NOS DAC with oversampling - anybody done it? - Page 3 - diyAudio
Go Back   Home > Forums > Source & Line > Digital Line Level
Home Forums Rules Articles diyAudio Store Gallery Wiki Blogs Register Donations FAQ Calendar Search Today's Posts Mark Forums Read

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

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
Thread Tools Search this Thread
Old 22nd July 2009, 10:53 AM   #21
diyAudio Member
Join Date: Oct 2004
Hi Oon,

What you see as beating is actually the attenuation of higher frequencies. Which is also why you don't hear sss's so much.
If you have attenuated treble, then I'd suspect you don't have a steep cut off low pass filter after the DAC. If you did, I think you'd see all kinds of problems with complicated waveforms, caused by phase and amplitude fluctuations as a function of frequency due to the large order filter's transfer function.
I believe the side affects caused by a digital filter is less severe than that so I feel its a good compromise to a bad design problem (i.e. CD's Fs=44.1kHz).
I'll have a look at the link anyway.
  Reply With Quote
Old 22nd July 2009, 11:01 AM   #22
diyAudio Member
Join Date: Oct 2004
Hi Oon,

However, when you push in a non sinewave, the behaviour becomes unpredictable. For example as shown by the interview at sakura systems, if you put in a square wave, you get a pulsed sine wave.
Are you referring to diagram 14? That is the impulse response to the FIR filter in the digital filter. If you consider that the impulse into the filter is a single sample of a non zero value, which has a period of 1/44100 s, you can see that the ripples before and after the impulse are shorter in period than that, and lower in amplitude. Its highly unlikely they'd impair the sound, especially if you have a sensible filter after the DAC (which is easy to do with an oversampling system as the sampling frequency is much higher).

  Reply With Quote
Old 24th July 2009, 09:51 PM   #23
diyAudio Member
Eric Juaneda's Avatar
Join Date: Dec 2006
Location: Avignon, France
Hi everyone,

There is few DAC where you can adjust oversampling level. The Philips DVP 720SA does. I wrote article about listening comparison.
Listening digital filter

I could listen for a long time many DAC, Up-sampled with low filtering (DAX), Um-sampled with few TAP filter (Wadia X64) and now my personal NOS DAC.
Digital filter tent to foggs music. You loose details, add echo on music (sound depth) and loose natural sound.
Comparing NOS and conventional DAC is not easy, because in some case, digital filter rounds music and it might be very pleasant.

Anyway digital filter is ONLY one part of DAC. NOS or Up-sampled DAC with poor analog stage or poor power supply stills poor DAC.

See picture of my personal NOS DAC (digital decoder on the right and analog stage on the left)

Attached Images
File Type: jpg jundac one.jpg (95.2 KB, 280 views)
  Reply With Quote
Old 24th July 2009, 09:58 PM   #24
diyAudio Member
Eric Juaneda's Avatar
Join Date: Dec 2006
Location: Avignon, France
Default NOS DAC error

You can easily hear error on NOS DAC.
Simply put a CD test with 20KHz sinewave, you can hear 20KHz (if you can and 10KHz or other high audio frequency.

This default disappear if sample frequency is 96KHz or more.

  Reply With Quote
Old 27th July 2009, 10:43 AM   #25
diyAudio Member
oon_the_kid's Avatar
Join Date: May 2008
Location: Singapore
Hi Phil,

It is not so much of how it handles the impulse per-se, but how it might handle music, which are a series of random unpredictable waveforms. You can choose to look at sound as a signal comprising of a sinewave and a whole bunch of harmonics (pretty much fourier transform), or a whole bunch of sguare wave, one after another.

I believe the digital filter has a computation based towards sine wave reconstruction, which looks great when you feed in a repetitive sine wave, but when you don't? Say for example, you have a sinewave for one cycle, triangle for half a cycle followed by a sinewave half the frequency for one cycle, and triangular wave with twice the frequency.

So in this matter there are people who would say that accuracy in time domain is more important than frequency domain (including the full range speaker people), therefore, an ability to produce the waveform accurately with regards to phase is more important, since music is more random signal anyway...

  Reply With Quote
Old 27th July 2009, 10:54 AM   #26
diyAudio Member
Join Date: Aug 2008
the time smearing theory was dismissed long ago, because when they downsample the original 24/ 96khz files in studio to 16/44, they use a steep filter . So good bye steep waveform transitions(UHF) and welcome ringing(fs*0.4X) .
  Reply With Quote
Old 27th July 2009, 12:17 PM   #27
diyAudio Member
Join Date: Oct 2004
Hi oon,

I don't understand you concern about more complicated waveforms. Basic Fourier theory states that all waveforms, no matter how complicated, are built up of sine waves. Therefore, as long as there a no considerable phase errors, if the impulse response is good (which it would be if you filtered out all that ringing) the signal should be in good shape too.
I think you're worrying too much about this.

  Reply With Quote
Old 27th July 2009, 02:53 PM   #28
diyAudio Member
Kurt von Kubik's Avatar
Join Date: Feb 2009
Location: Viby, Denmark
Send a message via MSN to Kurt von Kubik
To me the new NOS era always has been a pussle.
First of all, the brickwall filters needed (thatīs the analog filters) are of extremely high order ( order has been seeen), and therefore tilts the phase way down through the audioband. IMHO the sound of theese are spatialy currupted and also woolly in most implementations.
Some did solve this problem with a non filtered design. That means output containing aliasing residuals @ 1/2 Fs and upwards at same scale as the original audiosignal. That calls difinately for low bandwith gear, nothing else can handle these signals without stressing both circuits and speakers. Some amps might even go into clipping at low listening levels. One should really have respect for out of band signals, not related to music. They create very distructive IM distortion @ frequencies easily audible. But I realise that the NOS DACīs are very popular amongst people using very effective Horn speakers driven by low power tube gear.
This is in some way understandable, because in set-ups with wide bandwith gear, the unfiltered NOS DACs often sound terrible, as a function of their emission of malicious out of band noise to the preceeding amps. Hornloaded drivers will not reproduce this kind of information, as well as tube gear normally is heavily filtered by their output transformers and tubes do have some headroom in higher frequency regions.

The funny thing about oversampling is its history. Philips expected the CD media to become a 14 bit media, but Sony had a vision of 16 bit. philips had @ that time already developed 14 bit DACīs. They became useless this way, but the engineers @ Philips then implemented oversampling @ 4*Fs. Voila, the 14 bit DACs became usefull again, since every doubling of Fs, means the same as 1 bit resolution. 16 Bit wich calls for anti aliasing filtering @ 176.400 Hz instead of 22.050, and with the exact same resolution. Philips players were thus the prefered brand throughout the first years of the CD era.
Thus a lot of subprime brands not capable of building their own CD, used the Philips CD 100 and later models for minor tweakings and major uppricings, mostly they changed the type of up-amps and maybe the design a bit.

Nowadays DAC chips are completely different. Mostly they are delta sigma designs, but especially the filtering and the need for filtering is completely different.
Looking at the most common vendors, youīll find very different calls for filtering. I.e. BB needs smoothing filters, as they output theit analog signal in small steps. But the need for filtering is not very significant. CS nowadays has a built in analog filterstage which is a SCF stage. This filter completely eliminates the need for smoothing and anti aliasing, without any phaseshift, but the filter emits noise @ its working frequency, which is around 4-500 KHz.

So my point is that the concept around NOS DACs was build on past time technology, that did not have the present day tech in mind. It is though understandable, that fanatics seek for other solutions than what is the industry standard by now, because as well as these nerds I donīt either fancy the sound of the "of the shelf" solutions common these days.
  Reply With Quote
Old 27th July 2009, 05:56 PM   #29
diyAudio Member
Join Date: Apr 2007

So, the best compromise solution are a nos dac using an linear (analog) interpolation, 2X, 4X or 8X.


  Reply With Quote
Old 27th July 2009, 07:32 PM   #30
diyAudio Member
Bernhard's Avatar
Join Date: Apr 2002
Location: Munich
Filterless non os is NO GO.
First, the nearly unfiltered signal will come out of tge speakers.
Except for tube amps that have lots of transformers.
Second, there is a tendency to have a passive I/V to avoid having active circuitry treated with high slewing signals.
Then, why would somebody like to enter it into the power amp where bandwith issues are most critical ?

Otherwise, IMHO 60db / oct is steep enough and can be handled technically and practically.
  Reply With Quote


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
Non-Oversampling TDA's supra Digital Source 45 24th September 2004 06:56 AM
TDA5141 oversampling or non-oversampling ? Bernhard Digital Source 4 1st September 2004 10:27 AM
Non-Oversampling DAC lucpes Digital Source 42 6th July 2004 03:06 AM

New To Site? Need Help?

All times are GMT. The time now is 08:02 PM.

vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2017 DragonByte Technologies Ltd.
Copyright Đ1999-2017 diyAudio

Content Relevant URLs by vBSEO 3.3.2