I don't hold Meyer/Moran in any esteem, Jakob. I think it's a crap study to be blunt.
I know and it wasn't my intention to evoke a different impression. cdbd linked it a couple of days ago (apparently because he still thinks it is a good one) and vacuphile mentioned it was "well done" as others praised it already in the past. Despite all analysis and pointing to the obvious problems there seem to be no acknowledgement hence my comment.
hesitating to agree, but it surely depends on the topics examined..... honestly think everything I've read (that's hardly comprehensive for sure) in terms of audible perceptibility has large caveats that need applying
I feel like I'm being painted into a caricature: I don't try to rescue "blind test shows every difference" myth, either, as obviously experiments have huge grey areas of interpretation where we have to be cautious taking any sort of conclusion from them. I thought that's what my post you quoted emphasized! On the other hand, I think your rebuttal of "huge differences" is also an ad hoc "devastating caveat" to the idea that "blind testing has notable merits, if difficult".
Again the "painting into a corner" wasn't my intention, but the "no huge difference" or sometimes "can't be of relevance" (means if not detected in each and every DBT with ease) line of argument was used since the first ABX-test critic - written by Les Leventhal - was published in the JAES.
It goes without saying that "hears everyone who isn't deaf" doesn't need much precaution to be tested in an experiment.
Otoh we know that this kind of exaggeration is not uncommon in human talk/writing.
But if one takes _no_ precautions, states no hypothesis he is going to test, then he can't conclude from negative results that he would have found any difference of relevance.
We'll have to agree to disagree in terms of where the line lays. I find the perpetual claims of "you have to do this (insert this sometimes plausible, if unlikely) thing or you're doing it wrong" obnoxious.
Who's now painting one into a corner? 😉
As already often stated, it is not so much about "doing it wrong" but about what kind of conclusion can be justified by specific experimental procedures and results.
And again, as stated before, if really well planned and executed experiments were consistently giving negative results that would be real corrobation for the "can't of relevance" hypothesis.
But if otoh nearly all reported contradictionary evidence is dismissed (based on the long list of possible reasons, I've posted before) than it gets of course difficult.
Have really to agree to disagree on this?
Oh I think we only agree to disagree on where the line is, but that's nuanced, I was only trying to cover that I personally find both types obnoxious, not projecting onto you.
(Every 45 or so minutes) Pull a vinyl disc out of a sleeve, carefully clean it on a regular basis, set the needle to the correct location, flip the disc half way through and carefully return it at the end...
I'll stick with my digital manipulations. 🙂
The very best sound I have out of my system is the Ella Fitzgerald ‘Irving Berlin Songbook’ on vinyl.
I have some superb CD’s but they don’t trump that vinyl for sheer realism and aural pleasure.
🙂
(I’m with JC on this - and Micky Fremer of Stereophile).
I don't care what system I hear that on, it always grabs me. Might be the performance
🙂
cheers
Alan
🙂
cheers
Alan
We (hopefully) do this stuff for fun, and sounds like you've found your niche, Andrew. My comment was equally tongue in cheek in the sense of "look at all these hoops we have to jump through to get good sound with digital"
🙂
🙂
Alan, you're talking about the real time playback here, right?not deterministic, DSD (random modulator in all conversion) and any up or down sampling, integer or not.
When using properly coded software based SRC I get the same result every time, including conversion from DSD to PCM. The produced files null out completely when compared.
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I'll stick with my digital manipulations. 🙂
You probably still at least listen to whole albums. the yoof of today just have playlists of fave tracks! Like the mix tapes of the past taken to the extreme.
Maybe hire an expert?
While I am far from an expert, I was doing hardware development with Roon, and they turned me on to an audio driver JPLAY (JPLAY | hi-end audio player for Windows) which they use. I do not use the JPLAY player, but the JPLAYDriver64.dll audio subsystem works really well with Foobar2000. I select it in the Playback>Output>Device dropdown and the Auralic Vega DAC front panel bitrate display correctly tracks the raw bitrate of the file, including DSD. It will not do this when the generic Windows audio driver is used.
Just my 2¢ worth...
Howie
Benchmark has selected the ES9018 filters which provide the lowest pass-band ripple. We then frequency-shift the filter transition band upward so that it is centered at 110.5 kHz. We do this by operating the ES9018 at an input sample rate of 211 kHz. This means that the entire transition band of the ES9018 filter is always above the highest audio frequency contained in the incoming audio. At a 192 kHz incoming sample rate, the highest incoming frequency is 96 kHz. This is completely below the lower limit of the transition band that is centered at 110.5 kHz. Benchmark's system effectively eliminates the filters in the ES9018 by frequency shifting the filters out of the audio band. It also completely eliminates all traces of image fold-back. The Nyquist frequency of the D/A converter exceeds the Nyquist frequency of the incoming digital audio.
We used this same frequency-shifting technique in our DAC1 converters. When the DAC1 was designed, the available technology limited us to a D/A input sample rate of 110 kHz. In the DAC1, the D/A filters were out of band for sample rates up to 96 kHz. The DAC2 extends this unique technology to all sample rates up to 192 kHz. The goal for all Benchmark converters has always been to make the digital filters as transparent as possible. The accuracy and precision of the filters is a function of the oversampling ratio used in the filters. Benchmark moves the low-pass filter out of the D/A converter so that it can be executed at a much higher oversampling ratio.
ES9018 is ancient history.
While I am far from an expert, I was doing hardware development with Roon, and they turned me on to an audio driver JPLAY (JPLAY | hi-end audio player for Windows) which they use. I do not use the JPLAY player, but the JPLAYDriver64.dll audio subsystem works really well with Foobar2000. I select it in the Playback>Output>Device dropdown and the Auralic Vega DAC front panel bitrate display correctly tracks the raw bitrate of the file, including DSD. It will not do this when the generic Windows audio driver is used.
Just my 2¢ worth...
Howie
So while it was on my mind I just went and played some random HDTracks in Foobar2000 which are 192K/24 bit and...the DAC stays on 96K...😡😡 Using the (clunky command-line style) JPLAYmini player the DAC shows the correct bitrate. And interestingly enough JPLAY no longer shows up as a Output device in Foobar2000.
Apparently some Win update or other PC issue has changed the way the JPLAY Driver works. Freakin great. Anyone know of an audio driver that currently works across players correctly?
I thought I had this issue nailed down, but apparently not...There is not enough time in the day to keep track with everything changing!
Beers!
Howie
Using computers for audio sucks, it just happens to be what we have right now in our technological evolution.
I believe this is one of the selling points of Roon (and other similar packages) in that they have gone out of their way to make things work as correctly as possible.
SRC has come a long way over the years, see SRC Comparisons for test results of a wide variety of src's over the years. Atrocious to technically beautiful.
Thank you for the link I will have a good delve into that
I purchased a new Hp laptop for $350 from cinder block store.
1tb, intel icore, 8 Gb.
It's used as an audio analyzer mostly with motu m4 usb ad-da.
But has turned out to be a great music DAC too.
I locked it down too so no automatic w10 updates.
It was plug and play , asio motu driver works great.
My other w10 has been defiled with all sorts of free trial drivers.
Caused jitter and weird bugs with analyzer software etc.
1tb, intel icore, 8 Gb.
It's used as an audio analyzer mostly with motu m4 usb ad-da.
But has turned out to be a great music DAC too.
I locked it down too so no automatic w10 updates.
It was plug and play , asio motu driver works great.
My other w10 has been defiled with all sorts of free trial drivers.
Caused jitter and weird bugs with analyzer software etc.
and yet you can't configure one stinkin' computer that any teenager can?
Computer is an universal appliance, it was never ment to be a SOTA audio playback device out of the box.
If you can't master this part, you should refrain from making statements on sample rates, bit depths and modern digital audio playback technology, I must say.
A general purpose product, like a PC, is very flexible. I prefer not to spend time on it for audio. But for SIM, CAD, Google et al I enjoy. The quality isnt high enough on a PC IMO. So, I am not going to spend time with it for audio.
THx-RNMarsh
Using computers for audio sucks,
Mac and Windows currently want to make it seamless for you to handle everything that comes your way in terms of format. To that end they make a lot of compromises to not have the audio blast out at you at any time. For critical work it sucks. For background listening its "glorious".
To get any control over audio in a critical setting you have to move up to professional or at least qualified semi-professional tools. You need well written drivers that tend to be ASIO these days. And you need to use an interface that doesn't have built in problems, that pretty much excludes USB. There may be some USB/2 audio drivers that aren't bad, not sure, haven't kept up on that. So today its Thunderbolt or a built in card (Lynx, RME, etc) Win 10, as far as I know, has screwed it up even more. And Mac wants to SRC everything you do to the first audio format you used since logging in. Oh yeah.
* deterministic conversion- the incoming set of data defines the outcome in a repeatable fashion (in a perfect world) ie PCM data converted at the base rate.
not deterministic, DSD (random modulator in all conversion) and any up or down sampling, integer or not.
random thoughts
Cheers
Alan
A lot of my feelings are expressed here, as well.
THx-RNMarsh
Richard - " Not for me. Too masochistic."
Yes, it amazes me the engineering workload some of the younger generations will go through to get something realized, like the Raspberry Pi based stuff I see available.
I'm stuck in a paradigm of a very strong valence around real world analog; things I can see, touch, handle and manipulate with my own hands - and perhaps a few common power tools. While the rest of the world races onward with their own implementations using 0201 "ground pepper flake" parts, assembled for them by those contract PCB manufacturers -
I wouldnt be one of the guys trying to 3D print his own speaker cone or waveguide design...
🙂 😎
yes. My thoughts as well. I have lots of really interesting things to do and trying to help others get going. But then, someone needs to move the needle a small step at a time. This is for youth to waste their time on.
I will wait or save my pennies for a more complete professional package to get it done.
THx-Richard
So while it was on my mind I just went and played some random HDTracks in Foobar2000 which are 192K/24 bit and...the DAC stays on 96K...😡😡 Using the (clunky command-line style) JPLAYmini player the DAC shows the correct bitrate. And interestingly enough JPLAY no longer shows up as a Output device in Foobar2000.
Apparently some Win update or other PC issue has changed the way the JPLAY Driver works. Freakin great. Anyone know of an audio driver that currently works across players correctly?
I thought I had this issue nailed down, but apparently not...There is not enough time in the day to keep track with everything changing!
Beers!
Howie
OK,
Fixed for now...it turns out the JPLAY and Auralic drivers had updated to a version not compatible with the older version of F2k (Foobar2000). An update of F2k brought everything back OK.
Glad the resolution (pun intended) was relatively easy. I am back to being able to play everything up to 384k/32bit and DSD512 in Foobar.
Has anyone here heard a decisive advantage of any bitrate DSD over say 96/24 PCM? I am tipping my hand phrasing this way, but would like to hear from others. Also while I was updating the ASIO driver I came across this statement, and wondering if it is still true:
"In the Apple World where BitPerfect users live, playback of DSD requires the use of the DoP (“DSD over PCM”) protocol. This dresses up a DSD bitstream in a faux PCM format, where a 24-bit PCM word comprises 16 bits of raw DSD data plus an 8-bit marker which identifies it as such. Windows users have the ability to use an ASIO driver which dispenses with the need for the 8-bit marker and transmits the raw DSD data directly to the DAC in its “native” format. ASIO for Mac, while possible, remains problematic."
Cheers!
Howie
I thought I had this issue nailed down, but apparently not...There is not enough time in the day to keep track with everything changing!
Beers!
Howie
Been that way since day one with the PC. I gave up. Who has time for that?
Even assuming everything is Okey-Dokey and stable..... how do i get this PC music into my big rig system --- dont answer. Lots of ways but none very good. Lots of compromises.
A dedicated HD DAC which takes a digital input from my file works best IMO.
THx-RNMarsh
Doesnt matter... The results are excellent as implimented and used. Note it was the DAC2 model not the one made now - DAC3 which i use. Maybe use same part on new DAC3. Doesnt matter to me. Its the over-all end result that matters.ES9018 is ancient history.
There will always be a newer part.
THx-RNMarsh
Been that way since day one with the PC. I gave up. Who has time for that?
Even assuming everything is Okey-Dokey and stable..... how do i get this PC music into my big rig system --- dont answer. Lots of ways but none very good. Lots of compromises.
A dedicated HD DAC which takes a digital input from my file works best IMO.
THx-RNMarsh
Hi Richard, I feel the need to answer anyhew...I am using a pretty high-end DAC (Auralic Vega) running from USB, and the secret to performance and stability is most definitely the audio system driver. Although the vanilla ASIO driver worked OK, I have been very happy with the JPLAY driver feeding the Vega, it has been stable and the Vega verifies the bitrate it is receiving without glitching. This F2k update issue is the first time in four years I have had to mess with any software or settings. The only flaw I can find in my setup is perhaps on DSD files: I get a single click at the very start and sometimes at the very end. I think it is file handling by F2k...
Cheers!
Howie
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