Linux Audio the way to go!? - Page 227 - diyAudio
Go Back   Home > Forums > Source & Line > PC Based

PC Based Computer music servers, crossovers, and equalization

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 6th February 2014, 07:28 AM   #2261
phofman is offline phofman  Czech Republic
diyAudio Member
 
Join Date: Apr 2005
Location: Pilsen
Digital volume control belongs to the golden bestsellers among audiophiles :-)

Input precision (information): 16 bit number - X
Divisor: 16 bit number - K
Output: 16 bit number - Y

Y16 = X16 / K16

Now what happens if we add more zero bits to X, some non-zero bits to the divisor K and do the division in higher bitwidth:

Y48 = X48 / K48

Yes, Y48 will have 32 extra non-zero bits. BUT the first 16 bits will be absolutely identical to Y16, for any input values of X16 and K16.

If we decimate back to Y16 by removing the lower 4 bytes (32 bits), we will always get the same number as in the original Y16 calculation. It is simple math.

OK, decimation should be done with dither. But dither by principle affects only the LSB. That means the only difference can be at the 16th, least significant bit. No matter how precise the volume calculation is.

However, we often have 24bit DACs and 16 bit source music. The 24th bit (plus a few more) is always way below the noise of the DAC, dithering at 24 bits is useless (and sound-processing softwares do not do it, unless they are aimed at audiophiles, how typical). Therefore, anything above volume calculation at 24 bits without dither with output at 24 bits without subsequent decimation to 16bits is plain audiophile voodoo. Internal volume control in 24bit DACs does division in 24bits without any dither too and nobody asks about its quality. Well, in the end it is done in HW, thus must be perfect :-)

I understand devices claiming 48bit volume control sell better, as clueless customers do not understand it is just a marketing ploy.

OK, how to do the above. Define output alsa device accepting only 32bits, put softvol below, and let softvol output to the plug plugin adjusting sample width to that supported by your soundcard. That way softvol will run at 32 bits.

Or look at MPD source code, find its internal volume control algorithm (most likely in float or int32), and use that.

In all cases it will not have any effect on the resultant sound whatsoever. But the mind will be calm and that is what counts :-)
  Reply With Quote
Old 6th February 2014, 02:53 PM   #2262
1audio is offline 1audio  United States
diyAudio Member
 
Join Date: Mar 2004
Location: SF Bay Area
Blog Entries: 3
I think you missed my point- the internal registers for 2 24 bit numbers need to be at least 48 bits. The result if truncated to 16 bits (or even 24 bits) it would benefit from dither.

One of the significant enhancements in the "32" bit DAC's is 32 bit digital volume control. The DACs are still not much better than 22 bits but the reduction in artifacts from the volume control are supposed to be significant. This may explain better: The Well-Tempered Computer or this http://www.esstech.com/PDF/digital-v...me-control.pdf . The only reason I brought this up is that a better volume control, like a better sample rate converter, brings a burden in cpu demand, something that is always a tradeout.
__________________
Demian Martin
Product Design Services
  Reply With Quote
Old 6th February 2014, 04:22 PM   #2263
phofman is offline phofman  Czech Republic
diyAudio Member
 
Join Date: Apr 2005
Location: Pilsen
Quote:
Originally Posted by 1audio View Post
I think you missed my point- the internal registers for 2 24 bit numbers need to be at least 48 bits.
Honestly, I do not know the exact implementation of ALUs, but I am pretty sure the algorithm for multiplication of two numbers in them has been done correctly for a few decades.


Quote:
The result if truncated to 16 bits (or even 24 bits) it would benefit from dither.
How would the truncation to 24bits benefit from dither in the real world?


Quote:
One of the significant enhancements in the "32" bit DAC's is 32 bit digital volume control. The DACs are still not much better than 22 bits but the reduction in artifacts from the volume control are supposed to be significant.
If the 8 LSBits of 32bit DAC as opposed to 24bit DACs are below the noise level, how do they contribute to the higher resolution? What artifacts of volume control do you mean? It is a simple division/multiplication of all samples by a fixed number.


That material has been here several times. I do not see any valid argument for 32bit DAC in it, it is marketing IMO. If the noise floor is way above 24 bits, what is the advantage of the remaining bits?

Quote:
The only reason I brought this up is that a better volume control, like a better sample rate converter, brings a burden in cpu demand, something that is always a tradeout.
Sample rate conversion quality can be measured, pretty much at arbitrary precision. How do you define "better volume control", when the bits below the 24th bit get dropped (or get blurred in the noise for 32bit DACs)?
  Reply With Quote
Old 8th February 2014, 01:57 AM   #2264
diyAudio Member
 
Join Date: Jan 2012
Quote:
Originally Posted by phofman View Post
Digital volume control belongs to the golden bestsellers among audiophiles :-)

Input precision (information): 16 bit number - X
Divisor: 16 bit number - K
Output: 16 bit number - Y

Y16 = X16 / K16

Now what happens if we add more zero bits to X, some non-zero bits to the divisor K and do the division in higher bitwidth:

Y48 = X48 / K48

Yes, Y48 will have 32 extra non-zero bits. BUT the first 16 bits will be absolutely identical to Y16, for any input values of X16 and K16.

If we decimate back to Y16 by removing the lower 4 bytes (32 bits), we will always get the same number as in the original Y16 calculation. It is simple math.

OK, decimation should be done with dither. But dither by principle affects only the LSB. That means the only difference can be at the 16th, least significant bit. No matter how precise the volume calculation is.

However, we often have 24bit DACs and 16 bit source music. The 24th bit (plus a few more) is always way below the noise of the DAC, dithering at 24 bits is useless (and sound-processing softwares do not do it, unless they are aimed at audiophiles, how typical). Therefore, anything above volume calculation at 24 bits without dither with output at 24 bits without subsequent decimation to 16bits is plain audiophile voodoo. Internal volume control in 24bit DACs does division in 24bits without any dither too and nobody asks about its quality. Well, in the end it is done in HW, thus must be perfect :-)

I understand devices claiming 48bit volume control sell better, as clueless customers do not understand it is just a marketing ploy.

OK, how to do the above. Define output alsa device accepting only 32bits, put softvol below, and let softvol output to the plug plugin adjusting sample width to that supported by your soundcard. That way softvol will run at 32 bits.

Or look at MPD source code, find its internal volume control algorithm (most likely in float or int32), and use that.

In all cases it will not have any effect on the resultant sound whatsoever. But the mind will be calm and that is what counts :-)
After a review of some materials, Id agree with you.

We always talk about individual items like volume controls or dac chips, but if one considers the entire digital signal chain in a playback system, can it possibly support 24 bits of resolution end-to-end?

If not then what resolution?
  Reply With Quote
Old 8th February 2014, 06:52 AM   #2265
phofman is offline phofman  Czech Republic
diyAudio Member
 
Join Date: Apr 2005
Location: Pilsen
Quote:
Originally Posted by tppc View Post
can it possibly support 24 bits of resolution end-to-end?
24/32 is easily achievable, most likely already present in your system.
  Reply With Quote
Old 8th February 2014, 12:48 PM   #2266
diyAudio Member
 
Join Date: Jan 2012
I wasn't referring to the engineered bit width of the processors, yes of course my digital equipment and software support 24/32/64 bit.

I suppose dynamic range would be a better term. I doubt my system is delivering anything close to 24 bit dnr to my ears.
  Reply With Quote
Old 8th February 2014, 01:19 PM   #2267
phofman is offline phofman  Czech Republic
diyAudio Member
 
Join Date: Apr 2005
Location: Pilsen
You system is very likely delivering 24bit dynamic range to your DAC, some parts of it very likely at 32bits. Perhaps if you turn to CERN or NASA engineers, they might help you with a nitrogen-cooled DAC and amplifier delivering 24bit dynamic range to your ears.

However, if you put your volume high enough to discern the 24th bit, at full scale your head will blow up.

Look at e.g. Dynamic Range of Home Listening Roomes and Theaters
  Reply With Quote
Old 8th February 2014, 02:24 PM   #2268
diyAudio Member
 
Join Date: Jan 2012
I've already tried liquid nitrogen on the DAC and the music sounded too cold.

Its an interesting article that concludes maybe 20 bits under ideal conditions, and of course a golden brain and ears to process the extra bits.

If I knock off a few more bits for age related loss of hearing sensitivity then we are probably around 16 bits no?
  Reply With Quote
Old 22nd February 2014, 01:53 PM   #2269
diyAudio Member
 
soundcheck's Avatar
 
Join Date: Mar 2005
Location: D
Did you folks notice, that more and more "audiophile" Linux distros (images) for embedded systems are popping up that run either mpd or squeezelite as base !?!?

Usually people report "better sound" than any of the Windows/Mac based audiophile installations and programs ( and that includes those "audiophile" MS/MAC rip offs).

Yep. Took them a while to figure it out.

"Linux Audio - The way to go !!"

That's what I said in 2007.



I'm running a CubieTruck (100$) + (tailormade) ArchLinux + Squeezelite nowadays. This setup runs at 2.5W and even runs on batteries.
Arch is installed to Nand. With a 80$ USB DAC attached and a little extra tuning I've got my best system ever up'n running.
And I do think we're talking about a rather serious "audiophile" performance.

Last night I cleaned up my workshop. Not sure how many boards and parts I've been collecting over the years. PCs, power supplies, DDACS, Twisted Pear Sabres, DDX320, Squeezebox Duet and Touch, asf. asf,.............

All gone. What a relieve.


Great stuff.


Enjoy.
  Reply With Quote
Old 22nd February 2014, 02:56 PM   #2270
Julf is offline Julf  Europe
diyAudio Member
 
Join Date: Oct 2011
Location: Amsterdam, The Netherlands
Quote:
Originally Posted by soundcheck View Post
Usually people report "better sound" than any of the Windows/Mac based audiophile installations and programs ( and that includes those "audiophile" MS/MAC rip offs).
Usually people report better sound whenever they change anything in their system.
__________________
"To try to judge the real from the false will always be hard. In this fast-growing art of 'high fidelity' the quackery will bear a solid gilt edge that will fool many people" - Paul W Klipsch, 1953
  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



New To Site? Need Help?

All times are GMT. The time now is 11:55 PM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2014 DragonByte Technologies Ltd.
Copyright 1999-2014 diyAudio

Content Relevant URLs by vBSEO 3.3.2