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In Reply to: RE: What DACs can play 24/176 or 24/192? posted by jpbeckaudio on October 10, 2009 at 11:36:50
I think the most important attribute a DAC can have is the ability to be slaved by a master clock.
Having said that, I am starting to think Steve was right, clocks do make all the difference. For example a clock like the Antelope audio clock could make a so-so DAC a top performer, and any other DAC that can be use a master clock will never be more than what it is....
24/192 is all about an algorithm smoothing out a digital signal in such a was as to give the sonic quality of more resolution. I for one actually don't like upsampled music, or have yet to hear an algorithm I like...I'd rather use a tone control. gasp, I guess that would void my Audiophile card. Then again, maybe thats what this upsampling thing is all about, being able to alter the sound in a perceived positive way and still tout purity of playback.
Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋
"I for one actually don't like upsampled music..."
I felt like you before hearing cPlay's upsampling, and after a long spell with that, I'm now back in agreement with you. I suspect we are a tiny minority, but my experience might be salutary for others so here goes: The sound of good upsampling algorithms is pleasant and seductive - when you listen to it all the time it becomes addictive, and non-upsampling seems to be lacking something. Upsampling gives the illusion of higher resolution, and most audiophiles seem to like it. As with any addiction, it takes some time with non-upsampling to overcome it. Now when I switch over to upsampling, I hear an artificial smoothness and less defined dynamics, still pleasant, but I'm happier without it.
I have been saying this for years; started with dCS 972. Many software and chipped upsamplers are not very good.
Interesting. Which DAC/soundcard combination are you using? Have you switched to a NOS setup or are you simply bypassing cPlay's upsampling?
"We should no more let numbers define audio quality than we would let chemical analysis be the arbiter of fine wines." N.P.
having tried the upsampling in many different players. Some have sounded dreadful to me; a few are very tempting but ultimately less enjoyable. I've used several soundcards and dacs, but that's not what I'm talking about. To anyone who has been exclusively using player-upsampling for a long time, I'd suggest several weeks without the media player upsampling before comparing again the upsampled v. non. Addictive - not like narcotics, but maybe on the level of sugar-addiction!
I find 44/16 to have a sound characterized by what I'd call tiny "holes" or "gaps" in the midrange. Maybe I'm too aware of the limitations of 44khz sampling, but when I upsample, those gaps are "filled in" to my ear, and sound much closer to real hi res music files.
I think it is actually a complete restructuring based on the 44.1 data, not just a gap-filling exercise. It can never be "the-same-plus-more-detail", but will be new data - always an approximation based on the original 44.1 - therefore upsampling tries to create the illusion of high-res, but loses something of the original as well.
In part, you confirm what I was attempting to say about the addictive effect of upsampling. Going back to 44.1 after a long time with upsampling, you will feel the lack of the illusion of high-res ("holes" or "gaps") for quite some time, like going through withdrawal symptoms. It wears off eventually, and you hear the music for what it is - 44.1k, not high-res, and not fake-high-res. I prefer the vitality of 44.1 - I would like to magically transform all my music into real high-res, but it ain't gonna happen!
If anybody tries the exercise I suggested, and then still prefers the upsampling sound, fair go - it's all a matter of preference. I expect to remain happily in a small minority on this point.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Yes, that is the true high resolution, but apart from fake high-res, there are problems of cost, very limited choice, and impossibility of duplicating great artists and performances of the past.
Trouble is:- some are just other peoples' upsampled files!
and you know this, how?
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Simple, I do FFT analyses of files; some so called hirez material have no content above 21kHz.
"Simple, I do FFT analyses of files; some so called hirez material have no content above 21kHz."
I don't think it is quite so simple.
Some program material itself doesn't have any content about 21 kHz. This will depend on the musical instrumentation, microphone type/placement and recording technology. I've looked at a lot of analog recordings on cassette tape, all known to have been analog mastered. Some of these have high frequency content as high as 23 kHz (on my Nak CR-7a). But others don't. Of course this can depend on the original recording. But the most recent remaster had content up to 23 kHz, as plotted on the Izotope RX display. This was only when a triangle was playing. With acoustic guitar and chorus there wasn't anything above about 19 kHz. I digitize these recordings at 88.2/24 and after any necessary remastering adjustments I downsample to 44/16 for CD release, using the 64 bit Izotope RX SRC.
If you run a SRC with a large window it will time average over a long period and will not capture high frequency energy in transients. For example, a piano recording will have high frequency energy above 20 kHz if closely miked, but only during the transient attack times. Most of the time the strings are simply decaying and their won't be much high frequency content. This is immediately obvious on the spectrographs produced by Izotope RX. If you use a large window on the FFT you will miss most of the transient energy.
IMO, a better way to detect fake hi-res is to look at the recording noise in the silent portions. (Not digital black, but actual room noise.) This will generally be broadband. You will be able to see artifacts of any filters used in resampling, e.g. a sudden drop in noise content immediately below 22050 Hz.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
It is really simple if one uses some measurement and interpretive skills acquired from training and experienece.First, I never use a large FFT Window. What for, when the resolution isn't there? Large Windows were for Apples of 20 years ago with FFT cards.
Second, if you see the 44.1k brick wall filter signature in the FFT for the hirez file, you know it is upsampled from a CD. Some free hirez downloads are like that.
Of course one looks at the noise floor as well as part of the assessment.
Edits: 10/12/09
Simple, I do FFT analyses of files; some so called hirez material have no content above 21kHz.
Imagine that!.....Hmmmmm Hmmmmmm
Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋
Trouble is:- some are just other peoples' upsampled files!
Imagine that....Hmmmmmm
Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋
So if you had to choose, which would it be, upsampling or parametric Eq?
Funny, ever since the "perfect sound forever" format came into being people have been trying to make it live up to being perfect, or at least listenable for the long term. All these DAC's, software programs, upsampling, computer audio, tubed CDP, etc. are all just for one thing....take the digit sound out of digital. Kinda makes someone want to go back to vinyl.
Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋
Neither. I haven't heard an EQ that improves the sound I have now. It's been quite a while since I looked into it (even tried Izotope ozone), so you can tell us when you find the perfect equalizer (mild sarcasm intended).
Vinyl? My music selection would be about 1% of what I have now, and while the sound is great if equipment and records are perfect, I can't afford the equipment, and the records break, scratch, warp, etc., so no thanks.
No such thing as a perfect [anything], but there are some good and great eq's out there...many people including myself like the Behringer DEQ2496 as an eq in the digital domain. I dont like software eq's or upsamplers...having said that I have also heard some good hardware upsamplers. But since I am a purist Audiophile I don't use such things...:)
Scrutiny Strengthens The Truth and Breaks Down Lies 音楽は天国と地球のかけ橋
There is no general solution. If you have a flat system in a good room and play good recordings no equalization should be necessary. Otherwise, it is more the person setting the equalizer than anything else.
For best results, settings will have to be adjusted on a record by record basis, and this can take hours of work per album. While this might be appropriate if you were being paid mastering engineer hourly rates, for an audiophile/music lover this is a waste of time, IMO. Just set up the system so that it sounds good on a wide range of recordings and live with the results. Even if you attend live concerts the equalization will change from seat to seat.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
My 1212m can use an external SPDIF clock and it would be very interesting to try one. Would this approach actually get rid of ALL influence of the PC on jitter? Is an external clock essentially a reclocker that can fix even highs levels of jitter? I know external clocks are usually used to synchronize different pro gear, but is it also a good dejitter solution?
I'm currently using a 176khz Patchmix session with the Foobar sox resampler. I'm very happy with the sound (after making other changes recently) but don't know what the jitter is for this setup. It doesn't sound that bad - I get pretty clean high frequencies.
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