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NOS DAC sound more natural


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It could just be a matter of product evolution too. Perhaps the output stages on these NOS DACs (and IMO, NOS seems to start showing up more once you crest the $2k threshold, so IMO starting to get into the lower tier of "high end" components), along with power supplies and just chip implementation themselves are better than what were in those original NOS CD players.

 

 

I have no doubt there are better components in today's NOS DACs than in early CD players 25-30 years ago, so that should help contribute to better sound. Some of the expense of NOS DACs stems from the fact that R2R chips are more expensive, which is one of the major reasons sigma-delta chips took over the market.

 

I don't know whether aliasing distortion was higher in the early CD players. It may well have been, and they had jitter problems as well. Still, even with better components some amount of aliasing is pretty much unavoidable in a "brickwall" filter just from sheer mathematics.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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  • 2 weeks later...

Note that a lot of hi-res files are actually upsampled from 44.1khz and not recorded at hi-re, it's a scam really. With a NOS R2R Dac the difference is actually fairly obvious, analog vs digital sounding basically. Upsampling with those pre and post ringing really destroys the music.

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Using DSD end-to-end allows A/D and D/A conversions without intermediate sampling rate conversions and specifically allows to avoid rate down conversions which introduce ringing.

 

Precisely. Most of today's "PCM" recordings are downampled from Delta Sigma Modulators running at megahertz sampling rates, to as low sample rates as 44.1kHz. While those recordings shouldn't have been downsampled to 44.1kHz in the first place, keeping the recordings at this sample rate during playback (in NOS mode) is sure to increase aliasing distortion. In the end, one gets a recording with ringing, and loads of aliasing images to boot.

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What you failed to realize is that you can't mix DSD and all sound goes through 44/48khz during mixing (96khz if you are lucky, or unlucky if they are 44->96 then back). Then they upsample it again and that is what you have. Those DSD recording sounded horrible as a result versus PCM in practice.

 

The sound out of R2R NOS DACs sounds infinitely better than DSD DACs playing PCM or DSD. Just go buy a Directstream DAC if you really want to learn how bad it sounds. The price is crashing, now below 3k as people are waking up.

 

DSD is a bad idea that needs to die already.

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What you failed to realize is that you can't mix DSD and all sound goes through 44/48khz during mixing (96khz if you are lucky, or unlucky if they are 44->96 then back). Then they upsample it again and that is what you have. Those DSD recording sounded horrible as a result versus PCM in practice.

 

There are ways to avoid digital mixing by recording direct to DSD, doing on-site analog mixing before making a digital capture, or capturing the output of an analog console.

 

Delta Sigma Direct recording is a very good idea, when you consider the fact that 99,9% of modern converters employ delta sigma modulation anyway.

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The sound out of R2R NOS DACs sounds infinitely better than DSD DACs playing PCM or DSD. Just go buy a Directstream DAC if you really want to learn how bad it sounds.

 

AFAIK, Directstream is not a NOS DSD DAC (it internally oversamples and downsamples all signals, including DSD).

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What you failed to realize is that you can't mix DSD and all sound goes through 44/48khz during mixing (96khz if you are lucky, or unlucky if they are 44->96 then back). Then they upsample it again and that is what you have. Those DSD recording sounded horrible as a result versus PCM in practice.

 

Yes you can mix DSD/SDM, if I can do it (I do it for things like 5.0 channel to stereo downmix), others can do it too.

 

No it doesn't. Sonoma workstation processes DSD at native rate. Merging Pyramix allows editing in DSD and mixing and more complex operations at DXD (352.8 kHz rate).

 

Do you realize that practically all new PCM material is recorded using oversampled sigma-delta ADCs and the PCM you get is down-conversion of DSD-like data coming out of the actual A/D conversion stage?

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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The sound out of R2R NOS DACs sounds infinitely better than DSD DACs playing PCM or DSD. Just go buy a Directstream DAC if you really want to learn how bad it sounds. The price is crashing, now below 3k as people are waking up.

 

DSD is a bad idea that needs to die already.

 

Mods and members should note that Coli (or 'ilok') has been trolling and crapping DirectStream threads.

 

There are any number of fantastic implementations of NOS R-2R, R-2R with filter, all PCM and DSD converted as DSD (Playback Designs, DirectStream, etc), and so on.

 

Here's a list of R-2R DACs (flagships) on the market:

 

Aqua HiFi La Scala

PCM1704-K (4) Dual mono

 

CH Precision C1

PCM1704 (8) Dual mono

 

Computer Audio Design CAD 1543

DAC Philips TDA1543/N2 (16)

 

Metrum Acoustics Pavane

Propietary "Transient" chips

 

MSB DAC V

Discrete (proprietary)

 

Phasure NOS1

1704U-K (8)

 

Schiit Yggdrasil

AD5791 BRUZ (4)

 

totaldac d1-twelve

6 discrete ladders per channel (600 vishay foil resistors)

Trinity Electronic Design Trinity DAC

PCM1704

 

Vertex AQ Aletheia DAC-1

Philips TDA1543

 

Not all are NOS (Yggy, for example).

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  • 1 year later...
Finally got to audition a new NOS DAC using 1704Ks. It is a good one. Not rolled off, slow, or thick.

 

Has that extra bit of body and tone. How noticeable a difference as compared with the up/over sampling DACS in house comes down to what is playing at the time, but it is consistent. It does sound a bit more natural.

 

Can someone tell me what this extra tone is made of? Is it a result of distortion, or superior processing? Are the other DACs imparting their own sound through the upsampling and oversampling at work? Is the NOS DAC more accurate? The sound difference sort of reminds me of a tubes/solid state kind of thing, but not exactly. I don't recall hearing this type of tone from any other DAC that I have had in the system.

Well, this is my experience with a NOS, filterless DAC.

It is the so much praised (and priced) in its moment Luxman DA-07, a beast of discrete electronic components (28 kg weight) that only converts redbook CDs, 16/32k and 12/44,1k/32k formats.

The very first time I heard it at home (served by its companion, the monstruous DP-07 CD spinner) I just thougt that I've changed my whole equipment (carefully selected piece by piece and no cheap at all) an upgraded it to a league apart.

This took place at the very end of the '80s, and since then it has a permanent site of honor (and continuosly listening pleasure) in my audio gear.

I've changed other parts (the power amps, for example) but not, by no means, the main source, even with the arrival of hi res material. Not, because (and this can seem weird to someone) I still have to listen to (including SACD re-editions of CDs I already have) something that can give me the sense of immediacy, spaciousness and (in summ) truthfulness my beloved combo DP-07/DA-07 provides me. Of course, I'm not limited to redbook material, and I regularly purchase hi res music on-line, wich I play with a highly (and costly, to increase my triumphant way to bankrupcy) customized soundcard feeded from the PC. Sounds very good, indeed, and no complains with it, but it lacks that ultimate point of "credibillity" my old CD system has.

This is, with my Mission 767 semi-active loudspeakers and (don't laugh) my Cyrus II SE integrated amp + PSX additional power supply in my second and much smaller equipment, the three best buys in audio I've ever made.

 

VenturaRV

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I think a NOS DAC can be helpful, but everything is system dependent. My system has a tubed preamp and amp. I purchased a Metrum but preferred my (sabre based) Eastern Electric with discrete opamps. The sabre DAC provided more clarity without sounding the least bit harsh. I imagine some folks with solid state electronics may prefer a NOS DAC instead. One thing I've learned is you rarely can speak in generalities when it comes to audio playback. Existing equipment, room, and power all can influence end result. Trust your ears.

My DAC is NOS and filterless, and my pre and power amps are not solid state but tube.

My former CD player was a high quality Sony "bitstream" machine, and I tryed two or three oversampling models (they appeared in the market quite a long time ago, not today), and my final decission was, without a hint of a doubt, the NOS filterless system.

I choosed it because it sounded far more natural, no matter my amp section was (and is) valve and not SS.

 

VenturaRV

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Output analog waveform from filterless NOSDAC is very bad. I think it is also timing inaccurate because signal voltage change timing is quantized

 

[ATTACH=CONFIG]33114[/ATTACH]

 

[ATTACH=CONFIG]33115[/ATTACH]

 

[ATTACH=CONFIG]33116[/ATTACH]

Let's assume ALL nos filterless DACs behave the same way shown in your example plots, as a premise.

At 10k, example 1, the first harmonic distortion will be in 20k, a frequency NO ONE beyond his eighteens or so can hear.

At 20k, example 2, the sine wave itself NO ONE beyond his eighteens or so can hear, either, no matter how accurate it is reproduced at the DAC output.

Any claims about hearing such frequencies are simply self defeating.

One point, central in filterless DACs, is the fact that they take advantage from OUR OWN hearing physiology (which acts as a natural filter, we like it or not) thus avoiding audible issues analog and even digital filters most commonly produce, like phase group rotations.

 

Enviado desde mi G620S-L01 mediante Tapatalk

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Let's assume ALL nos filterless DACs behave the same way shown in your example plots, as a premise.

At 10k, example 1, the first harmonic distortion will be in 20k, a frequency NO ONE beyond his eighteens or so can hear.

At 20k, example 2, the sine wave itself NO ONE beyond his eighteens or so can hear, either, no matter how accurate it is reproduced at the DAC output.

Any claims about hearing such frequencies are simply self defeating.

One point, central in filterless DACs, is the fact that they take advantage from OUR OWN hearing physiology (which acts as a natural filter, we like it or not) thus avoiding audible issues analog and even digital filters most commonly produce, like phase group rotations.

 

Enviado desde mi G620S-L01 mediante Tapatalk

 

The graphs just show two types of distortion - reproducing (1) the frequency response and (2) the timing of a single sine wave. With real music, one runs into other problems, such as intermodulation distortion, which produces distortions at frequencies well within audible range for adults.

 

But of course NOS does not equal filterless. NOS DACs do have higher distortion levels with 44.1KHz input than DACs that use oversampling (though they can be excellent when fed higher sample rate material), but filterless DACs are quite a bit worse.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Attached image 1 and 2 are calculation of frequency response of ideal filterless NOSDAC.

There are aliasing noise problem. Also it has high frequency roll-off problem and

I think, in 44.1kHz PCM signal, people who has golden ear can tell SQ difference by this -4dB attenuation of the edge of hearing range?

There are DSP compensation technique available for this high frequency roll-off problem (image 3).

 

I don't have golden ear :)

 

I tried to listened to the aliasing noise by normally downsample 44.1kHz signal using SoX to get 11.025kHz PCM and upsample 4x with zero-order hold using my program. Aliasing noise is non-musical, very annoying noise something like crickets are chirping. I understand why people tried to eliminate this noise. If my ear have ultrasonic hearing ability, I will hear this annoying noise from NOSDAC playing 44.1kHz PCM. from this experience, I feel uncomfortable with NOSDAC sound because of some sort of psychological effect :)

 

Calc.png image 1

 

Fr_Graphs.png image 2

 

fr_compensation.png image 3

 

N.B. DF:OFF == filterless ZOH, DF:SHARP == sharp roll-off digital filter (conventional DAC). this high frequency roll-off compensation filter is designed for ZOH and when it is applied to conventional DAC, FR will be overcompensated, this phenomenon is depicted in purple curve on the graph

Sunday programmer since 1985

Developer of PlayPcmWin

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I think actually blind tests do show people hear 1dB differences (the same music played at one volume, then another) though not necessarily awareness of some particular non-linearities in overall frequency response.

 

Edit: Also, as mentioned before, regardless of any ability to sense ultrasonics that may or may not exist, intermodulation can result in distortions easily within audible range.

 

 

Sent from my iPhone using Computer Audiophile

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Attached image 1 and 2 are calculation of frequency response of ideal filterless NOSDAC.

There are aliasing noise problem. Also it has high frequency roll-off problem and

I think, in 44.1kHz PCM signal, people who has golden ear can tell SQ difference by this -4dB attenuation of the edge of hearing range?

There are DSP compensation technique available for this high frequency roll-off problem (image 3).

 

I don't have golden ear :)

 

I tried to listened to the aliasing noise by normally downsample 44.1kHz signal using SoX to get 11.025kHz PCM and upsample 4x with zero-order hold using my program. Aliasing noise is non-musical, very annoying noise something like crickets are chirping. I understand why people tried to eliminate this noise. If my ear have ultrasonic hearing ability, I will hear this annoying noise from NOSDAC playing 44.1kHz PCM. from this experience, I feel uncomfortable with NOSDAC sound because of some sort of psychological effect :)

 

[ATTACH=CONFIG]33255[/ATTACH] image 1

 

[ATTACH=CONFIG]33256[/ATTACH] image 2

 

[ATTACH=CONFIG]33257[/ATTACH] image 3

 

N.B. DF:OFF == filterless ZOH, DF:SHARP == sharp roll-off digital filter (conventional DAC). this high frequency roll-off compensation filter is designed for ZOH and when it is applied to conventional DAC, FR will be overcompensated, this phenomenon is depicted in purple curve on the graph

If IMD produced by ultrasonic aliasing can be audible in NOS filterless DACs (like annoying "crickets"), I wonder why pre and post ringing always present in oversampling and filtered DACs (oh, Fortune!) does not intermodulate within audible range and it is imperceptible for golden and not golden ears. ;-)

N,B. : Until TIDAL streaming in 16/44,1k, "golden ears" people were condemning the very format BECAUSE they "clearly" hear the annoying effect of filtering, among other issues of the same sort. Today, the same format with the same oversampling and filtering techniques is highly praised for the same people. Another turn of the voluble godness Fortune, indeed!

 

VenturaRV

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Pre/post ringing is not always produced from sharp roll-off antialiasing lowpass filter. It is produced only where lowpass filter does its job, i.e. where original signal contains ultrasonic component (>20kHz on 44.1kHz sampling PCM). If original signal does not contain ultrasonic component, ADC/DAC chain outputs faithful copy of original input analog waveform.

Sunday programmer since 1985

Developer of PlayPcmWin

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Pre/post ringing is not always produced from sharp roll-off antialiasing lowpass filter. It is produced only where lowpass filter does its job, i.e. where original signal contains ultrasonic component (>20kHz on 44.1kHz sampling PCM). If original signal does not contain ultrasonic component, ADC/DAC chain outputs faithful copy of original input analog waveform.

Well, that's not the case with computer generated square waves and sharp transient impulses in test CDs. Oversamplig filtered DACs ALWAYS show very visible pre and post ringing in the oscylloscope. Post ringing has, in transient response, minimal or null consequence since loudspeakers will vibrate after a sharp impulse aniway (unless you have an amp with an enormous damping factor). But PRE ringing (audible or not) will move tweeters out of their repose position BEFORE the signal arrives, in some sort of inversion of causality, or mysthic premonition ( ! ). ;-)

 

VenturaRV

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Well, that's not the case with computer generated square waves and sharp transient impulses in test CDs. Oversamplig filtered DACs ALWAYS show very visible pre and post ringing in the oscylloscope.

 

True. But these synthetic test signals are not band-limited, and viewed from the point of view of the entire signal chain (ADC-DAC) they are illegal.

 

What Yamamoto rightly said is that a filter rings only when it is hit by signal at the filter's cut-off frequency. For a DAC there should be no energy at this frequency, if the recording-side anti-aliasing filter has done its job properly.

 

Proper ADC-side anti-aliasing should attain full attenuation at Fs/2, and should do so with a transition band wide enough so as not to make its own ringing audible (to newly-borns, that is). Given such a signal, the DAC filter will not ring itself, it will only pass on the ADC-side ringing. That is mathematical fact.

 

And a NOS DAC also passes on the ADC-side ringing.

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What you failed to realize is that you can't mix DSD and all sound goes through 44/48khz during mixing (96khz if you are lucky, or unlucky if they are 44->96 then back). Then they upsample it again and that is what you have. Those DSD recording sounded horrible as a result versus PCM in practice.

 

The sound out of R2R NOS DACs sounds infinitely better than DSD DACs playing PCM or DSD. Just go buy a Directstream DAC if you really want to learn how bad it sounds. The price is crashing, now below 3k as people are waking up.

 

DSD is a bad idea that needs to die already.

Go listen to a Lampizator "chipless" DSD engine playing DSD and get back to me. The PSA DirectStream kills DSD with all that processing. It actually sounds better playing PCM. They should have created a pure DSD pathway, instread of mixing everything into a "equalization blender"!

 

Phison, HQP DSD1 module, T+A Dac8DSD and a few others out there do direct bitsream playback and do it right. MSB and PBD sounds good in DSD as well.

 

If you have not heard it done right, then you dont know.

 

I own a Lampi GG and it has 3 converters in one box. R2R Ladder for PCM, a mini chipless DSD256 converter attached to the Ladder and a separate/discrete full out DSD512 chipless converter where you can select it with the push of a button. Best of all worlds.

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I heard a Playback Designs Merlot and it sounded great with PCM and DSD. It is chipless - works on an FPGA. If I understand how it works correctly, it resamples everything to DSD128 and then filters it. So if you input DSD, no conversion to PCM.

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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I heard a Playback Designs Merlot and it sounded great with PCM and DSD. It is chipless - works on an FPGA. If I understand how it works correctly, it resamples everything to DSD128 and then filters it. So if you input DSD, no conversion to PCM.

 

From the PD website it says the Merlot accepts DSD256, so I would expect that it up samples everything to that rate.

 

 

Sent from my iPad using Computer Audiophile

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From the PD website it says the Merlot accepts DSD256, so I would expect that it up samples everything to that rate.

 

 

Sent from my iPad using Computer Audiophile

 

But you would be wrong. I oversimplified what I wrote before. It upsamples everything below DSD64 rates first to DSD128 and then in a separate process to 50Mhz before sending to the filter. If you feed it DSD128 it goes directly to the filter. If you feed it DSD256 it undergoes just the conversion to 50Mhz and then to filter.

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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I heard a Playback Designs Merlot and it sounded great with PCM and DSD. It is chipless - works on an FPGA. If I understand how it works correctly, it resamples everything to DSD128 and then filters it. So if you input DSD, no conversion to PCM.

 

Not to pick on you, but the Golden Gate was mentioned previously as well. A dac with an FPGA is not NOS as I understand it. Maybe the OP should specify what they mean by NOS. Even the Phasure is not really NOS IMO, even if the OS is being done in software, there is still OS.

 

To my mind, NOS "sound" means NOS period- such as a TDA 15xx or PCM 1704 running straight redbook. Then again, I got in trouble for suggesting a single ended triode amp ought to contain a triode tube.

Forrest:

Win10 i9 9900KS/GTX1060 HQPlayer4>Win10 NAA

DSD>Pavel's DSC2.6>Bent Audio TAP>

Parasound JC1>"Naked" Quad ESL63/Tannoy PS350B subs<100Hz

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