[rgs9200m]

Is there any technical reason why physical SACD discs sound (at least to me) magnificent, but high res or DSD files don't make it to this level to my ears?
SACD combines extreme detail and speed and tight bass but avoids any dreaded over-the-top ringing or glare in the upper mids and vocals that annoys me in digital.
(I know this the Dave thread and I don't have one, just a Hugo TT, but there seems to be a major DSD tangent here, so I wanted to ask since I've been an SACD junkie and believer since it started in 1999.)
 
One more thing is that, with an SACD player, generally there is no EQ possible, and I find that none is needed.

 

[ecwl]

I think the short answer to your question is probably yes. Because most DACs you've listened to for high-res or DSD files actually convert DSD files into PCM first whereas your SACD player is probably playing the DSD stream as the original DSD stream using pulse-width modulation (PWM). There are still DACs that play DSD files like your old SACD players, e.g. T+A DAC8, Teac UD-503, etc but they would play PCM as PCM. Moreover, there are DACs that convert PCM to 2xDSD and convert DSD to 2xDSD (so not the original waveform), e.g. Emm Labs/Meitner DACs, PS Audio Directstream, Playback Design DACs.
 
One of the discussions in this forum in the past was that even though most admirers/users of DAVE believe DAVE (or even Hugo TT) would reproduce the most accurate analog waveform from PCM or DSD files, we have noticed some people prefer DACs of other designs. DAVE's final output uses the Pulse Array DAC which can be more accurate with less analog distortions. But it's possible that your ears and your gears just prefer music played back using pure DSD/PWM method. And ultimately, if you like one of those DACs over Hugo TT/DAVE, it's your money.

 

[GreenLeo]

  I've dabbled with HQPlayer converting redbook FLAC files into DSD into my iFi iDSD. Even with that low cost portable player driving my trusty old HD600's, the superiority of HQPlayer over JRiver was very obvious at any upsampling/filter option, but converting to 256DSD and upwards really made this combination sing for me, although some on the iFi thread preferred highly upsampled PCM to any DSD. For me though, DSD just sounded more natural and lifelike.
 
The trouble was, some of those HQPlayer options put a real load onto my laptop's processor causing the fan to kick in horribly. And the higher rate DSD conversions were by far the worst for loading the processor. Also, the HQPlayer's user interface is an acquired taste, so it can all be hard work to get the best, but what it did prove to me was that manipulating the original redbook file can work wonders.
 
The more adventurous that liked HQPlayer thought that the exotic and obscure BugHeadEmperor was even better. I never tried this one because just reading the summary of the user instructions gave me a headache. Apparently BHE fundamentally reconstructs the original file, rather that merely upsampling and filtering, but I didn't delve deeply enough to really understand what it was all about. It's a strange world out there.  

BHE does sounds better than the HQPlayer according to the forum in JPLay (JPLAY - hi-end audio player for Windows→ Forum→ Computer Audio).  There were lots of posts and many mentioned that they changed from HQPlayer to BHE.  BHE is not that user friendly but not difficult to use.  It is relative Buggy and has OS and relative strong H/W requirements.  In terms of SQ, its better than Foobar and JPlaymini (I've never tried HQPlayer).  It's a player that requires patience to use because it crashes relatively easily and to learn how to use it because lots of tweaks may be done for all its worth.

 

[Christer]

  Is there any technical reason why physical SACD discs sound (at least to me) magnificent, but high res or DSD files don't make it to this level to my ears?
SACD combines extreme detail and speed and tight bass but avoids any dreaded over-the-top ringing or glare in the upper mids and vocals that annoys me in digital.
(I know this the Dave thread and I don't have one, just a Hugo TT, but there seems to be a major DSD tangent here, so I wanted to ask since I've been an SACD junkie and believer since it started in 1999.)
 
One more thing is that, with an SACD player, generally there is no EQ possible, and I find that none is needed.

I partly agree with you.Especially when  comparing the rbcd layer and SACD layer,there was a wonderful relief to get away from "any dreaded over-the-top ringing or glare in the upper mids and vocals that annoys me in digital." as you say and which has always annoyed me a lot too.
SACD was a blessing for quite a few years to me too.
None of that  horribly annoying top ringing or that  pcm glare was present even from  well pressed good old LPs.
They were problems introduced with digital and pcm.
Unfortunately  those ugly not natural abberrations and  aspects of PCM rear their ugly head more often even with hi res pcm than with DSD to my ears.
But nice and sweet and beautiful as natively DSD recorded SACDs can sound via an SACD player that plays DSD natively,SACDs are still  limited to DSD 64 and judging from what I have heard so far from DSD 128 at least, DSD 256 or in the future DSD 512 or even higher  could well be what DSD needs to become a real contender again.Or maybe not?
DAVE seems to rule as far as HD  PCM is concerned but does it really squeeze all that is possible out of higher sampling rate DSD?
Maybe there isn't any more to be squeezed out?
I know some people in the recording business,who are against the   "truncation and decimation" approach with DSD and prefer DSD natively over any PCM processing approach.
People who can  regularly compare live to recorded sound.
Although they often sound a bit  less resolved to me personally, I can listen for longer hours and sometimes with more joy to DSD recordings than many  PCM ones which tend to induce listener fatigue more often and quicker than DSD even in hi res form.
But if I buy DAVE  I may find my priorities changing more than they have via my Hugo.
 
Both 24/96 , 24/176.4 and 24/192 and DXD sounded sublimely realistic via DAVE to me with good recordings.
And very revealing of the faults with badly multi mic'd ones.
I suppose in the end it is one's musical enjoyment  that counts most.
Surprisingly,Rob Watts  himself seems to prefer a "less resolved", "softer", "slower, "papercone" dynamic headphone over planars, for much the same reasons some of us  here obviously ,still like listening to SACDs and DSD files in spite of their theoretically lower resolution and limitations.
DSD at its best sounds quite close to what I normally hear from a midhall seat with the warmth and bloom and harmonics from acoustic instruments playing live.
And DXD   tends to present more of a "warts and all" on stage  sound .
My dream is of course of  a digital recording and reproduction chain that would bring me back to the real thing when my limited memory after a while, fails to bring me the  live concert fresh again in all its glory.
Cheers Chris and  keep enjoying both  your SACDs and Hugo TT.

 

[Rob Watts]

   
With digital VCs isn't also the analog noise floor of the DAC important even if the digital noise floor is at -385dB? I mean, as you attenuate digitally the signal gets closer to the analog noise floor of the DAC which remains constant, meaning reduced SNR. The quoted 127dB dynamic range translates to a little over 21 bit analog performance which would give you 5 bits of attenuation or 30dB before starting to lose resolution for 16 bit signals. But what happens with a 24 bit signal? Just those 3 extra bits over the analog resolution of the DAC to throw away for VC before starting to affect the sound quality?

Absolutely - but we have two problems to worry about. One is fixed (unvarying) analogue noise. The second is distortion and noise that is signal dependent and applies to small signals and conventional DAC's (particulalrly R2R or ladder DAC's) have big problems handling small signals - they add distortion and noise, so that as the digital volume control is applied, more small signals get distorted and resolution is lost.
 
In the case of ladder DAC's, the resistors can never be matched well enough, so there is always very substantial distortion with small signals.
 
In the case of conventional multi bit noise shapers (DSM) they can have better small signal non-linearity (it depends upon the DAC architecture) but have amplitude errors for small signals - as the signal gets closer to the noise shaper's noise floor, the wanted signal gets attenuated.
 
In the case of DSD, we have even bigger problems with noise shaper resolution, plus RF noise, plus jitter sensitivity, plus idle pattern noise....
 
Now because DAC's have severe small signal distortion and resolution issues, people naturally assume that dynamic range is an indicator of how well the DAC handles small signals in terms of resolution and distortion and noise, and forget about the fixed noise aspect, as this is seen as being small.
 
But with Dave we have something very unusual - the dynamic range is totally dominated by the small signal fixed analogue noise. There is no DAC noise, nor is there any distortion of small signals. And I talk about the digital domain performance not because you will measure it on the analogue side - but because it was important subjectively, and it was interesting that the brain could detect such minute errors. So small is the distortion, that when I increase the signal level from -60dB (the usual setting for dynamic range), I get exactly the same -127 dB at 2.5v output - so measurement wise, the fixed noise (hiss) dominates everything. And from a SQ point of view, if you can't hear the hiss, then it has no bearing whatsoever.
 
Rob

 

[romaz]

  Regarding your question whether Chord listen to the competition, I would say I wish more  DAC designers than Chord, listened to real live acoustic music instead  of tuning and trying to  cover up faults in  their gear according to expected subjective preferences among their customers.

I agree with you on this.  When I listen to a demo setup up by a DAC manufacturer, the first thing I pay attention to is what music they demo and very often, they are using a studio recording that potentially sounds the same on all DACs either because it has no real depth in the recording or has so much reverb mixed in that it sounds holographic on all DACs, whether it is on a $100 DAC or on a $10,000 DAC.  A good example of this is Rebecca Pidgeon's performance of Spanish Harlem for Chesky Records. 
 
With many DAC presentations, the manufacturers frequently talk about things like accuracy of timbre and detail and dynamic range and PRAT but it's much less commonly they talk about the DAC's spatial abilities, especially depth.  Even with DAC reviews that I read, the reviewers don't seem honed in on this quality.  Well, here's what Rob had to say recently:
 
"Generally I have two primary motivations - getting closer to the un-amplified sound I hear in concert halls..."
 
"Will we get lifelike depth reproduction? I want to hear an organ at 100m away sounding like its 100m away in your living room."
 
"Yes, the sound of real acoustic instruments playing in real space is the only absolute reference."

 

[yellowblue]

Tyll finds the Ether C "overwhelmingly musical". And I just can conclude - overwhelmingly musical epescially with the Dave!
 
Romaz, how about your HD800S after some more days of burn in?

 

[romaz]

  I've dabbled with HQPlayer converting redbook FLAC files into DSD into my iFi iDSD. Even with that low cost portable player driving my trusty old HD600's, the superiority of HQPlayer over JRiver was very obvious at any upsampling/filter option, but converting to 256DSD and upwards really made this combination sing for me, although some on the iFi thread preferred highly upsampled PCM to any DSD. For me though, DSD just sounded more natural and lifelike.
 
The trouble was, some of those HQPlayer options put a real load onto my laptop's processor causing the fan to kick in horribly. And the higher rate DSD conversions were by far the worst for loading the processor. Also, the HQPlayer's user interface is an acquired taste, so it can all be hard work to get the best, but what it did prove to me was that manipulating the original redbook file can work wonders.
 
The more adventurous that liked HQPlayer thought that the exotic and obscure BugHeadEmperor was even better. I never tried this one because just reading the summary of the user instructions gave me a headache. Apparently BHE fundamentally reconstructs the original file, rather that merely upsampling and filtering, but I didn't delve deeply enough to really understand what it was all about. It's a strange world out there.  

I think it's always a good rule to do what you ears tell you is best but here is what Rob had to say about this matter.  It was made in reference to the Mojo but I believe this should be equally applicable to the DAVE (probably even more so):
 
"Converting the original file into DSD or up-sampling is a very bad idea. The rule of thumb is to always maintain the original data as Mojo's processing power is way more complex and capable than any PC or mobile device.
 
DSD as a format has major problems with it; in particular it has two major and serious flaws:
 
1. Timing. The noise shapers used with DSD have severe timing errors. You can see this easily using Verilog simulations. If you use a step change transient (op is zero, then goes high) with a large signal, then do the same with a small signal, then you get major differences in the analogue output - the large signal has no delay, the small signal has a much larger delay. This is simply due to the noise shaper requiring time for the internal integrators to respond to the error. This amplitude related timing error is of the order of micro seconds and is very audible. Whenever there is a timing inaccuracy, the brain has problems making sense of the sound, and perceives the timing error has a softness to the transient; in short timing errors screw up the ability to hear the starting and stopping of notes.
 
2. Small signal accuracy. Noise shapers have problems with very small signals in that the 64 times 1 bit output (DSD 64) does not have enough innate resolution to accurately resolve small signals. What happens when small signals are not properly reproduced? You get a big degradation in the ability to perceive depth information, and this makes the sound flat with no layering of instruments in space. Now there is no limit to how accurate the noise shaper needs to be; with the noise shaper that is with Mojo I have 1000 times more small signal resolution than conventional DAC's - and against DSD 64 its 10,000 times more resolving power. This is why some many users have reported that Mojo has so much better space and sounds more 3D with better layering - and its mostly down to the resolving power of the pulse array noise shaper. This problem of depth perception is unlimited in the sense that to perfectly reproduce depth you need no limit to the resolving power of the noise shaper. 
 
So if you take a PCM signal and convert it to DSD you hear two problems - a softness to the sound, as you can no longer perceive the starting and stopping of notes; and a very flat sound-stage with no layering as the small signals are not reproduced accurately enough, so the brain can't use the very small signals that are used to give depth perception.
 
The second issue in using the transport to up-sample (44.1 to 176.4 say) is that the up-samplers in a PC or mobile device are very crude, with very limited processing power and poor algorithms. This results in timing problems, and like with DSD you can't hear the starting and stopping of notes correctly. These timing problems also screw up the perception of timbre (how bright or dark instruments sound), the pitch reproduction of bass (starting transients of bass lets you follow the bass tune), and of course stereo imagery (left right placement is handled by the brain using timing differences from the ears). Now Mojo has a very advanced algorithm (WTA) that is designed to maximise timing reconstruction (the missing timing information from one sample to the next) and huge processing power to more accurately calculate what the original analogue values are from one sample to the next. Its got 500 times more processing power than normal, and this allows much more accurate reconstruction of the original analogue signal.
 
So the long and the short is don't let the source mess with the signal (except perhaps with a good EQ program) and let Mojo deal with the original data, as Mojo is way more capable.
 
Rob"

 
Here is another post by Rob relating specifically to the DAVE:
 
"Oh dear. Do NOT use your computer to up-sample or change the data when you use one of my DAC's.
 
All competent DAC's up-sample and filter internally; the issue is how well that filtering is done, in terms of how well the timing of transients is reconstructed from the original analogue. Computers are poor devices to use for manipulating data in real time as they are concurrent serial devices  - everything has to go through one to 8 processors in sequence. With hardware and FPGA's you do not need to do that, you can do thousands of operations in parallel. Dave has 166 DSP cores with each core being able to do one FIR tap in one clock cycle. That is incredibly powerful processing power way more powerful than a PC.
 
But its not just about raw processing power but the algorithm for the filter. The WTA filter is the only algorithm that has been designed to reduce timing of transients errors, and the only one that has been optimised by thousands of listening tests.
 
Rob"
 
Lastly, much has been made about Rob not liking DSD.  He has said it is a flawed format and gave his reasons above but he has also said that if something is recorded in DSD, it should be kept in DSD and not converted to PCM and vice versa.  Here is his response in blue after he was asked the following question:
 
Rob... should I take it that your better off sticking with the format it was originally released in... if it was recorded in Red-Book stick to Red-Book & vice-versa DSD?
 
"Yes of course, best to stick to the format the master tape was on. But Floyd would be on analogue tapes, so we have a legitimate choice, assuming the DSD and the CD comes directly from the analogue master tape. I have heard Norah Jones on DSD and it sounded dreadful - I think they just converted CD to DSD, which is just plain dumb. This is always the problem with recordings, you do not know what you are getting!"
 
Rob"

 

[romaz]

  Tyll finds the Ether C "overwhelmingly musical". And I just can conclude - overwhelmingly musical epescially with the Dave!
 
Romaz, how about your HD800S after some more days of burn in?

The HD800S continues to improve.  The HE-1000 is still dimensionally fuller, especially with respect to air around the instruments and with depth and for my tastes, it is still the more musical (more evocative) headphone and easier to listen to for long stretches of time but the HD800S definitely does certain things better than the HE-1000 and for certain tracks, it is preferable to the HE-1000.  I am finding they have complementary strengths although I am waiting to re-evaluate an Abyss to see which one I like better because I likely will keep only one of these two.

 

[romaz]

As a follow up to my post #2363 above, I own several native DSD256 acoustical recordings that are truly spectacular and demonstrate wonderful and realistic depth on the DAVE in DSD+ mode and so I don't believe that DSD always has to sound flat but Rob did also mention that the quality of the recording is much more important than the format and that is probably the case with these recordings meaning it's the quality of the recording that makes them sound so great and it's possible they could sound just as good (maybe better) had they been recorded in DXD or even 24/192 PCM.

 

[Christer]

  I agree with you on this.  When I listen to a demo setup up by a DAC manufacturer, the first thing I pay attention to is what music they demo and very often, they are using a studio recording that potentially sounds the same on all DACs either because it has no real depth in the recording or has so much reverb mixed in that it sounds holographic on all DACs, whether it is on a $100 DAC or on a $10,000 DAC.  A good example of this is Rebecca Pidgeon's performance of Spanish Harlem for Chesky Records. 
 
With many DAC presentations, the manufacturers frequently talk about things like accuracy of timbre and detail and dynamic range and PRAT but it's much less commonly they talk about the DAC's spatial abilities, especially depth.  Even with DAC reviews that I read, the reviewers don't seem honed in on this quality.  Well, here's what Rob had to say recently:
 
"Generally I have two primary motivations - getting closer to the un-amplified sound I hear in concert halls..."
 
"Will we get lifelike depth reproduction? I want to hear an organ at 100m away sounding like its 100m away in your living room."
 
"Yes, the sound of real acoustic instruments playing in real space is the only absolute reference."

Thanks for your response romaz,
Yes before my posts annoyed him  Rob  told me those same things  too.I found them very  laudable indeed, and they  are  generally what I hear  more of from his DACs compared to several others.
Regarding music choice,I personally ALWAYS bring my own  reference files of acoustic music and my MPO ,not the Malaysian Philharmonic Orchestra,but  my laptop, to any audio audition,where I also know how things sounded live in the hall at sessions and  unedited raw  playback in the studio plus  lots of other well recorded symphonic and operatic material solo voice and piano and politely refuse the standard  electronica demo stuff used to impress with  certain products but often of very  limited musical value to me anyway because I don't listen to the popular genres.
I remember  when Esa Pekka Salonen was  asked what he thought of modern  pop music in general and he answered "it tells me nothing!"
Me neither.
But he told me he was quite a fan of SACDs compared to rbcds with the big company he was recording for in those days, DGG.
 
Currently I am most  tempted to get  the HE1000 to go with my Hugo for the time being.
It sounded pretty good even via little Hugo.I was supposed to audition it again today, but a more pressing need to go to the dentist plus the fact that there are no real HI FI shops at all in central KL stopped me from doing so today.
It would have taken me up to an hour and a half even to get to a dealer who stocks them here way out in the suburbs of huge  KL.
And in a few hours there is a live  concert to attend again.
Regarding depth perception organs are rarely listened to from as far as 100m away in a  real situation.
The distance from front to back in an orchestra on stage is rarely more than 20-30 metres.
And basically ALL recordings of classical music are made with the main stereo pair of  mics hanging slightly in front of and above the orchestra.
There are no modern  commercial recordings I am aware of made from a mid hall perspective as far as miking is concerned.
The only mics mounted mid hall you would see at sessions are those used for picking up ambience at mch ie surround recordings.
In fact in most cases the mics are among the musicians in the orchestra for most multitrack/multimic'd productions and they are almost always far too many and far too close imho.
But time is money and they all want to be on the safe side.
The width with a large orchestra far exceeds depth. But  my old Linn Sondek  LP12/Supex mc  and my electrostatic speakers  are  fully capable of setting the part of the orchestra that doesn't fit inside my living room well into my back garden and well outside on the sides.
Good old LP is very capable of portraying realistic  depth and width from simply mic'd  well recorded albums. Often more so than most modern digital which generally  tends to sound  flatter the lower the resolution is to me.
I  hear clearly more "air" around instruments from DSD64  than from  rbcd for example.
Loosing depth perception was one more of the real drawbacks like ringing and hardness, of early digital. Rob Watts seems to be  busy bringing us those things back even with pcm luckily.
But I think he needs a bit more assistance from those actually doing the recordings to get it completely right in the end.
To me it seems as a lot of what we are striving for here could be relativley easily achieved with a  simple  Blumlein two mics approach and a good orchestra playing in a good hall already in the late 50s.
Listen to some Karajan Philharmonia late 50s early 60s EMI recordings on LPs and voila!
Limited dynamic range yes , surface noise yes, a bit thin string sound yes,but inspite of all limitations there is both depth and space rarely achieved in such clarity today with many mics and pcm.
For good realistic  stereo you need no more than two mics, and for mch up to five no more is really necessary imo.

 

[lovethatsound]

Tyll finds the Ether C "[COLOR=333333]overwhelmingly musical". And I just can conclude - overwhelmingly musical epescially with the Dave![/COLOR]

[COLOR=333333]Romaz, how about your HD800S after some more days of burn in?[/COLOR]

Hi yellowblue
Like you i have the hd800, not modify and i love them with the Dave,but I've add my eye on the Ether C for a while now,could you do a bit of a review with the Ether C used with the Dave.

 

[yellowblue]

Not easy for me to answer because the Dave has 100 hours on it and the Ether C has just 30 hours and I am sure that it is not fully burned in yet. 
 
The HD800 sounds really smooth without loosing any detail. Fantastic soundstage and air. Never heart the HD800 with so silky treble. Could even to some extend depend on my cable (Effect Audio Eros). I don´t miss the HD800S (which I never heart) at all because I think that the Dave makes the HD800 sounding at their best. Never had a headphone which was so depending on source, amping and cable. 
 
The Ether C yet sounds a bit bright (I have to experiment with the pads when it is fully burnt in). Piano for example is not as clean sounding as on HD800, and a little fatiguing. But it should change when the Ether C is fully burned in. The Ether C hits a bit harder than the HD800 (awaiting the bass to improve furthermore and be clearly better than the HD800s bass). It´s more "you-are-there-feeling" over the Ether C. Voices are more present and where a choir sounds airy and refined with the HD800, but more distant you get a feeling that the microphone is stuck in the middle of the singers with the Ether C. 
 
The Ether C for me is more involving but not as refined as the HD800. The treble with the Ether has more sparkle but is not as airy and silky. Soundstage is not as wide but you have a feeling that you are getting nearer to the music. 
 
Maybe I wrote more than I should at this time and my Ether C will transform to something that will better my HD800. But I myself expect that the HD800 will stay more refined with the Dave as source whereas the Ether C will sound more musical and involving. I am sure that they will complete each other. Not selling one of the two.

 

[Priaptor]

Rob

Unfortunately I can't get a unit to try but am getting ready to "pull the trigger" on the DAVE. My plan is to use it as my go to for my headphones based on what I currently get from HUGO + Cavalli Carbon. I will obviously give it a comparison to my main systems MSB Diamond DAC and let the chips fall wherever they fall.

My question which is more bland than all the technical stuff is what your perspective is for the (very expensive) matching stand?

Thanks

 

[shuttlepod]

Priaptor:  Unless money is no object and you think the matching stand is just the sexiest thing ever, I would save the considerable money and spend under $500 for good isolation. That might be Stillpoints or Nordost isolation footers or the Acoustic Revive stand or a lot of other isolation tweaks that have gotten decent reviews. I like Stillpoints myself.