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

Digital Line Level DACs, Digital Crossovers, Equalizers, etc.

DAC gallery
DAC gallery
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 2nd July 2018, 04:59 AM   #291
3lite is offline 3lite  Poland
diyAudio Member
 
Join Date: May 2016
Quote:
Originally Posted by abraxalito View Post
Lots of hours spent in PCB layout there, kudos.

I see you have some fairly complex active filters after your DAC, based on opamps. Have you tried LC (passive) filters? I ask because I tried a complex opamp-based elliptic filter after my DAC and didn't like the sound much. These days I only use passive filters - attached prototype 9th order LC filter I/V stage.
It's frequency dependent negative resistance and those opamps are not in the path of an audio signal. Take a look:

Click the image to open in full size.

Instead of NE5532 there are LM4562 though.

Last edited by 3lite; 2nd July 2018 at 05:01 AM.
  Reply With Quote
Old 2nd July 2018, 06:48 AM   #292
xx3stksm is offline xx3stksm  Japan
diyAudio Member
 
xx3stksm's Avatar
 
Join Date: Jun 2017
Location: Hokkaido(north area)
Default moving average filter

I suppose your x32 moving average filter is similar to the way used in DSD by Sygnalyst. AD1865 can't operate at high sampling freq like DSD topology. I guess 768kHz is the max. In such situation, the full addition of 32 outputs (every coefficient are 1) inevitably has high freq droop. How do you organize your 32 outputs for proper frequency response in the audio band? Are your coefficients not equal to 1?
  Reply With Quote
Old 2nd July 2018, 06:57 AM   #293
dddac is offline dddac  Germany
DDDAC Audio
diyAudio Member
 
Join Date: Jul 2003
Location: Wiesbaden
DAC gallery
Quote:
Originally Posted by xx3stksm View Post
I suppose your x32 moving average filter is similar to the way used in DSD by Sygnalyst. AD1865 can't operate at high sampling freq like DSD topology. I guess 768kHz is the max. In such situation, the full addition of 32 outputs (every coefficient are 1) inevitably has high freq droop. How do you organize your 32 outputs for proper frequency response in the audio band? Are your coefficients not equal to 1?
maybe you can run a 1kHz squarewave (Fs44.1) through it and show the result?
__________________
www.dddac.com
Happy listening and building
  Reply With Quote
Old 2nd July 2018, 07:27 AM   #294
3lite is offline 3lite  Poland
diyAudio Member
 
Join Date: May 2016
Quote:
Originally Posted by xx3stksm View Post
I suppose your x32 moving average filter is similar to the way used in DSD by Sygnalyst. AD1865 can't operate at high sampling freq like DSD topology. I guess 768kHz is the max. In such situation, the full addition of 32 outputs (every coefficient are 1) inevitably has high freq droop. How do you organize your 32 outputs for proper frequency response in the audio band? Are your coefficients not equal to 1?
Moving-average is just a simple FIR filter, but the results are calculated within analog domain. In fact, its ability to filter and attenuate is quite poor in general. The idea is the same as within the following thread:

Building the ultimate NOS DAC using TDA1541A

It does exactly the same thing - linear interpolation in the analog domain.

Impulse response of that D/A converter looks like this:

Click the image to open in full size.

1 kHz sinusoidal signal in a D/A converter with one AD1865 and NOS mode looks like this:

Click the image to open in full size.

Having 32x AD1865 to interpolate that signal using linear interpolation (well, 16 to be precise since the other 16 are used to create inversion of the signal) looks like this:

Click the image to open in full size.

4 kHz with 1x AD1865 in NOS mode:

Click the image to open in full size.

4 kHz with 16x AD1865 and linear interpolation:

Click the image to open in full size.

10 kHz with 1x AD1865 in NOS mode:

Click the image to open in full size.

10 kHz with 16x AD1865 and linear interpolation:

Click the image to open in full size.

The higher we get the ability to reconstruct the signal using such simple FIR is falling down quite heavily. It's expected and normal behavior.

However, after all - it does sound quite good. In fact, besides sounding good it does measure quite well since that kind of output swing from multiple AD1865 heavily decreases THD compared to only one AD1865:

Click the image to open in full size.

There is not much to it. It's a really simple idea.

Last edited by 3lite; 2nd July 2018 at 07:33 AM.
  Reply With Quote
Old 2nd July 2018, 11:55 AM   #295
xx3stksm is offline xx3stksm  Japan
diyAudio Member
 
xx3stksm's Avatar
 
Join Date: Jun 2017
Location: Hokkaido(north area)
Thank you for your detailed explanation. It's very interesting and exciting! This is the first time to see 16 averaging by analog DAC and the performance.
I did SIM from your screenshot of 10kHz with 16 averaging. I guess original data is sampled by 48kHz and upsampled to 768kHz to do analog interpolation.
The attached is 1024 length FFT. The second one is magnifying of the first. The third is 12kHz sinewave before and after the filter. It looks like your pic of 10kHz.
Attached Images
File Type: jpg fir1.jpg (136.6 KB, 24 views)
File Type: jpg fir2.jpg (186.2 KB, 23 views)
File Type: jpg fir3.jpg (138.0 KB, 19 views)
  Reply With Quote
Old 2nd July 2018, 12:03 PM   #296
3lite is offline 3lite  Poland
diyAudio Member
 
Join Date: May 2016
Cool! Indeed it is, I'm sorry I did not mention it. It's upsampled by a factor of 16 which means that for 48 kHz input it is running at 768 kHz on the output and for 192 kHz input it is running at... whooping 3.072 MHz
  Reply With Quote
Old 2nd July 2018, 01:03 PM   #297
Tam Lin is offline Tam Lin  United States
diyAudio Member
 
Join Date: May 2010
Location: Texas
Quote:
Originally Posted by 3lite View Post
Cool! Indeed it is, I'm sorry I did not mention it. It's upsampled by a factor of 16 which means that for 48 kHz input it is running at 768 kHz on the output and for 192 kHz input it is running at... whooping 3.072 MHz
Linear linear interpolation is not upsampling. It simply changes the output waveform, prior to reconstruction, from zero-order hold to first-order hold. The true sample rate is unchanged but the distortion and high frequency attenuation is increased. This is easily seen in the photos in post #294.

Adding galvanic isolation to I2S adds massive jitter. Look at the ISO7640 data sheet. Added jitter is measured in NANO-seconds. And driving multiple DACs and shift registers with the same clock signal doesn't help, either.
__________________
Jon Bokelman
  Reply With Quote
Old 2nd July 2018, 01:22 PM   #298
3lite is offline 3lite  Poland
diyAudio Member
 
Join Date: May 2016
It's just a matter of expression towards the question from xx3stkstm. There is no need to be cocky about it since I'm very well aware of how that D/A converter works.

How about you provide actual measurements of jitter? I performed several measurements including a J-Test in ARTA of this D/A converter and others and there is hardly anything to worry about. You do seem to scream a lot, but you have not much idea what is actually going on since reading datasheets won't get you anywhere if you don't know how real world works.

It is obvious for anyone who knows about filtering that this D/A doesn't do that very well. That was the whole point of this D/A converter - to try out how zero-order hold (NOS) and first-order hold (this DAC) differs from each other in terms of sound quality.
  Reply With Quote
Old 3rd July 2018, 05:50 AM   #299
xx3stksm is offline xx3stksm  Japan
diyAudio Member
 
xx3stksm's Avatar
 
Join Date: Jun 2017
Location: Hokkaido(north area)
I agree with 3lite about jitter performance. I have measured jitter of optical SPDIF several times. Recovered clock from SPDIF receiver usually has 3 nanosecond jitter at max. It's easy to measure output jitter of DAC with jittery clock and with precise one. The clock to DAC is irrelevant to output jitter of DAC even if 3 nanosecond jitter exists. Jitter interference results in noise power of DAC output. Three nanosecond jitter is far less than 110dB THD+N, which is high-end performance.

What you need to measure is not the clock to DAC but the output of DAC. Noise power coming from the jittery clock is proportional to the DAC output frequency. The audio band frequency is very low(20kHz at max) while RF application is very high (more than 200MHz). In this situation, audio application inherently has less jitter oriented noise power by 80dB. That's why the output of DAC is not sensitive to jittery clock than a usual prospect many people have. That's my conclusion in the real world. The audio frequency is low.
  Reply With Quote

Reply


DAC galleryHide 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
diyclub.biz DAC-38, DAC-60, DAC-68. Anybody heard? TonyDasilva Digital Line Level 21 13th April 2009 05:10 PM


New To Site? Need Help?

All times are GMT. The time now is 05:49 PM.


Search Engine Optimisation provided by DragonByte SEO (Pro) - vBulletin Mods & Addons Copyright © 2018 DragonByte Technologies Ltd.
Resources saved on this page: MySQL 14.29%
vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2018 DragonByte Technologies Ltd.
Copyright ©1999-2018 diyAudio
Wiki