[Nezer]

I haven't had time to make a sample of the sound yet but I have realized the *perfect* description of what I'm hearing.

It hit me in the car this morning...

It sounds as if higher frequencies of some vocals were processed through a ring modulator effect. If you're fimiliar with the sound of a ring modulator listen for that in the super high frequencies. It's especially obvious in human vocals particularly females (but not all, I'm listening to Cranberris and Deloris' voice exhibits absolutly NONE of this effect but then agan, it's run through a lot of muted dark reverb on this album, an effect that works well for her voice and the mood of this album).

 

[Nezer]

Well, I got my files made, both .ape files (Max compression) are in a compressionless zip archive that totals jsut over 800K (which is too large to post here).

Anyone intrested in hearing what I'm talking about send me a PM with an email address taht will accept file attachments and it shall be sent.

I took a phrase from Pearl Jam's Soon Forgot and posted the virgin file (just truncated) as well as one with the cooledit's scientific filter-high-pass filter at 12kHz. Listening to the high-ass version will reveal, in *great* detail what I'm talking about. Listen to both tracks back to back and you should be able to hear it in the full-bandwidth version. The effect is less than subtle on loudspeakers but it's all in your face with some good headphones on.

What I want to know is the quality of those high frequencies through various equipment. Does an $$$ DAC make that more palatable? If it sounds fine through your $$$ setup try something more consumer-like. If you still can't hear it then I have to conclude that you are deaf.

 

[Dusty Chalk]

I've been out for a few days, so not meaning to drag that other thread out, I'll just post my responses in this thread. Out of ignorance, I will probably not participate, though I wish I could get some of the companies who perpetuate "upsampling" (Musical Fidelity, et al) as different from oversampling to participate. In the current issue of Stereophile, Anthony Michaelson of Musical Fidelity chides Stereophile for using upsampling and oversampling interchangeably, stressing that they are different. Sure wish the guys at Stereophile understood this well enough to explain it. Quote:

Originally posted by nec Thank you for trying to explain this to such an idiot like myself...

Sorry about the snideness on my part, was still on "momentum" from arguing with Ricky. You know "quote your sources!" Sincerely, I do apologize. I obviously know just enough about this to be dangerous -- know just enough to think I know more than I do. All I really do know is that they do sound different to me. And upsampling sounds better to me.

I am serious about learning more about DSP -- would the books that you mentioned in that other thread be good books to get? I was also thinking about getting the CSound book. Quote:

Originally posted by Joe Bloggs 1. What have you got to say on the topic of what's the difference between over- and upsampling? Seeing as that article shows that oversampling already includes interpolating intermediate values...

You're kidding, right? I thought I'd stated this, but I'll state it one last time -- it is not my understanding that oversampling includes the kind of interpolation that exists in upsampling. It certainly couldn't have interpolated to 24 bits back in the early 80's, when it was first introduced. I have to believe that that article is incorrect in implying that all oversampling DAC's perform interpolation. I do understand Andre correctly in that they are put through a filter (read: same thing) after oversampling, but it was not my understanding that it was part of the same process. My understanding of the process -- previously -- was that they just repeated the last value. It has since been made clear that they usually just introduce zeros.

Maybe the big difference is the context. Oversampling was always done in a DAC -- it had to be part of the process that after the signal was converted to the analog domain it was put through an analog filter. None of these things were ever taken out of context and expected to sound adequate in an isolated context, the designer was only interested in the overall -- D-to-A -- process. Upsamplers can act as standalone products, so their output has to be an improvement on the input in a wider variety of circumstances. Some upsamplers are D-to-D, some are daughter boards, just as two examples.

But again, and I'll stress this, I'm guessing. This is based entirely on a bunch of advertising literature -- I have yet to find any technical discussion from any of the manufacturers -- and magazine articles. Quote:

Uh, bear with me--so are the OS chips in upsampling DACs set to oversample as well? E.g. In the specs for the ART DI/O I saw 128x oversampling as well as upsampling to 96kHz...

This is another example of using the term "upsampling" in its classic DSP sense, not in the current hi-fi equipment sense. In this sense, it is no different than oversampling.

Again, I blame the manufacturers for using a term that had a long standing meaning in a different sense. But I stand by my belief that they are different. Quote:

Originally posted by MirandaX Strictly speaking, CD audio is not perfect. First, even though 44.1kHz sampling is sufficient to perfectly represent any <20kHz signal given an ideal DAC, there is the 16-bit quantization issue. For loud recordings, it doesn't tend to matter, but for wider range classical (e.g. Mahler's 1st symphony, or the 4th movement of Mahler's 2nd symphony), there is a perceptible difference in resolution as signals get quieter.

Although I agree with Mr. X (

) in his assessment, there is an additional usefulness in going 24-bit -- not just near the noise floor. I also feel that (true) 24-bit recording allows for more separation between instruments, providing better imaging and soundstage as well. Upsampling (to 24-bit with interpolation, not just dithering) should provide some improvement in this arena as well (although this is a glaring case of "you can't create what isn't there").

 

[AndreYew]

Nezer, Kelly,

I brought my Max home this weekend to hook it up to the main system, and listened for the artifacts you guys mentioned. I dropped by a used CD shop and picked up Pearl Jam's Binaural and The Police's Synchronicity (the AMPlus release). I listened to the beginning of Breakerfall for Pearl Jam, and Wrapped Around Your Finger for The Police, paying particular attention to the cymbals. Headphones used were Sennheiser HD-600 and Etymotics ER-4S. Max settings were highest HF boost setting, no crossfeed, and low gain with stepped attenuator almost at maximum level. I connected the Max directly to the outputs of the Sony DVD-S7000 with homemade Canare RCA interconnects.

Pearl Jam: it's just plain distortion. The cymbals were being modulated by something, and everything was so compressed that it wouldn't surprise me that the extreme compression was adversely affecting other things.

The Police: it's definitely a multi-miked recording pasted together in the mixer. Sting's voice sounds like it's recorded in a different room and pasted in on top of the instruments. The cymbals didn't have very good decay, but the relatively mild compression for a pop release was quite welcome. Interestingly, the cymbals had a sound I typically associate with lower-bandwidth, non-flat analog recording. The highs were missing (no sparkle), and the sound and its decay generally sounds truncated.

Have you guys heard any Chesky or dmp recording of cymbals?

--Andre

 

[Nezer]

I don't know that Breakerfall is overly compressed as it is just plain loud! PJ aren't big fans of a lot of post-production crap.

The artifacts I'm hearing are most clearly found on Soon Forgot in Ed's vocals. It's an acoustic track and appears to me to be quite naked.

Process that track through a high-pass filter at 12k and you'll hear the unnaturalness I'm refering to. I doubt it's the recording process unless every producer is making the same mistakes (which is possibly I suppose).

Anyway, try out Soon Forgot and leave Breakerfall alone. The only testing Breakerfall is good for is to see how many dB it takes to make your ears bleed.

 

[kelly]

Heard Chesky, yes. They still owe me two CDs (dammit). ANd yes, their stuff is unebelivably well recorded as is the Telarc stuff I've heard. Telarc has a Bruebeck disc out that is pretty damned incredible in the cymbal department, but you'd have to get more info on that from dparrish.

Synchronicity is actually later Police but a better album so I don't blame you for picking that one. The digital artifacts on sibilants should still be present around Sting's vocals if you listen through the S7000 -- and you'll notice them diminshed through your better DAC equipment. It's comparing it on the two different DACs that should make it obvious.

 

[dparrish]

Hello everyone,

I'm back from my 5-day outing to Colorado

My wife and I enjoyed a break in the COOL mountain air of Breckinridge, which was in between two of the fires (no burning near us but the smoke drifted our way when the wind blew it our direction).

At any rate, I've been catching up on the discussion here and find this topic fascinating, though admitedly over my head (I'm a musician, not an engineer

)

I WOULD like to chime in on Kelly's mention of the cymbals issue.
I am listening right now to the Telarc/Bruebeck 40th Anniversary recording he talked about. I will not attempt to explain the rationale for what I hear, but I would definately say that the cymbals in this recording sound MOST realistic (decay, presence) when the SACD layer is played (it's a hybrid disk, as are all Telarc SACDs). The PCM layer is excellent on my XA777ES, but the SACD layer is a step above this.

I would ASSUME this is mostly due to the added high frequency information DSD/SACD is able to capture/reproduce? I know that most of us can't hear even into the 20khz range, but my assumption has been that one of the main rationales for the hi-rez formats has been research (I can only think of Robert Greene's article series in TAS) that shows that the "unheard" higher frequencies actually do have a psychoacoustic effect on how we perceive/hear.

Again, going only by my personal listening experience, I have noticed that the Chandos 24/96 Vaughan Williams Symphony No. 2 recording I have (it was recorded in 24/96, but I guess downsampled {is that the correct term?} to 16 bits for the PCM pressing) sounds less "digital" than other recordings I have, even on the same label (I know this is not a fair assessment, as I do not have the same recording in a lower bit to compare with).
I also have a 24/96 remastering of the historic 70's Horowitz/Rachmaninoff Piano Concerto No. 3 (BMG/RCA Red Seal, I think), which sounds quite good to my ears (I have the original vinyl release as well). Whether the improved sound of these is due to the higher bit rate or a difference in mixing, I don't know (the Vaughan Williams was recorded in 24/96, the Horowitz was "remixed"), but the sound is one notch above average PCM sound to these ears.

To my deductive but still ignorant mind on these matters, the conclusion that I would come to is that the differences in recording/format/mixing must have something to do with the differences in sound, especially since I have played all of these disks on the same equipment. Again, I realize I have very little "hard" evidence to back up my claim (other than the numerous hybrid disks I own, where the SACD layers consistently sound better that the PCM layers) and my "testing" methods are VERY unscientific, but nevertheless, that's my "take" at this point (until someone can convince me otherwize

)

 

[AndreYew]

Nezer,

I'll try out the track you recommend. BTW, the reason Breakerfall sounds so loud is because it's been heavily compressed.

Kelly,

I did listen to Sting's voice as well, and it sounds like it's just been overly processed. I noticed the sibilants on his voice, but chalked that up to bad production. The reason I brought up the Chesky cymbals was to show an example that 44.1 kHz PCM isn't necessarily flawed, because of flawed cymbals.

--Andre

[edit for clarity]

 

[aos]

Marc Heijliger's DAC project

Erland Unruh's DAC

Heh, I have both

.

Ok, not completely true. First, I have only the DAC/analog section which was designed by Erland's friend Eric, not the actual resampling digital board. That "only" all-discrete amp cost me probably thousand in parts for bulk metal foil resistors, Multicaps and Black Gates and high precision opamps. I talked to Unruh, but he still hasn't redesigned board for anything but 44.1kHz and I didn't have money (or will to pay) for his PCB / custom programmed chips (!!).

But I DO have fully assembled Marc Heijligher / Guido Tent etc. digital board right beside my monitor. Supposedly using all-time best parts for digital filters and DAC.

I'm missing power supply and analog section. Gotta use up those now obsolete but all time favorites, EL2008...

Once I'm done with it I won't care about redbook anymore.

I really want to hear this thing...

$100-$200 clock upgrade can probably give you very close approximation of this, assuming CD player is good stuff to begin with.

 

[Joe Bloggs]

Are you implying that if we eliminate jitter, we've already achieved 90% of all that these fancy DACs aim to achieve?

 

[aos]

Maybe not 90% of what they strive to achieve, but probably 90% of total sound improvement

. But that's all subjective. I'm just saying you don't need to have two dozen kilobucks or spend a year doing advanced DIY to improve your sound a lot.

However, total reclocking like those two DACs do is NOT an easy task to design! It's a non-trivial piece of engineering and not a black box you just pick off the shelf, like a DAC chip or opamp.

 

[AndreYew]

Joe,

The value of Erland's and Marc's DAC designs is in the way they control noise, of which reducing jitter is a small subset.

--Andre

 

[aos]

Those two DACs have - in part! - different approaches to noise elimination. One advocates use of multiple parallel capacitors to bypass digital supplies, the other use of only one high frequency capacitor (such as OS-CON or polymer). Someone on Headwize posted a very nice link to an application note describing bypassing to the stuff that runs at hundreds of MHz all the way to 1.7GHz - in cell phones. There you DO use multiple parallel capacitors of different properties, but they are chosen so that each eliminates one of spectral components present in the signal, and eliminates it by having a resonance present around that frequency, a resonance that exists because of its parasitics! At 10-20MHz I would side with Guido Tent that one high frequency capacitor, used as a capacitor, not as a resonant circuit, would suit better. But I'm just talking from logic point of view, all these guys have expensive test equipment and I have nothing

.

 

[kelly]

Quote:

Originally posted by AndreYew Kelly, I did listen to Sting's voice as well, and it sounds like it's just been overly processed. I noticed the sibilants on his voice, but chalked that up to bad production. The reason I brought up the Chesky cymbals was to show an example that 44.1 kHz PCM isn't necessarily flawed, because of flawed cymbals.

Andre
The stuff you're calling "overly processed"--AB it between the S7000 and a higher end player/DAC and let me know what you discover.

 

[AndreYew]

Aos,

The problem with OSCONs is that they will have a resonance because there's always some ESR and ESL in series. The beauty of the multi-cap solution is that you can make the impedance resistive beyond what frequencies you need. The OSCONs by themselves will have resonances much lower than what a multi-cap solution will, and will dictate what your resonances are instead of having you control where they are. Very astute of you to pick up on this difference in philosophies, because Pete Goudreau, the designer of Erland's triple-cap bypass scheme, does not like OSCONs for that very reason, and that's why he did his multi-cap scheme.

Kelly,

Will do.

--Andre