Reference DAC Module - Discrete R-2R Sign Magnitude 24 bit 384 Khz - Page 8 - diyAudio
Go Back   Home > Forums > Commercial Sector > Vendor Forums > Vendor's Bazaar
Home Forums Rules Articles diyAudio Store Gallery Wiki Blogs Register Donations FAQ Calendar Search Today's Posts Mark Forums Read

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

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 23rd July 2014, 08:53 PM   #71
soekris is offline soekris  Denmark
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by GourouLubrik View Post
Sorry if the question is stupid... but I fail to guess if the design is full balanced or balanced by the active output buffer.
Is the J7 unbuffered output balanced too ?
The DAC itself it not balanced, the output buffer convert to balanced, check out data on the TI LME49724 fully balanced opamp.
__________________
Søren
  Reply With Quote
Old 23rd July 2014, 09:07 PM   #72
diyAudio Member
 
Join Date: Mar 2012
Location: Grenoble, France
Thanks, got it in the datasheet.

Is native balanced output possible with 2 pcbs and a typical USB => I2S board like an amanero ?
  Reply With Quote
Old 23rd July 2014, 09:10 PM   #73
soekris is offline soekris  Denmark
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by Ken Newton View Post
In the past, an FPGA based FIR filter would push the designer toward an hardware compact filter implementation, which has typically been an half-band digital filter. Of course, an half-band filter compromises filter band-edge performance (being down only 6dB at the Nyquist frequency) for hardware compactness.

If hardware resources inside that Spartan 6 FPGA will permit, I suggest implementing an non-half-band filter which is fully in to the filter's stop-band at CD's 22.050KHz Nyquist frequency, while flat by 20KHz.
The Spartan-6 LX16 used have 15K logic cells, 32 DSP48A1 MAC blocks, they need to be used in blocks of 4 to do 35 x 35 bit multiplications plus 70 bit summers, so 8 full high resolution MAC's will be available. The -16 also has 576 Kbit RAM blocks.
I will probably assign 2 for the first x2 most critical oversampling FIR filters, running them at just 49.152 Mhz makes space for 1024 coefficients if needed, then 2 more for the rest of the FIR filters, and then 2 more for other functions, like de-emphasis, volume control and digital crossover filters. Assuming there is space for the control logic.

So the Spartan-6 LX16 have plenty of DSP resources. And if that it not enough, the boards can take the larger LX25....
__________________
Søren
  Reply With Quote
Old 23rd July 2014, 09:25 PM   #74
soekris is offline soekris  Denmark
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by GourouLubrik View Post
Thanks, got it in the datasheet.

Is native balanced output possible with 2 pcbs and a typical USB => I2S board like an amanero ?
In principle yes, just need a configuration setting to make it a balanced single channel module, I'll try to remember to add that.

Although I don't see the big need for the DAC resistor string to be balanced, balanced signals are most useful for box to box interconnect.... You will get 3 db better S/N ratio, not that it's needed, there should already be 126 db S/N ratio at the resistor string....
__________________
Søren
  Reply With Quote
Old 23rd July 2014, 09:29 PM   #75
diyAudio Member
 
Join Date: Mar 2012
Location: Grenoble, France
It's also useful when using balanced headphones
You also get noise rejection, more power, and better dynamic (in fact, better "drive" of the magnets - dynamic range figures at that point are quite useless, but driving drivers with two opposites phases is not ).

Last edited by GourouLubrik; 23rd July 2014 at 09:37 PM. Reason: precising dynamic/driving
  Reply With Quote
Old 23rd July 2014, 10:26 PM   #76
Calvin is online now Calvin  Germany
diyAudio Member
 
Calvin's Avatar
 
Join Date: Nov 2004
Location: close to Basel
Hi,

in #59 You wrote:
Quote:
... you can select any even oversampling rate you want (up to hardware limits), or none....
.
The Q is not if it is doable, but if Your DAC automatically detects the incoming music stream's clock rate and if it automatically switches its own clock rate, locking to an even factor of the incoming rate.
In other words, when music streams of 44.1, 88.2 or 176.4 are to be played will it clock at 176.4?
And will it switch to 192, when music streams of 48, 96 or 192 are to be played?
While on a CD the clock of course can't be any different from 44.1, but with music servers or clients the clock may change from track to track.
I know that You can use the SIS clock chips for this kind of task and You can cut down re-locking period to unhearable low values.
I never really understood why CD-Players introduced oversampling to 192 instead of 176.4?

jauu
Calvin
  Reply With Quote
Old 23rd July 2014, 10:55 PM   #77
soekris is offline soekris  Denmark
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by Calvin View Post
Hi,

in #59 You wrote:
.
The Q is not if it is doable, but if Your DAC automatically detects the incoming music stream's clock rate and if it automatically switches its own clock rate, locking to an even factor of the incoming rate.
In other words, when music streams of 44.1, 88.2 or 176.4 are to be played will it clock at 176.4?
And will it switch to 192, when music streams of 48, 96 or 192 are to be played?
While on a CD the clock of course can't be any different from 44.1, but with music servers or clients the clock may change from track to track.
I know that You can use the SIS clock chips for this kind of task and You can cut down re-locking period to unhearable low values.
I never really understood why CD-Players introduced oversampling to 192 instead of 176.4?

jauu
Calvin
The DAC will continuous measure the incoming bit rate and adjust the internal programmable clock as needed, taking up slack in a FIFO buffer (with configurable size), when clock changes it will select overclocking rates as needed (plan to stick to even ones only) and select oversampling filters as needed, one or more sets of optimized filter co-efficients per standard clock rate. No sample rate conversions here, the goal is bit perfect. And as it's not a fixed analog PLL the switching of rates can be very fast, target is something like down to maybe 5 mS, again configurable.

As you can read, I plan for a very configurable DAC, it can be optimized for the intended use, that being home hifi, home theater, live performance or something else....
__________________
Søren
  Reply With Quote
Old 24th July 2014, 03:02 AM   #78
Brendon is offline Brendon  Canada
diyAudio Member
 
Join Date: Nov 2002
Location: Calgary
Hi, I'm interested in a DAC too.

Thanks,


Brendon
  Reply With Quote
Old 24th July 2014, 03:32 AM   #79
casshan is offline casshan  Hong Kong
diyAudio Member
 
Join Date: Jul 2006
If I connect the raw output of this R2R, should I add the i/v stage?
  Reply With Quote
Old 24th July 2014, 04:12 AM   #80
derekr is offline derekr  Barbados
diyAudio Member
 
Join Date: Mar 2009
Put me down for a 0.02.

Great concept!
  Reply With Quote

Reply


Hide 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
Trackbacks are Off
Pingbacks are Off
Refbacks are Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
e18 DAC - 8 channels at 32bit /384 kHz exa065 exaDevices 30 29th June 2012 05:11 PM
384 Khz DAC? SunRa Digital Source 8 1st October 2009 11:14 PM
24 bit/192 kHz via USB? gentlevoice Everything Else 3 22nd December 2008 06:24 AM
sign magnitude DAC Bernhard Digital Source 0 30th January 2007 01:40 PM
24 bit / 192 kHz Tube DAC questions Overlord Digital Source 4 29th April 2003 05:14 PM


New To Site? Need Help?

All times are GMT. The time now is 11:01 AM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2017 DragonByte Technologies Ltd.
Copyright ©1999-2017 diyAudio

Content Relevant URLs by vBSEO 3.3.2
Wiki