so literal.... yes, i know they are dual. Ok should have said 747.
I have not used the miniDSp at all after changing opamps. Ordered another brand with better over-all performance. I hope.
THx-RNMarsh
Thanks!
Ok so the "fish" would never "emerge" if you run the signal in slow-mo and watched a level meter follow the *envelope*! The envelope is that path you would travel if you "walked" the signal - right?
Hmm, still wondering... when you use waveform and envelope in the same sentence - must mean they are different for you - how?
//
All of the above and color, too.
And, the slope characteristics, I wanted more flexibility with individual slopes.
-RM
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Yes, thereby possibly creating IM distortion components in the sub 18Khz range.
Hans
If you feed speakers with twin or more tone frequencies >22kHz, they will create IM products in audio band. This is the "ultrasound hearing miracle".
Given you can program your own biquads what is missing from flexibility? If you really MUST have FIR then some miniDSP products have that as well. I'd be interested to know what you think you need that is missing.
Perhaps you could get some funding for some cutting edge research? Think of the emotional pleasure it could bring......maybe. Are you prepared for Jacob's Wrath - Trailer on Vimeo
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That's a pretty cool method of explanation. Let's see, if you feed amplifiers with twin or more tone frequencies >22kHz, they will create IM products in the audio band. This is the "amplifier sound difference miracle".
Yep, works and I like it.
Send me your "audible difference miracle" that you would like to have resolved and I'll do it. It most likely will only take one to three sentences ....😎
@scott wurcer;
Fourier does, but the usual FFT result graphs do not and a lot of people looking at the images (and try to interpret) tend to forget about the phase information.
The problem is, my dear Jakob, that I can easily demonstrate that it is the speaker that creates IM. The amp IM is about 50dB better than that of the speaker.
2 plots, one with L channel playing 22kHz and R channel 27kHz. This shows that microphone is clean and does not create distortion.
2nd plot, one speaker plays 22+27kHz tones and we can see IMD products in audio band.
Dear Jakob, this is the issue that Oohashi did not take into account. He should have used 2 speakers instead of one, one for ultrasound, one for <20kHz programme.
You know I do not need weeks of discussions to prove somebody wrong.
Attachments
The audibility is questionable. Yes in case of quiet environment and not too far from speakers. Probably not if masked by music programme.
Those levels will be quit audible with music... rarely will there be only 2 freqs. A lot of HF freq are there. From imperfect LPF from CD player.
I have said this already here recently and other forums... it causes IM. But only if you have a VFA with falling open loop gain such that it become more and more non-linear amp in region >22Khz. many CfA/CMA are linear into HF and beyond and sound cleaner in comparison.
Alternately, a LPF on VFA PA input gives similar sounding performance then.
THx-RNMarsh
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That LPF depends on the DAC.
This filter only blocks everything above Fs/2, so when using a DAC that converts all input to a higher frequency, the filter will also be wider Accordingly.
So when your input file is 96 or 192, many ultrasonics will pass unfiltered.
Forget the amp being used.
IM from the speakers will be much higher as PMA already mentioned.
Hans
The units of angles and radians are unimportant to the argument. The identity doesn't care. All the user has to do is scale properly, 180/pi conversion.
The identity is the key for others to understand. It details why, when I input sin(20k)*sin(2.5k), the filter sees and removes 22.5k.
It is simple AM radio stuff.
Jn
Too much of a blanket.
This story contains two parts. First is the filter before the sampler. It prevents violating nyquist.
The math algorithm is a simple one which by definition, produced two pure sines.
The filter removes one. All this has nothing to do with sample rate. The filter sample rate needs to be high enough to NOT remove pertinent envelope information. If the filter worked at a gig rate but removes 22.5k, the same thing will happen.
Your I/O difference Gibbs plot shows exactly where the filter removes content, the Gibbs envelope might be used to determine if it's fundamental frequency content being removed (which we do not care about), or envelope created sidebands.
Jn
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Thats a good point about BW of LPF. All the more reason to have linear wideband PA.
Regarding speaker interfacing issues; being a separate subject with separate solutions. All - not one or the other - causes need to be addressed.
THx-RNMarsh
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