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post #31 of 980


This is super helpful! Will try it out right now and see how it turns out.

As for Yulong's website there isn't really one as far as I am aware of, except for a sub-forum of the erji.net. http://bbs2.erji.com/thread.php?fid=150

You could try reading it using google translator, or you could tell me what you are interested in and I could try searching it for you.

Quote:
Originally Posted by slackman View Post

So far it appears to me nobody is using the D18 to anywhere near its full potential.

The D18 does not have an upsampler built in (and I'm not 100% sure but I think also no PLL jitter attenuation etc, other than what the Sabre DAC itself does)

The proper way to use the D18 is to upsample the music on the computer to 192Khz 24bit (or 32bit) and use a very high quality asynchronous USB to AES/EBU or spdif converter with low jitter like the Audiophileo2, Stello U3 or M2tech EVO.

 

The computer can upsample better than any included upsampler in any DAC, also in real time (SOX for foobar or cPlay for instance). Set it to linear phase with 99% bandwidth for correct upsampling (not nonsense minimal phase which has nothing to do with reconstructing the original wave according to the sampling theory upon which digital audio is built).

This upsampling will give you better than 175dB THD+noise, this is inaudible.

 

The difference between doing the above correctly and simply streaming 44.1 Khz audio through a mediocre USB-spdif converter (or directly from a cd transport) will be huge.

 

Btw, to make it lighter on the computer one can also upsample 44.1Khz audio to 176.4Khz instead of 192.

 

edit: here is a link to the SOX resampler for foobar2000: http://www.hydrogenaudio.org/forums/index.php?showtopic=67373 simply unpack the zip to your foobar2000 components folder and then in foobar load and configure the resampler under preferences - dsp manager.



 


Edited by Philharmoniker - 12/21/11 at 1:28pm
post #32 of 980


 

Quote:
Originally Posted by Philharmoniker View Post


This is super helpful! Will try it out right now and see how it turns out.

As for Yulong's website there isn't really one as far as I am aware of, except for a sub-forum of the erji.net. http://bbs2.erji.com/thread.php?fid=150

You could try reading it using google translator, or you could tell me what you are interested in and I could try searching it for you.

Correction: Apparently they do have a website, but like project86 said, it seems pretty ancient...



 

BTW Slackman, do you know what shows that the SOX upsampling is working, since in the corner foobar still tells me the original information of my track...
 

 


Edited by Philharmoniker - 12/21/11 at 1:41pm
post #33 of 980

Hi Project86,

 

Thanks for your reply and the website link!

Very much looking forward to your review of the D18 :)

 

A good DAC needs upsampling.

Many people think that a DAC is supposed to reconstruct the stepped 16 bit 44.1Khz waveform and then simply filter off the sharp edges so everything above half the sampling frequency is removed.

However this is wrong! This is not at all how the sampling theorem works.

Yet our NOS DACs work this way.

 

Here is how reconstructing the original sample waveform should work:

http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem#Reconstruction

 

Note also that it isn't actually a "filter" and it doesn't "ring".

The thing that's refered to as ringing is actually caused by the low pass filter used when encoding the data.

A good computer oversampler reconstructs the digitally encoded waveform by far perfect enough (with inaudible THD+noise).

And I'll say it again, to use a "minimal phase oversampling filter" is nonsense. Linear phase high bandwidth oversampling is the only technically correct reconstruction of the waveform.

 

To not oversample as in a NOS dac with 44.1 Khz data as described in the top of this post will give very audible (and measurable) errors (which we often perceive as warm btw, since specific frequency bands in especially the trebble are much lower in volume because of this error)

Contrary to popular belief these errors are not only for certain frequencies, but also for amplitude / impuls behaviour (these are not seperable in the wave, they come from one and the same thing).

So the funny thing is where people always say that NOS DACs etc have better impuls behaviour it's actually the other way around! :)

You need correct high tones to make a correct amplitude curve.

To make clear in a short way the error of NOS DACs btw: take 44.1 Khz sampling frequency. Now play a 22.004999 Khz sine wave (this is slightly under the Nyquist frequency so it can be reproduced.)

Depending on where in the phase of the sine wave the sampling occurs, it can mean that actual samples can be recorded at either the very top and bottom of the sine wave form, or exactly at the 0dB crosspoints (so the samples record actually nothing! Silence!) And the sine wave then very slowly changes phase to the sampling frequency.

To simply output the bit values (as a NOS DAC does) and then simply filter above the Nyquist, gives ACTUAL SILENCE where the wave is in such phase that the samples record nothing.

This is a severe error. It goes like a riple all the way through the entire frequency band till the lowest frequencies. However the effect is much stronger in the treble than in the mids than in the bass. At the highest frequencies the error can be 100% (silence when there should be a full scale wave), at half the Nyquist the error is maximally -6dB or -3dB (can't remember out of the top of my head), and at again half that frequency it is very small (4 times as small perhaps, possibly even smaller, again don't remember the math and too lazy to look it up :)

 

Now the sampling theory works differently. It says one can reconstruct the original waveform by making the "slowest angle" waveform that runs through the connected dots (it can do so because it assumes the sample values represent only a waveform that has no content whatsoever above the Nyquist)

Therefore one CAN actually recreate the waveform in the example above that is just a fraction below the Nyquist. So no silence where there should be a wave, but we get the actual wave back (even though a large timespan of the samples were taken at the 0dB crossing. All the errors disappear. (this also helps make clear the ringing that must happen when we bandlimit (filter) the original wave during the analogue to digital conversion, If the frequency is high already, it can't have a fast volume envelope as the volume envelope modifies the waveforms angle speed, this means it makes part of the wave higher in frequency and this is removed when this leads to a higher frequency than the nyquist. So to limit frequency is to limit impulse, ringing is an integral part of bandlimiting, no way around it, and bandlimiting is an integral part of the sampling theorem. It's not bad, it's high in frequency I don't know if we can hear it, but a DAC sure as hell can't fix it should it need fixing without doing other things completely totally wrong.

Sorry I had to get that off my chest it had been bothering me for a long time what I read about these issues in popular hifi talk.

 

Now so we need upsampling to give an honest reproduction of the waveform.

44.1 vs upsampled to 96 will make a big difference. 44.1 192 is slightly better than 96, audible on a good system but subtle if I'm correct.

384 very slightly better than 192, don't know if it's audible.

(edit: btw you can also upsample from 44.1 to 88.2 or 176.4 or 352.8, it will be lighter on the computer and give less errors with not so great upsamplers)

 

As for computer upsampling vs an asynchronous sample rate converter onboard a DAC:

The asynchronous here does not mean a good thing (like it does in USB data transfer), it means a very bad thing. It means there is no "lock" between the original and upsampled frequencies, it has errors in upsampling not only in bit depth rounding of values but also in timing.

However, it still reconstructs the waveform better than if we were to not upsample at all (only it gives us new errors)

The computer doesn't have these errors and performs "synchronous" SRC where the 2 frequencies are locked (well of course it has some errors, long story to explain, but these are so low for a good upsampler they don't matter)

So computer is better than upsampler on DAC. It's as sample as that, computer does it better.

I do belief that there is some other reasons why DACs use upsamplers, there are some tricks with them to reduce or modify jitter or certain types of jitter, I don't know the specifics of this.

 

In the Sabre DAC itself there is already ASRC being done (in a completely different way that as described above, as it's a very high frequency DAC it needs to do this) and also jitter reduction.

If the sabre is fed with a low jitter bit perfect perfect signal after the computer upsampler, this is logically the best possible way to feed the SABRE.

This is possible now. Take for instance the Audiophileo2 which has a claimed jitter of only 2.6ps rms (compared to about 30ps rms jitter beeing fed in the Anedio D1 to the Sabre DAC after USB conversion and it's PLL's) and does 24 bit 192Khz.

 

This should make all the difference in the case of the D18, and it could possibly perform better then the Anedio D1 etc when used in this way.

Please try it out, I don't have my D18 yet. I think you could be going for words like "incredibly neutral and revealing" instead of "warm" etc when you do :)

 

 

EDIT: What I wrote is not correct. The Sabre DAC in the D18 does it's own "upsampling" so it does reconstruct the waveform correctly without additional user upsampling. The only difference with using computer upsampling will be the slightly increased quality of good computer upsampling.

Sorry for being stupid. Thought I'd edit the related posts for the people who brows this thread later.


Edited by slackman - 12/25/11 at 5:44am
post #34 of 980

Ok I downloaded the file, is it called (Resampler PPHS)? I don't know If I did this correctly. When I configured it the target sampling rate only went to 96000. I am running this through the  Musiland Monitor 03

post #35 of 980
Quote:
Originally Posted by Philharmoniker View Post


 

BTW Slackman, do you know what shows that the SOX upsampling is working, since in the corner foobar still tells me the original information of my track...
 

 



When it is under "Active DSPs" in foobar's DSP manager then it's active.

To see if it upsamples to your desired frequency click on the "Resampler (SOX)" and see the resampling frequency displayed. (btw if you set it higher than your driver supports foobar will give an error)

 

With my Lavry DA10 I could see a light go on indicating the changed sample rate, but the difference was very small (since the Lavry resamples everything with ASRC to somehwere in the vicinity of 90Khz I belief (could be a bit higher, don't remember)

But with for instance my low cost Echo Audiofire2 there are no lights to indicate anything but the difference is very audible (which btw uses the same converter chip as the Apogee MiniDac I belief)

So I've simply set it to the highest sample rate your driver can support and listen for the differences by switching the resampler in and out.

(it will depend on your entire system of course how much difference you'll be able to hear)

Please do report back :)

 

post #36 of 980
Quote:
Originally Posted by Sonic77 View Post

Ok I downloaded the file, is it called (Resampler PPHS)? I don't know If I did this correctly. When I configured it the target sampling rate only went to 96000. I am running this through the  Musiland Monitor 03



No it is Resampler (SOX). Resampler PPHS comes with foobar2000 as standard but is of lower quality.

You must unpack the SOX resampler to the folder called "components" in your foobar2000 folder under program files and then restart foobar2000.

post #37 of 980


Ok, I'll try again, thank you!
 

Quote:
Originally Posted by slackman View Post



No it is Resampler (SOX). Resampler PPHS comes with foobar2000 as standard but is of lower quality.

You must unpack the SOX resampler to the folder called "components" in your foobar2000 folder under program files and then restart foobar2000.



 

post #38 of 980

Btw, another nice tip for people that don't have an asynchronous USB to spdif interface, but an isochronous USB interface:

You can improve the jitter comming off the computer slightly by using cPlay instead of foobar2000.

cPlay will load the entire track in RAM and streams from there as accurately as possible. I could detect subtle differences myself.

If you have an asynchronous USB interface this shouldn't make any difference at all (except for slightly less computer HD noise haha, which you should eliminate anyhow if you can hear your computer!)

post #39 of 980

I think it is working, since when I remove SOX upsampling from the list of DSP and then play the same track there is a brief (500ms) pause.

There is a slight difference in the sound, but I do need to listen longer to tell the difference.
 

Quote:
Originally Posted by slackman View Post



When it is under "Active DSPs" in foobar's DSP manager then it's active.

To see if it upsamples to your desired frequency click on the "Resampler (SOX)" and see the resampling frequency displayed. (btw if you set it higher than your driver supports foobar will give an error)

 

With my Lavry DA10 I could see a light go on indicating the changed sample rate, but the difference was very small (since the Lavry resamples everything with ASRC to somehwere in the vicinity of 90Khz I belief (could be a bit higher, don't remember)

But with for instance my low cost Echo Audiofire2 there are no lights to indicate anything but the difference is very audible (which btw uses the same converter chip as the Apogee MiniDac I belief)

So I've simply set it to the highest sample rate your driver can support and listen for the differences by switching the resampler in and out.

(it will depend on your entire system of course how much difference you'll be able to hear)

Please do report back :)

 



 

post #40 of 980
Quote:
Originally Posted by Philharmoniker View Post

I think it is working, since when I remove SOX upsampling from the list of DSP and then play the same track there is a brief (500ms) pause.

There is a slight difference in the sound, but I do need to listen longer to tell the difference.
 



 



Yes, I get that slight pause too when switching in and out.

This slight difference once you understand what it does to the sound can make all the difference.

It seriously changes the representation of certain things. Things like reverb become more palpable and realistic etc and so many other good effects.

Btw, do be careful to check that when switching the resampler back in it hasn't reset back to 44.1. This happens sometimes. (I think when closing the DSP window with the resampler out, but not sure)

 

Also, one other tip.

Here a great website with data on resamplers:

http://src.infinitewave.ca/

They downsampled 96Khz to 44.1Khz for that data, but should be indicative for upsampling performance as well.

 

Again don't be put off by ringing, this is supposed to happen :) If 96Khz impulse isn't ringing in 44.1 Khz it isn't properly band limited (and don't pay attention to the non ringing "ideal impulse" as that one isn't band limited to 44.1Khz, infact not even to 96Khz so it's nonsense and misleading to put it there)

 

Do be put off by things like non linear phase when upsampling, which can absolutely destroy things like realistic soundstage in certain recordings. (but it can also make bad recordings sound more listenable if that's your thing, simply be careful then when you apply it)

Minimum phase when downsampling can be a creative choise for the producer / mastering engineer, it is natural filtring after all and gives only post-ringing, and then your linear phase upsampler will accurately reconstruct this post-ringing.

 

post #41 of 980

Well.. I just figured out the Sabre ES9018 performs actual upsampling when it does its ASRC..

Sorry for shortly giving the expectation of big performance increase when using upsampling on the computer, when there is actually little to be had (the only difference will be the slightly higher quality of computer upsampling).

I should've waited with posting such things till I actually had my own D18 and tested it mysef.

post #42 of 980

Well.. my head is kind of spinning.

I've been looking at so many DACs the past 2 weeks since deciding to sell my old one :)

And can't really figure out from behind the computer how it can be that 2 DACs, the Anedio D1 and Yulong Sabre D18 who both use the same chip and are very similar in construction and both measure exceptionally well can apparently sound pretty different.

 

Wish that review was already here Project86 :)

 

Of the somewhat affordable (less than $1500) Sabre DACs the D18 and D1 seem to have the best measurements.

And to go by the short reviews I've read so far, they are the best sounding.

And I'm actually slightly too broke right now (after expensive speakers and moving to a new place with huge great sounding room) and not patient enough to wait for the new D1, so I think I'm going with the D18 for now.

 

My speakers are very critical overall but especially critical in the mids, if anything is wrong there they'll let me know (including even very slightly recessed mids or over represented treble or bass in relation)

The descriptions so far seem to indicate that the D18 won't disappoint in this.

Can anybody confirm that the soundstage is still very good with the D18? And that the warm musical character / deviation from the D1 or Weiss DAC2 or Invicta etc is subtle and not very big?

I'd just hate getting a DAC and then not being happy with it and having to wait an undetermined amount of time and loss of money again before I can get a new one.

Thanks!

 

post #43 of 980
Quote:
Originally Posted by slackman View Post

Well.. my head is kind of spinning.

I've been looking at so many DACs the past 2 weeks since deciding to sell my old one :)

And can't really figure out from behind the computer how it can be that 2 DACs, the Anedio D1 and Yulong Sabre D18 who both use the same chip and are very similar in construction and both measure exceptionally well can apparently sound pretty different.

 

Wish that review was already here Project86 :)

 

Of the somewhat affordable (less than $1500) Sabre DACs the D18 and D1 seem to have the best measurements.

And to go by the short reviews I've read so far, they are the best sounding.

And I'm actually slightly too broke right now (after expensive speakers and moving to a new place with huge great sounding room) and not patient enough to wait for the new D1, so I think I'm going with the D18 for now.

 

My speakers are very critical overall but especially critical in the mids, if anything is wrong there they'll let me know (including even very slightly recessed mids or over represented treble or bass in relation)

The descriptions so far seem to indicate that the D18 won't disappoint in this.

Can anybody confirm that the soundstage is still very good with the D18? And that the warm musical character / deviation from the D1 or Weiss DAC2 or Invicta etc is subtle and not very big?

I'd just hate getting a DAC and then not being happy with it and having to wait an undetermined amount of time and loss of money again before I can get a new one.

Thanks!

 


Grant Fidelity is offering a 30 day free trial of some sort. I didn't get it from them, but I think project86 mentioned it. It's worth looking into.

 

post #44 of 980
Quote:
Originally Posted by slackman View Post

Well.. I just figured out the Sabre ES9018 performs actual upsampling when it does its ASRC..

Sorry for shortly giving the expectation of big performance increase when using upsampling on the computer, when there is actually little to be had (the only difference will be the slightly higher quality of computer upsampling).

I should've waited with posting such things till I actually had my own D18 and tested it mysef.


I was a little curious what you were going on about.  Just because I'm not at all sure that my computer music friends or the people on computer audiophile would agree at all that the PC should be doing the heavy lifting on this - not in real time, in any case, unless your computer is a totally dedicated music server. But whatever - I'm completely out of my league with the theory, so I'll leave that for you to sort out.  There are many here (not on this particular thread, but in sound science, and in computer music) that have spoken about these issues before, and computer audiophile is probably the better forum for this if you want to people with the most technical/theoretical chops.  

 

(btw - young and riley are personal heros ;)

post #45 of 980
Quote:
Originally Posted by Philharmoniker View Post


Grant Fidelity is offering a 30 day free trial of some sort. I didn't get it from them, but I think project86 mentioned it. It's worth looking into.

 



Thanks a lot for the suggestion.

But I'm located in the Netherlands, it would be too much hassle and shipping costs and import duties.

The free auditioning isn't very well done here. Last time I auditioned something (Geithain speakers) I cost me over 200 euro for shipping and recalibration costs after not taking them.. (if anybody is ever looking for ultimate warmth / pleasant huge sound get these speakers haha, but not good for critical music making listening)

 

Have found a German seller though that sells it for €599.

I have a registered music production company so I get the 19% VAT back and it will end up costing me about €500 which is a good price if I end up keeping it (but if I sell it on again I'll have to charge VAT myself which doesn't make such a good resell price which means I'll loose a lot)

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