Anyone else find this disturbing marketing jargon?

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NeveSSL

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I found this on M-Audio's site. Don't get me wrong... not knocking the gear at all... but this REALLY bothers me... because it is completely false (in my opinion)...

"Based on the M-Audio preamp technology that won Pro Audio Review’s highest accolades, the DMP3’s amazing 20Hz to 100kHz frequency response makes it ideal for today’s 96k recording work."

...

You may be able to hear a difference of 44.1k and 96k, but thats because it has nothing to do with frequency response. It has to do with a sampling rate. Nyquist theory comes into play, but really only matters at (in theory...) 40k and below...

Unless, of course, maybe someone is recording bats? :D

Please, don't misunderstand me... I love M-Audio... I use their products. :) I just find that line to be a line of bull...

Am I wrong?

Brandon
 
I'm willing to bet that M-Audio's design engineers didn't write that. Some marketing guy who's tied closely at the hip with their sales manager and bean-counter made that call. The engineer was probably doing one of these :rolleyes: when he saw it for the first time.
 
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. Without decent gear, these never get captured in a recording. Not to imply that it's true or false, but there are elements to ones sound as a group that doesn't always fall into normal human hearing. Wouldn't it be annoying if that high pitched humm that TV's and some lights put out gets captured and reproduced for everything you watch on TV?

Although it is kind of moot if your mic can't / doesn't pick up anything past the 20kHz range. But there is a noticeable difference in sound when using a higher sampling rate for most of my mics.
 
NeveSSL said:
I found this on M-Audio's site. Don't get me wrong... not knocking the gear at all... but this REALLY bothers me... because it is completely false (in my opinion)...

"Based on the M-Audio preamp technology that won Pro Audio Review’s highest accolades, the DMP3’s amazing 20Hz to 100kHz frequency response makes it ideal for today’s 96k recording work."

...

You may be able to hear a difference of 44.1k and 96k, but thats because it has nothing to do with frequency response. It has to do with a sampling rate. Nyquist theory comes into play, but really only matters at (in theory...) 40k and below...

Unless, of course, maybe someone is recording bats? :D

Please, don't misunderstand me... I love M-Audio... I use their products. :) I just find that line to be a line of bull...

Am I wrong?

Brandon
Click to expand...

If I'm not mistaken, a higher sample rate is only useful for recording higher and higher frequencies, per the Nyquist theorem, saying that to record frequency X and reproduce it accurately, you need to sample X*2 (+1?) times per second.

Therefore, it is in fact beneficial to have a preamp that has a higher frequency response, if you are recording at a samplerate that can catch those higher frequencies.

Whether the DMP-3 can actually accurately process frequencies up to 100kHz, I have no idea. But if you are recording at a 96k samplerate, then you are able to record frequencies up to 96,000/2 Hz, or 48kHz. If your preamp is still only able to accept and send out amplified frequencies up to, say, 22kHz and pass them into your converter, it is pointless recording at that higher samplerate since nothing over 22kHz will ever be encountered.

Some of this falls under the cats vs. dogs debate about whether recording at a higher samplerate with preamps that put out higher frequencies actually yields a "better sounding" or more accurate audio reproduction of a source, simply because it is said that the human ear and brain can only process up to somewhere around 16kHz-low 20's kHz. However, the math above (if I am correct on all of it), speaks for itself.

Theoretically, if the DMP-3 can reproduce frequencies up to 100kHz, you would need converters that can sample at 200k/s to get the top functionality of the unit. Of course, what you're recording that produces such a frequency, I don't know :confused: Recording at 96k, you would only need a pre that can reproduce up to 48kHz signals, and I suppose the argument is that instruments, etc. can produce upperharmonics that might go beyond the 22kHz ceiling. Again, back to the debate, do you really need to record them even if they exist, since we can't hear them anyway? You decide :D
 
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. Without decent gear, these never get captured in a recording. Not to imply that it's true or false, but there are elements to ones sound as a group that doesn't always fall into normal human hearing. Wouldn't it be annoying if that high pitched humm that TV's and some lights put out gets captured and reproduced for everything you watch on TV?

Although it is kind of moot if your mic can't / doesn't pick up anything past the 20kHz range. But there is a noticeable difference in sound when using a higher sampling rate for most of my mics.
Click to expand...

Interesting point. I'd love to hear some recordings done at 48k and at 96k to hear a potential difference. Since I usually only record guitars, basses, drums and voices, I've never done a thorough comparison of 44.1k and 96k. Since I can't even tell the difference between 128k and 192k audio compression, I probably wouldn't notice... whether its an untrained ear or its the hearing loss I've suffered from 8 years of marching band, i don't know :p
 
this is hardly a new debate. some prominent industry folks-- engineers, a/d/a converter designers, etc. maintain that anything over 44.1khz is unnecessary for exactly the reasons being cited, others posit that higher sample rates are useful in the digital domain once you start using many layers of plug-in based effects (especially dynamics and applied to the mix buss) that benefit from the additional samples.
 
cusebassman said:
Whether the DMP-3 can actually accurately process frequencies up to 100kHz, I have no idea.
Click to expand...

I agree with the rest of your post wholeheartedly, but the DMP3 doesn't actually "process" audio. They are claiming it passes signal up to 100k. You'd really have to look at the frequency response chart to see how much of a fall off there is that high. It's more helpful if they give a stat like "up to 100k -3db", or something like that. If it is flat up to 100k, then you know you are going to be getting all those frequencies that dogs and bats can hear.
 
It's just marketing. It's everywhere. If you really want to read bullshit, go to the CAD mic site.
http://www.cadmics.com/condensers.htm

GXL3000
Multi-pattern Condenser Microphone
Borne of precious metal and pure electricity. Tune two golden membranes intimately across polished bass and inject electrons with oscillatory high tension. Drive emotion across copper and iron by field-effect detection. Wrap it in Faraday's cage to conceive the GXL3000. Live all three polar patterns as you align space and tone. Features include hi-pass filter and attenuator. Elastic shock mount and protective pouch are included. P48 (48V) phantom power is required.
Click to expand...

But, I think they're going for tongue-in-cheek humor.
 
There have been op amp microchips for 25 years now that can pass a 100kHz bandwdth, from just over DC to 100kHz. I guess if you use any of 'em in your signal chain, you can claim that frequency response whether or not you're actually using all of it. :rolleyes:
 
SonicAlbert said:
I agree with the rest of your post wholeheartedly, but the DMP3 doesn't actually "process" audio. They are claiming it passes signal up to 100k. You'd really have to look at the frequency response chart to see how much of a fall off there is that high. It's more helpful if they give a stat like "up to 100k -3db", or something like that. If it is flat up to 100k, then you know you are going to be getting all those frequencies that dogs and bats can hear.
Click to expand...

Yea, I couldn't think of an appropriate word for what a preamp does to a signal beyond amplifying it - got tired of saying "passes signal", etc. :p Thanks for the clarification!
 
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:)
 
Thanks for the replies, guys.

That CAD Microphone stuff was GREAT! :) I believe tongue-in-cheek, too, but its all good. :)

So I guess it could potentially make sense, but is still rather pointless and jargon in my opinion. :) And as was mentioned, you would need 192k sampling to really use 100k audio...

Oh well!

:cool:

Brandon
 
Ok here's my question. Say you have two seperate tones at 20khz but they are 90 degrees out of phase with each other wouldn't you need a 40khz sampling rate to capture both of them? and wouldn't both of them be audible (assuming you could hear 20khz) and likewise if you had 5 such tones sightly out of phase or sync with each other wouldn't you need a 100khz sampling rate to capture all of them? It seems to me with a sampling rate of 44.1khz you would only capture the high frequency information that was right in sync with the sampling. I'm probably way off here but I'm just trying to figure out how digital audio works.
 
brsanko said:
I'm probably way off here but I'm just trying to figure out how digital audio works.
Click to expand...
You are... ;)

The frequency referenced in this instance does not relate to the audio frequency of the captured material but is actually the frequency of the sine wave that the captured digital information is stored... The bit rate is the number of sample points captured from this sine wave...

Overly simplified... but I think it states it...
 
brsanko said:
Ok here's my question. Say you have two seperate tones at 20khz but they are 90 degrees out of phase with each other wouldn't you need a 40khz sampling rate to capture both of them? and wouldn't both of them be audible (assuming you could hear 20khz) and likewise if you had 5 such tones sightly out of phase or sync with each other wouldn't you need a 100khz sampling rate to capture all of them? It seems to me with a sampling rate of 44.1khz you would only capture the high frequency information that was right in sync with the sampling. I'm probably way off here but I'm just trying to figure out how digital audio works.
Click to expand...
The combination of multiple waveforms results in a single resultant waveform. The resultant may contain frequency information that is higher than the ear (or electronic capturing devices) can discern, but it is still a single resultant. It's not like the waves have to be processed separately and then combined in the device.

The second simplest waveform is probably the combination of two sine waves. Any combination of waves is interpreted by the ear as a single waveform, and that waveform is merely the sum of all of the waves passing that spot. Here are a few rules about the addition of two sine waves:

* If both have the same frequency and phase, the result is a sine wave of amplitude equal to the sum of the two amplitudes.
* If both have the same frequency and amplitude but are 180 degrees out of phase, the result is zero. Any other combinations of amplitude produce a result of amplitude equal to the difference in the two original amplitudes.
* If both are the same frequency and amplitude but are out of phase a value other 180 degrees, you get a sine wave of amplitude less than the sum of the two and of intermediate phase.
* If the two sine waves are not the same frequency, the result is complex. In fact, the waveform will not be the same for each cycle unless the frequency of one sine wave is an exact multiple of the frequency of the other.

If you explore combinations of more than two sine waves you find that the waveforms become very complex indeed, and depend on the amplitude, frequency and phase of each component. Every stable waveform you discover will be made up of sine waves with frequencies that are some whole number multiple of the frequency of the composite wave. (AKA Fourier analysis.)
Click to expand...

http://arts.ucsc.edu/ems/music/tech_background/TE-04/teces_04.html
 
The combining of different frequencies at different degrees of phase relationship does affect the output amplitude but not the resulting frequency...
 
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. Without decent gear, these never get captured in a recording. Not to imply that it's true or false, but there are elements to ones sound as a group that doesn't always fall into normal human hearing. Wouldn't it be annoying if that high pitched humm that TV's and some lights put out gets captured and reproduced for everything you watch on TV?

Although it is kind of moot if your mic can't / doesn't pick up anything past the 20kHz range. But there is a noticeable difference in sound when using a higher sampling rate for most of my mics.
Click to expand...

by "upper hamonics" do you mean below or above 20k. Human beings can't typically hear above 20khz, regardless of how much these sounds are "reinforced".

You could make an argument that frequencies very close to each-other and above 20k could create an audible beat-frequency...but then the frequency content closer to the fundamental will likely mask the beat frequency (simply due to amplitude differences).

It would be interesting to find out if such interferences would even be above the threshold of hearing...very interesting.
 
Well I guess the real question is can digital audio capture high frequency signals that are out of phase less than 180 degrees with the sample clock? I understand that you get a sample 44100 times a second but what if you have a signal that peaks between those samples would it just be recorded quieter than it actuallyis or would it be missed all together? Does anyone understand what I'm asking?
 
brsanko said:
Well I guess the real question is can digital audio capture high frequency signals that are out of phase less than 180 degrees with the sample clock? I understand that you get a sample 44100 times a second but what if you have a signal that peaks between those samples would it just be recorded quieter than it actuallyis or would it be missed all together? Does anyone understand what I'm asking?
Click to expand...

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:

http://en.wikipedia.org/wiki/Image:AliasingSines.png

As you can see in that picture, the reconstructed wave looks significantly lower in frequency because not enough samples were taken of the original waveform.

If this wasn't your question, I apologise - but its pretty cool stuff anyway :)
 
The link just has a picture, no article. But yeah I think that is what I was talking about.
 
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