Go Back   Home > Forums > > >
Home Forums Rules Articles diyAudio Store Blogs Gallery Wiki Register Donations FAQ Calendar Search Today's Posts Mark Forums Read

Vendor's Bazaar Commercial Vendors large & small hawking their wares

Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
Please consider donating to help us continue to serve you.

Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
Reply
 
Thread Tools Search this Thread
Old 25th June 2019, 09:54 PM   #1251
NATDBERG is offline NATDBERG  United Kingdom
diyAudio Member
 
Join Date: Nov 2006
Me too (given the right amount of time to make sure I have the funds)
  Reply With Quote
Old 26th June 2019, 02:43 AM   #1252
gadut is offline gadut  Indonesia
diyAudio Member
 
gadut's Avatar
 
Join Date: Dec 2008
Location: Jakarta, Indonesia
Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
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
__________________
AD1865||DCB1||FW PSU||J2||M2||SIT L'Amp||Open Baffle
  Reply With Quote
Old 26th June 2019, 02:06 PM   #1253
Tranquility Bass is offline Tranquility Bass  Australia
diyAudio Member
 
Tranquility Bass's Avatar
 
Join Date: Jun 2013
Location: Australia
Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
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 Pre-order details. Still finalizing the details so shouldn't be too much longer

cheers
david
  Reply With Quote
Old 6th July 2019, 03:39 AM   #1254
DSP_Geek is offline DSP_Geek  Canada
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 multiply-accumulates (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])
which features a gain of 2.356080688e+04 (about 14.5 bits) around the denominator feedback loop. Given a 32 bit floating point mantissa is 24 bits, scaling the input down by that much before pushing it through the filter leaves about 9.5 bits of resolution, or about 57 dB signal to noise ratio (SNR) through this filter. Arguably you can get away with that filter -- maybe -- but the situation gets quadratically worse as the sample rate increases.

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.
  Reply With Quote
Old 6th July 2019, 04:24 AM   #1255
Tranquility Bass is offline Tranquility Bass  Australia
diyAudio Member
 
Tranquility Bass's Avatar
 
Join Date: Jun 2013
Location: Australia
Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
Quote:
Originally Posted by DSP_Geek View Post
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 multiply-accumulates (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])
which features a gain of 2.356080688e+04 (about 14.5 bits) around the denominator feedback loop. Given a 32 bit floating point mantissa is 24 bits, scaling the input down by that much before pushing it through the filter leaves about 9.5 bits of resolution, or about 57 dB signal to noise ratio (SNR) through this filter. Arguably you can get away with that filter -- maybe -- but the situation gets quadratically worse as the sample rate increases.

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?
There are dipswitch settings on the board to run the native sample rate at 48KHz or 96KHz but of course you need to change your filter coefficients to match. You can also use the down convert and up-convert modules in Audioweaver to reduce the sampling rate for better use of the DSP resources.

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
  Reply With Quote
Old 6th July 2019, 03:54 PM   #1256
DSP_Geek is offline DSP_Geek  Canada
diyAudio Member
 
Join Date: Aug 2003
Location: Santa Cruz, California
Quote:
Originally Posted by Tranquility Bass View Post
There are dipswitch settings on the board to run the native sample rate at 48KHz or 96KHz but of course you need to change your filter coefficients to match. You can also use the down convert and up-convert modules in Audioweaver to reduce the sampling rate for better use of the DSP resources.

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
I imagine 80 bit math takes more time than 40 bits, but anyone doing FIRs in 80 bits wants a gentle talking-to.

<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!
  Reply With Quote

Reply


Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC projectHide this!Advertise here!
Thread Tools Search this Thread
Search this Thread:

Advanced Search

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off

Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Full Digital / CrossOver / Multi Amplifier integrated KOON3876 Class D 71 21st November 2014 11:56 AM
Multi channel Linux audio player with integrated amplifier questions. ollepetersson PC Based 5 17th March 2013 02:46 PM
Multi purpose multi channel gainclone PCB: Team project - group buy rick57 Chip Amps 14 29th March 2003 05:54 PM


New To Site? Need Help?

All times are GMT. The time now is 08:46 PM.


Search Engine Optimisation provided by DragonByte SEO (Pro) - vBulletin Mods & Addons Copyright © 2019 DragonByte Technologies Ltd.
Resources saved on this page: MySQL 14.29%
vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2019 DragonByte Technologies Ltd.
Copyright ©1999-2019 diyAudio
Wiki