrite Saffire 24 / W7, i7 4770, 16GB / MXL V67g / Yamaha HS7s / HD598
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But I find it hard to believe they are only making these high rates for fun, or to impress spec-heads. There must be some utility in it - maybe because we are not only duplicating the sounds (as if listening to a CD), but also processing them. I don't know - just guessing. Hypothesising here, but if you wanted to change the pitch of a sample down an octave or two, then maybe it would be useful. You'd be taking ultrasonic frequencies and making them audible, so it might sound weird. Perhaps there's a niche for recording bat-calls, or scientific / technical applications for recording ultrasound. But for simple recording and playback I can't think of a good reason. There were a bunch of artists like Neil Young pressing for 192 kHz / 24-bit recordings to be made standard recently, but I think that was because they didn't understand / believe that it doesn't actually sound better and is a waste of disk space. So there's no advantage in setting it above 48 kHz. Setting it to 44.1 kHz will not affect the audible quality at all, but will require CPU time.
What exactly do you mean here? Any playback frequency above 40 kHz is "perfect" according to the human ear. You don't lose any audible information by downsampling. However your computer will have to do some work to change the frequency of the digital signal, rather than reading it straight from the disk. It may be that this is only a negligible amount of work, but it's unnecessary so why not stick with 48 kHz.
Kawai CA95 / Steinberg UR22 / Sony MDR-7506 / Pianoteq Stage + Grotrian, Bluethner / Galaxy Vintage D / CFX Lite In the loft: Roland FP3 / Tannoy Reveal Active / K&M 18810
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Adjustment from 48 to 44 - yes, that's what I thought you meant. So in fact the higher rate demands less CPU power in this case? There could be issues to do with aliasing - audio strobe type intrusions, perhaps.
Roland HP 302 / Samson Graphite 49 / Akai EWI
Reaper / Native Instruments K9 ult / ESQL MOR2 Symph Orchestra & Choirs / Lucato & Parravicini , trumpets & saxes / Garritan CFX lite / Production Voices C7 & Steinway D compact
Focusrite Saffire 24 / W7, i7 4770, 16GB / MXL V67g / Yamaha HS7s / HD598
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Adjustment from 48 to 44 - yes, that's what I thought you meant. So in fact the higher rate demands less CPU power in this case? Yes. But maybe this would be cancelled out by extra work needed on reverb and EQ effects at the higher frequency. It probably doesn't really matter too much. There could be issues to do with aliasing - audio strobe type intrusions, perhaps. Are "audio strobe intrusions" a thing? My understanding is that the sample rate being double the highest audible frequency is enough to ensure you can reproduce them without inadvertently hitting the peak / trough every time. But I'm no expert.
Kawai CA95 / Steinberg UR22 / Sony MDR-7506 / Pianoteq Stage + Grotrian, Bluethner / Galaxy Vintage D / CFX Lite In the loft: Roland FP3 / Tannoy Reveal Active / K&M 18810
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