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25th June 2019, 09:54 PM  #1251 
diyAudio Member
Join Date: Nov 2006

Me too (given the right amount of time to make sure I have the funds)

26th June 2019, 02:43 AM  #1252 
diyAudio Member

as the owner of minidsp 2x4HD, I also consider getting this one someday but yeah alot of saving required. looking at the price which is a fraction of XTA

26th June 2019, 02:06 PM  #1253 
diyAudio Member

Sorry for not keeping you all updated as I am currently in the middle of renovating the lab so have been busy with that. For those wanting pricing we will release pricing soon when we post details of the Ultimate Preamp Plus along with the Preorder details. Still finalizing the details so shouldn't be too much longer
cheers david 
6th July 2019, 03:39 AM  #1254 
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

The DSP processing is specified at 192 kHz. Is that hard & fast or could I run at 96 or 48 kHz? I suppose a sample rate converter block for lows and mids would be reaching for the moon...
Here's the deal: FIR filters need the square of DSP clocks as the processing rate goes up. Say for example you have a 100 point FIR at 48 kHz; it must be 200 points to keep the same frequency response at 96 kHz as at 48. However, you're executing the longer filter twice as many times, so you need 2 x 2 = 4 times the clocks to run it at double the sample rate. 192 kHz, well, the filter is 4 x as long and runs 4x as often, so it wants 16 times the clocks. Suddenly a middling frequency resolution 100 tap FIR looks like it'll chew up a fair bit of DSP power, and never low to mid crossover FIRs wanting 960 taps at 48 kHz! You go from 46 million multiplyaccumulates (MMACs) at 48 K to 737 MMACs at 192 kHz. For one filter. Each channel takes one for the woofer, one for the midrange, two channels: now we're talking 2949 MMACs, or about three billion operations a second. That'll warm up the room some. There's another gotcha: the poles of IIR filters approach 1 as their corner frequency decreases. For example, a 100 Hz Butterworth lowpass filter sampled at 48 kHz has poles at 0.9907442546 + j 0.0091708587, 0.9907442546  j 0.0091708587, leading to a recurrence relation of Code:
y[n] = ( 1 * x[n 2]) + ( 2 * x[n 1]) + ( 1 * x[n 0]) + ( 0.9816582826 * y[n 2]) + ( 1.9814885091 * y[n 1]) This Ultimate Preamp 2 says math can happen in 40 bit floats (helpful) or 80 bit fixed point (really useful), which would alleviate the IIR problem. Can we specify which mode is used by each filter, or do all filters use the same operations? Last edited by DSP_Geek; 6th July 2019 at 03:47 AM. 
6th July 2019, 04:24 AM  #1255  
diyAudio Member

Quote:
Floating point numbers are handled internally with 40 bit precision and fixed point at 80 bits. There are fixed point and floating point filters in the Audioweaver library so you get to choose what best suits your application cheers david 

6th July 2019, 03:54 PM  #1256  
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Quote:
<look outside> Funny, there's no snow on the ground but it sure feels like Christmas. I couldn't ask for any more. Well, maybe for delay specs on the up & down converters Happy dance! 

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