Anyone else find this disturbing marketing jargon?

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So wouldn't this phenomenon require a sample rate about ten times higher in order to truly acurately reproduce the high frequency information up to 20khz? because you could get this same result with a signal well below the sample rate, at any frequency really.
 
cusebassman said:
Are you asking what happens when a wave greater the max determined by the samplerate gets analyzed?

If that's the question, you get something called "aliasing", where the set of samples taken gives you a much lower frequency when converted back to analog:
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brsanko said:
So wouldn't this phenomenon require a sample rate about ten times higher in order to truly acurately reproduce the high frequency information up to 20khz? because you could get this same result with a signal well below the sample rate, at any frequency really.
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That's why ADC's use anti-aliasing filters, so that frequencies higher than the sampling rate are removed, to avoid introducing their lower-frequency aliases into the signal.

http://www.maxim-ic.com/glossary/index.cfm/Ac/V/id/358/Tm/Anti-Aliasing
 
Shadow_7 said:
In brass playing there's some upper harmonics in the sound that reinforce each other and become audible when playing in a groups of brasses with good intonation..
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dont want to call total bullshit on you here but for the most part it is... what you're reacting to is your position in the ensemble... not unlike nodal build up in room responces... though the harmonics are there all the time whether they reinforce or not is relitive to the recievers position...whether that be mic/audience/or performer...


also as to the extended frq response... one reason it can be important in the analog stages is that there's some phase shift as we approach the limit of the circuit... so if it's good to 100K there's no phase shift in audible band...
 
brsanko said:
Does anyone understand what I'm asking?
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Yes. Your perception is that each second of music is divided up into 44,100 segments and sampled. You are asking if that is correct, because to you that seems to indicate that unless the information was perfectly timed, a sample rate of 44.1k could easily miss high frequency information.

Your perception is indeed not correct. That isn't how it works.:) Just the fact that indeed you can capture and play a 20k sine wave with a 44.1k sample rate regardless of phase should tell you that.
 
Ah..Yeah I'm gonna hafta read that again when I'm not hung over. My head hurts. But hey thanks guys this is just the info I've been looking for. I don't have too many friends that would have any clue what I was talking about if I asked about sample rates, and aliasing you guys are a great knowledge base.
 
Might I shed some light on the matter? :) I agree that for the most part, it's marketing BS and even if it's true, it's useless to musicians. However, in the field of creative sound design, extreme processing, like pitching something down over an octave, is not uncommon at all. If you pitch a sound down an octave, and it was samples at 44.1 khz, you are going to end up with a product that has absolutely nothing about 10khz. Another octave down and you are stuck with an upper limit of 5khz. That you are going to hear. Pitching a 96khz source down is still going to give you some high end (be it, a high end you would normally not hear, but still). when tuning down an octave (or slightly more).

As a sound designer, I'm very happy with 96khz, for some uses. My main sampling rate is 44.1khz though.
 
My console is rated + or - 3 all the way out to 300k. Interestingly, a reviewer in mix did a review a few years ago on another D&R console. He didn't beleive the D&R specs. He did his own testing and found that D&R had actually underrepresented what the real specs were and that the preamp went well past even 300k within resonable fluctuation
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christ...

now i know that there's little science to support it, but i have a feeling that while we may not be able to "hear" past 20k, we can "sense" frequencies beyond that in a subtle manner; i know that rupert neve has been kicking around the same theory, and i think it might have to do with why many people have such a preference for analog desks and recorders.

but...like so much stuff that gets kicked around these places, it's just speculation...
 
There's really a very simple answer to all this. Screw the marketing hype and TRUST your EARS:D
 
dementedchord said:
dont want to call total bullshit on you here but for the most part it is... what you're reacting to is your position in the ensemble... not unlike nodal build up in room responces... though the harmonics are there all the time whether they reinforce or not is relitive to the recievers position...whether that be mic/audience/or performer...


also as to the extended frq response... one reason it can be important in the analog stages is that there's some phase shift as we approach the limit of the circuit... so if it's good to 100K there's no phase shift in audible band...
Click to expand...

There are difference tones when the harmonics from multiple horns interact, which are quite audible (guitarists often use the same trick for tuning, generating beat tones between equivalent harmonics). So, not bullshit.

The stuff about analog phase shift is important to the DIGITAL conversation. Remember, anti-aliasing filters are analog circuits. Brickwall filters have all sorts of weird phase issues. Pushing the upper frequency for anti-aliasing higher means the pre-DAC filtering has less influence on audible octaves. So yes, 96khz helps, but it's due to analog issues, not digital.
 
Halion said:
Might I shed some light on the matter? :) I agree that for the most part, it's marketing BS and even if it's true, it's useless to musicians. However, in the field of creative sound design, extreme processing, like pitching something down over an octave, is not uncommon at all. If you pitch a sound down an octave, and it was samples at 44.1 khz, you are going to end up with a product that has absolutely nothing about 10khz. Another octave down and you are stuck with an upper limit of 5khz. That you are going to hear. Pitching a 96khz source down is still going to give you some high end (be it, a high end you would normally not hear, but still). when tuning down an octave (or slightly more).

As a sound designer, I'm very happy with 96khz, for some uses. My main sampling rate is 44.1khz though.
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Beautiful, amazing, incredible post. This is the only real, tangible, rational explanation for using a sampling rate over 44.1 that I have EVER heard. Hat's off to you, sir.

In fact, this makes so much sense that I am actually considering what it would take to start tracking in 96 - since 8 out of 10 singers that come through here to record are not very good at all, and end up in Melodyne being shifted all around ("I don't even remember singing it that way! I'M AWESOME!"). I can actually see how this will help with issues that arise sometimes in Melodyne.

Thanks a million Halion, I think you just made my job a LOT easier!

I guess I need to figure out how this 'reputation points' stuff works now....because I want to give you a zillion points on this post.
 
kludge said:
There are difference tones when the harmonics from multiple horns interact, which are quite audible (guitarists often use the same trick for tuning, generating beat tones between equivalent harmonics). So, not bullshit.

The stuff about analog phase shift is important to the DIGITAL conversation. Remember, anti-aliasing filters are analog circuits. Brickwall filters have all sorts of weird phase issues. Pushing the upper frequency for anti-aliasing higher means the pre-DAC filtering has less influence on audible octaves. So yes, 96khz helps, but it's due to analog issues, not digital.
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Modern digital convertors do not have phase issues in their antialiasing filters (which are not analog brickwall filters, as you describe).

Read this:

http://www.audioholics.com/educatio...ick-wall-digital-filters-and-phase-deviations

And all of these whitepapers (especially on sampling theory and FIR):

http://www.lavryengineering.com/index_html.html


Also, the resultant created from intermodulation from two higher tones is smaller amplitude, usually much much much smaller. Thus, as energy tends to decrease with increasing frequency, you are talking about capturing a very low amplitude ultrasonic signal for the purpose of creating an even smaller audio frequency resultant, which, if loud enough to be audible, would be captured by the audio-band microphones anyway.

That's not necessarily true of synthesis, but again you would still be relying on the ultrasonic capability of the listener's playback system to recreate that effect. A lot of gear, especially higher-end stuff, does have ultrasonic response--but at what THD figure? It's not too hard for a preamp to manage 100kHz--most would, unless it was purposely filtered out--but even high-end tweeters that reproduce ultrasonic frequencies may do so at rather high distortion, and tremendously varying frequency response.

So even a higher sample rate will lend no assurance of accurately reproducing any such effect.
 
It is marketing info to hype the product. It is probably true, but irrelevant to most who will use the product. Unless, as stated above, you are seriously processing signals, it will be of little benefit. The major production studios that need that kind of ability are using gear that greatly exceeds those specs, only to record signals through mics with frequency responses of 20 to 20khz. Amazing, isn't it?
 
solo2racr said:
There's really a very simple answer to all this. Screw the marketing hype and TRUST your EARS:D
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It's a good thing I don't know what all those fancy specs mean or I'd be more confused than I am already.:rolleyes:

If all we needed were specs to make equipment decisions life would be a lot easier,but it seems that in a lot of cases specs are there to confuse rather than help the prospective buyer.
 
Even the old standby 20 - 20k spec is just marketing buzz as far as how the average consumer understands it... or doesn't understand it. The specs are a good starting point, but in the final analysis you have to listen to it and like or not.

Me talking to a music store salesman in 1980…

Me: “So what do you think of this new Yamaha whatchamathingy?”

Salesman: “Well, It’s 20 - 20k.”

Me: “Cool”

Salesman: High five! (Actually it was still “Gimme five” back then, but who’s counting)

I could list a thousand examples, but one that comes to mind… IMO the old Lexicon LXP-1 sounds better than the newer MX300 for reverb and delay. The 16-bit LXP-1 has an upper wet frequency response of 15k and a sampling rate of 31.25kHz. The MX300 is 24-bit, 48kkHz, 10Hz to 22kHz ±0.5 dB @ 48kHz, and no audible harmonic distortion. The MX300 looks better on paper, but the LXP-1 is subjectively more natural sounding to my ears.

Marketeers (intentional spelling) often place too much emphasis on the limited measurements collectively known as “specs.”

But anyway, to me having something that will pass 100k just means I need a costly low pass filter. :D
 
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