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Is DSD HF Quantization Noise a Real Concern? A case study with modest measurements and a comparative survey.

Scytales

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A concern is often expressed about the effect of Direct Stream Digital (DSD) high frequency (HF) quantization noise on downstream electronics, especially power amplifiers, and the tweeters of loudspeakers.

For a quarter of century since the inception of SA-CD, the first carrier of DSD, I have never seen or heard about a single properly documented case of a piece of electronic or speaker unit disturbed, let alone damaged, by the HF shaped quantization noise of DSD signals. Except one possible case which was publicly acknowledged by Mr Andrew Demery, then in charge at Philips for assisting the deployment of DSD and SA-CD in the pro market.

Despite the widespread concerns expressed by some people about DSD HF quantization noise in the Hifi press and on the Web, I have never stumbled across a properly conducted experience designed to assess possible effects of said noise on the behaviour of downstream electronics or speakers, except one published in the British magazine Hi-Fi News and Record Review (more of that latter).

Therefore, I propose to tackle this issue.

I have no access to DSD test signals of any kind save the very limited Denon Audio Check SA-CD (reference COGQ-28), which is intended to help tweaking an actual stereo or multichannel loudspeaker installation, not to test electronics or loudspeakers. I have no high performance test equipment either.

Nevertheless, I propose to compare DSD HF noise level to the HF switching noise level of amplifiers using the now popular class D technique.

To do this, I will start to point out to a relevant post by our kind host Amirm, who made comments about the noise-shaped quantization noise level of a pro-audio interface, the Focusrite Scarlett 2i2 Gen3. Amirm not only provided a relatively wide-band FFT analysis of the frequency spectrum of this interface when converting PCM signal in order to visualize the shape of the high-frequency noise spectrum, but he also measured the actual RMS voltage level of this noise inside a 90 kHz bandwidth.

Having no Audio Precision, I made some measurements of the same kind as Amirm with more modest analogue test equipment. I used a laboratory-grade active high pass filter with a sufficiently high frequency bandwidth (10 MHz @ -3 dB) to remove the low frequency content at the output of SA-CD players in order to only keep the part of the spectrum above a set corner frequency. Then, I measured the RMS noise level at the output of the active filter with a psophometer (a kind of voltmeter especially designed to measure noise).

As I own a player (Marantz DV-12S2) that uses the same DAC chip or at least of the same family as those used in the above mentioned Focuriste interface, I have measured first the shaped quantization noise level at the output of this player when it plays a signal recorded in PCM on a test disc. Here are the results:

marantz-dv-12s2-as-cd-player-hf-ouput-noise-r.png


Overall, the result seems to me consistent with those published by Amirm. Obviously, most of the noise is above 50 kHz, as shown in Armim's FFT analysis of the Focusrite.

As I consider that this first results validate the methodology, I have measured the same way the HF noise level of all the operational SA-CD players I have at hand:

sacd-players-hf-ouput-noise-r.png


The slightly higher noise level in the 20 to 300 kHz bandwidth measured with some players is most probably due to leakage in the pass band of signal content in the stop band, because the active filter I used has a relatively low rate of attenuation.

I have checked the noise levels with a Leader LMV-181A AC voltmeter that has a bandwidth of 1 MHz @ -3 dB. Despite the wider bandwidth of this voltmeter, the noise level readings are actually slightly lower than with the psophometer. I think that is due to the better detector circuit of the psophometer, which is capable of converting noise-like signal having high crest-factor (peak to average level) in more accurate RMS values. The point is that we can deduce from the measurements made with the Leader voltmeter that the quantization noise levels at the output of each SA-CD player have a significant frequency content mostly above 50 kHz and well under 300 kHz or so. This is consistent with the wide-band frequency response plot of typical SA-CD players most of us must be familiar with.

The Marantz player obviously has the widest bandwidth and lets much more quantization noise leaks at its output. Consequently, I decided to pursue the experience with this particular player considering it as a worst-case.

I have measured the noise level at the output of my preamp with its volume control set at the usual position to listen to well recorded classical music discs. Here are the results, in more bandwidths than before in order to show the approximate frequency above which the player-preamp combination begin to roll off the noise:

marantz-dv-12s2+vincent-sav-c1-dsd--hf-ouput-noise-r.png


There is no surprise: the measured levels are about 24 dB below the volume control position that corresponds to unity gain.

Now, I have not yet a power load at my disposal to measure directly at the output of my power amplifier. But it is possible to calculate the worst-case noise level after amplification by multiplying the preamp output noise voltage by the voltage gain of the power amplifier.

I have set the input level of my amplifier (Cabasse Polaris AM1000) such as an input RMS voltage equivalent to 0 dBFS or 0 dB SA-CD at the output of the player (that is, with the overall gain of the preamp set at unity) does produce an output voltage of about 29 V RMS (210 W/4 ohms). That corresponds to a voltage gain of 15 with unbalanced signal. That way, the amplifiers still have about 4.65 dB of peak voltage level headroom above 0 dB SA-CD or 0 dBFS for dealing with over-modulated DSD signals or inter-sample PCM overs.

The worst case noise level at the output of the power amplifier at usual listening level should thus be 15x6.7 mV equal to 100.5 mV.

I think I can confidently speak about worst case noise level because I will make two further observations.

Firstly, it is well known that a DSD signal is a constant power signal. The more power in the signal pass band, the less power in the noise pass band. I have also measured RMS noise levels at the preamplifier output when playing the musical tracks on the Denon SA-CD test disc: the HF noise levels did varied between at least 4.8 mV and at most 6.5 mV in a 50 to 300 kHz bandwidth depending on the specific track. The pink noise track did produce an HF noise level of 5.2 mV in the same bandwidth. From this results, we can say that the HF noise level with a -16 dB SA-CD sinus signal is representative of the worst case noise level that a musical recording will produce at the output.

Secondly, I have simulated the frequency response of the input stage of the power amplifier with the Caneda software. This stage limits the frequency response of the overall power amplifier (not taking into account an input RFI filter whose action is visible from about 8 to 10 MHz). This stage brings a further attenuation of -1.6 dB@50 kHz, -2.7 dB@70 kHz, -3.9 dB@90kHz and -5.5 dB@ 100 kHz. I should point out that the rate of attenuation of the power amplifier input stage frequency response above 100 kHz is much less than the rate of change of the simulated frequency response of the Marantz DV-12S2 output analogue low pass filter. This fact alone made me confident that the power amplifier will cope with the HF noise level produced by said Marantz.

At this point of the experience, I cannot say that the measured HF noise levels have no effects whatsoever on the electronics (pre or power amp) or the loudspeakers. But I find useful to compare the computed HF noise levels at power amplifier output with those measured at the output of some class D power amplifiers. I am perfectly aware that is not comparing like with like, because the low pass filtered DSD HF quantization noise is a broadband noise signal, whereas the low pass filtered switching noise of class D amplifier resembles more a narrow band signal or even single frequency signals. Nevertheless, the comparison could be enlightening.

For comparison, I rely on the data published by Stereophile. There are all public data found on the website of the magazine, so feel free to check them. Bear in mind they are measured with no input signal. To the best of my knowledge, when actually reproducing signals, the switching noise of some amplifiers can be higher and/or occupy a larger frequency span depending on their specific topologies.

amplifiers_hf_noise_stereophile-r.png


At this point, I can only add a summary of an experiment realized by Mr Keith Howard and published in the May 2002 issue of Hi-Fi News (Reference 1).

Mr Howard has recorded the DSD quantization noise produced by a Philips SACD-1000 SA-CD player playing the 1kHz@-160 dB SA-CD signal of the Philips Super Audio CD DAC Test Disc (ref. 3122-783-00632) with its analogue output filter set to the higher cutoff frequency. This noise was digitally analyzed to filter a white noise produced by the Cool Edit Pro software to obtain a noise spectrum at 192 kHz sample rate resembling DSD HF noise. Then, the resulting noise was mixed with some 24 bits/96 kHz musical excerpts up-sampled to 192 kHz. M. Howard then used a replay software to blind test the WAV files containing the altered version of each music tracks against the original (also previously up-sampled at 192 kHz). The replay chain consisted of a LynxTWO sound card, a potentiometer-based passive preamp and a pair of Exposure XVIII mono power amplifiers driving B&W CDM1NTs supplemented by prototype ribbon super-tweeters.

The conclusion of the blind test that Mr Howard has made on himself are : "In fact I pretty quickly abandoned attempts to achieve a significant score on ABX blind testing as a hopeless cause. Whatever the differences were, they were too subtle to secure a statistically significant result in just eight trials - and how anybody could remain sane for more than eight trials I can't imagine." Nevertheless, M. Howard claims he was able to identify a subtle difference in sighted listening, but that was to be expected in an article published in a Hi-fi magazine.

That's about as much experimentation I am aware of concerning the subject of this topic.

Reference :

1. Keith Howard, Noises off, Hi-Fi News and Record Review, Vol 47 No 5, May 2002, page 78.
 
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That's a very interesting set of measurements. My instinct has always been that the SACD quantization noise should not really be a real-world issue for most amplifier/speaker combinations.

You mention Class D, and I agree with you that you can't really compare the self-clocking behaviour which (of the high-quality products) is usually closer to a high-Q resonance in the 0.5MHz space than the broad curve of noise behaviour from SACD starting just above the audio band.

However, I remembered this observation from Bruno Putzeys about the original Hypex NCORE:
Then there is the loop structure that allows better control of the closed loop frequency response. My previous amp* has approximately a 1st order roll-off. Now, since the output filter naturally has a second order roll-off it means that this amp could be overdriven with out of band noise from e.g. DSD recordings (in fairness, only when you cranked a quiet recording high).

This may be the only time I've seen an amplifier designer comment on potential issues from HF noise in a source.

His fix:
Eigentakt has a well-defined 2nd order response which stays dead flat in the audio band and rolls off in a fully controlled manner with a sensible -3dB point of 60kHz. The loop structure used to get this behaviour is subject of a patent application. A few other items are not really germane at the moment but were worthwhile enough to put in a separate patent application.
 
I don't have any SACDs or DSD files...

This isn't something I'd worry about but it seems like a "good idea" to filter/attenuate the ultrasonics. IMO any DSD to PCM conversion should include filtering.

Somebody once said, "The wider you open the window, the more dirt comes-in."

...I did fry a power amplifier once when I was building a preamp that turned-out to be an ultrasonic (maybe RF) oscillator! My tweeters survived.

There is an audio library for the Arduino (a hobbyist microcontroller without a DAC) that uses PWM (pulse width modulation). I don't know the frequency but it's near the audio range. (It's a slow processor). Unfiltered, it's putting-out 5V peak-to-peak pulses no matter the audio content (even with silence) and I wouldn't feed THAT into any amplifier that I cared about!
 
IMO any DSD to PCM conversion should include filtering.
Indeed. Gear able to play DSD do have a low pass filter. Most of the time around 50 kHz to remove all the high and very loud quantization noise.
The moment you convert to PCM, the hardware won't apply the DSD filter anymore.
As it is PCM, one must filter to avoid having frequencies in the input above half the sample rate.
 
Indeed. Gear able to play DSD do have a low pass filter. Most of the time around 50 kHz to remove all the high and very loud quantization noise.
The moment you convert to PCM, the hardware won't apply the DSD filter anymore.
As it is PCM, one must filter to avoid having frequencies in the input above half the sample rate.
+1
here is where the concern about noise is real .
I think a DSD player apply the requisite low pass filter to remove most of it ?
A proper DSD DAC should also do this ?
Proper DSD playback equipment apply the filter demanded by the SACD standard and the residue is very small as described by the OP all is well ,this is not a problem ! and never was

But the tinkering audiophile.

* experiments with DSD to PCM and PCM to DSD conversions .
* experiments with filters.
* employ DAC's without proper filters .

Consequently its quite possible to end up with shedloads of ultrasonic garbage that *should not be there * the DSD signal contains a lot of it unfiltered !!

There are reason why designers in the past happily bandwidth limited any input to pre amps and amps and had quite extensive DC protection. They did not trust the content to be proper at all times .
Since we had DC to dayligth amplifiers since the 80's for better specs .. Be a bit vigilant and all should be fine again .

Don't apply hair brained DAC and filter schemes :)
 
I don't have any SACDs or DSD files...

This isn't something I'd worry about but it seems like a "good idea" to filter/attenuate the ultrasonics. IMO any DSD to PCM conversion should include filtering.

Somebody once said, "The wider you open the window, the more dirt comes-in."

...I did fry a power amplifier once when I was building a preamp that turned-out to be an ultrasonic (maybe RF) oscillator! My tweeters survived.

There is an audio library for the Arduino (a hobbyist microcontroller without a DAC) that uses PWM (pulse width modulation). I don't know the frequency but it's near the audio range. (It's a slow processor). Unfiltered, it's putting-out 5V peak-to-peak pulses no matter the audio content (even with silence) and I wouldn't feed THAT into any amplifier that I cared about!
foobar's SACD plug-in also includes (selectable at 30,40,50,60Khz or user defined) filtering in case someone choses to to do DSD>PCM.

Edit:also,in case someone searches for perfect DSD test signals Multitone Analyzer does them and I believe is the only one as @pkane built the whole DSD thing from the ground up,there was not any around.
You can see some examples here.
 
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Checked some DSD to PCM converted tunes from Octave Records (PS Audio). Most of them were noisy at and below - 60 dB and 48 kHz. So the noise above 20 kHz is in the vicinity of 0.1 THD which is not much. Therefore I don't think that this is harmful to loudspeakers. But bad designed ampfifiers perhaps cannot handle the high frequencies.
 
Proper DSD playback equipment apply the filter demanded by the SACD standard and the residue is very small as described by the OP all is well ,this is not a problem ! and never was
As I have at hand the Super Audio CD System Description, Part 2: Audio Specification, Version 2.0 of March 2004 (what is called the "Scarlet Book"), I just want to make the precision that the characteristics of the low pass filter is described in Annex E of said specifications.

This Annex E is about Audio Signal Recommendations and is labelled Informative, contrary to Annex D Audio Signal Requirements, which is described as Normative.

Hence the specifications of the low pass filter are not mandatory, but left to the application designer.

The recommendations made in the Scarlett Book are :

"To protect analog amplifiers and loudspeakers, it is recommended that a Super Audio CD player
contain at its output an analog low pass filter with a cut-off frequency of maximum 50 kHz and a slope
of minimum 30 dB/Oct. For use with wide-band audio equipment, filters with a cut-off frequency of
over 50 kHz can be used.
" (Annex E2)
 
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Just for all of your possible reference and interest, at least in my home audio setup;
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532 ___on my project thread
 
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Thank you for your reply.

So, in a nutshell, If I understand correctly what you wrote in your messages #42, you and three relatives were unable to audibly distinguished between DSD files and 16 bits/44.1 ksps PCM known to originate from the exact same master and played back at the exact same level with your "Setup-1".

That's a clue that DSD HF quantization noise, even at 64 Fs, did no harm and/or had no effect on any of the device of your playback chain.

I for myself thought about a convenient way to subjectively or objectively test the effect or lack thereof of DSD HF quantization noise on preamp, amp or speaker and I came to this proposal:
DSD_HF_quantization_noise_test-set-up.jpg


The advantage I see in this set-up is that the full output spectrum of the DSD HF quantization noise can be mixed with the analogue output signal of the source device without the bandwidth limitation that a digitization of said HF noise in low-rate PCM will introduce.

Providing anyone have access to a sufficiently good analogue mixer, this test set-up could be use to check very rapidly if a particular preamplifier or amplifier is disturbed by DSD HF quantization noise.
 
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So, in a nutshell, If I understand correctly what you wrote in your messages #42, you and three relatives were unable to audibly distinguished between DSD files and 16 bits/44.1 ksps PCM known to originate from the exact same master and played back at the exact same level with your "Setup-1".

That's a clue that DSD HF quantization noise, even at 64 Fs, did no harm and/or had no effect on any of the device of your playback chain.

Sorry for my belated response. Yes, your understandings are correct in my (our) specific case on the specific hybrid SACD.

I should not generalize my "observation" to any of other hybrid SACDs having DSF=DSD64(1x) layer and PCM 44.1/16 layer since they may possibly have "intentional" mixing/mastering sound quality differences between the two layers for which we cannot blame/criticize.

At least in my audio setup (and for my ears and brain:D), I have various reasons and rationales for on-the-fly format conversion of all the tracks into 88.2 kHz or 96.0 kHz 24 bit PCM to be fed into system-wide DSP center "EKIO".
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

Furthermore, if you would be interested, please find here #931 on my project thread for the latest setup of my DSP-based multichannel multi-amplifier audio system.
 
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I for myself thought about a convenient way to subjectively or objectively test the effect or lack thereof of DSD HF quantization noise on preamp, amp or speaker and I came to this proposal:
DSD_HF_quantization_noise_test-set-up.jpg


The advantage I see in this set-up is that the full output spectrum of the DSD HF quantization noise can be mixed with the analogue output signal of the source device without the bandwidth limitation that a digitization of said HF noise in low-rate PCM will introduce.

Providing anyone have access to a sufficiently good analogue mixer, this test set-up could be use to check very rapidly if a particular preamplifier or amplifier is disturbed by DSD HF quantization noise.

Sorry, but I still do not fully understand the mechanism and purpose of your proposed diagram.

First, how can you fully synchronize the two digital sources?
WS00007757.JPG

(In this perspective, I assume my post here would be of your reference even though somewhat out of the scope of this thread.)


Second, the purpose of your proposed setup...
At least I myself am not so worried about "potential disturbance/mal-effect" of UHF noises on preamplifier/amplifier since almost all of modern HiFi amps are essentially "transparent" for UHF noises, say up to 40 kHz or higher, even though we humans cannot hear sound exceeding 25 kHz.

My concerns, on the other hand, would be potential damage to our treasure supertweeters as well as potential physiological damage/harm to our beloved pets (dogs, cats, birds, etc.) which can hear the UHF noises given by poorly QC-ed DSD tracks into "excellent" tweeters/supertweeters. Just for example, my treasure supertweeters, FOSTEX T925A, have almost flat response up to around 45 kHz in very high efficiency of 108 dB/1W1m (ref. here, here and here (PDF product sheet for T925A).
WS00007758.JPG

I am also a little bit concerned about possible long-time-span (long-year) mal-effect of such inaudible UHF noises to human ears and brain.;)

In any way, I decided to always have low-pass (high-cut) -48 dB filter at 25 kHz as shared in my post;
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532
 
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Sorry, but I still do not fully understand the mechanism and purpose of your proposed diagram.

First, how can you fully synchronize the two digital sources?
The purpose of the proposed set-up is to add or remove at will DSD HF quantization noise in the analogue domain to easely measure (or listen to) a DUT in both conditions, with or without the noise added. That way it seems to me that it would be possible to detect an effect, if it exists, of this quantization noise on the DUT.

To my mind, as the mixing is done in the analogue domain for the sole purpose to add a stable analogue noise source, synchronization of the two sources wouldn't be critical.
 
The purpose of the proposed set-up is to add or remove at will DSD HF quantization noise in the analogue domain to easely measure (or listen to) a DUT in both conditions, with or without the noise added. That way it seems to me that it would be possible to detect an effect, if it exists, of this quantization noise on the DUT.

To my mind, as the mixing is done in the analogue domain for the sole purpose to add a stable analogue noise source, synchronization of the two sources wouldn't be critical.

OK, now I almost understand your points.

As I wrote in my above post #13, I have little interest in the effect of the UHF noises on the DUT (Device Under Test); I myself, therefore, am reluctant to perform your proposed experiment(s), sorry about this.

If you would be seriously interested in your proposed experimental setup and measurements, you would please proceed into the actual experiment(s) using your own audio gears/system. I am much looking forward to hearing your own experimental setup and the results thereof in detail (e.g. detail enough like the post here and here) in the very near future.;)
 
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Edit:also,in case someone searches for perfect DSD test signals Multitone Analyzer does them and I believe is the only one as @pkane built the whole DSD thing from the ground up,there was not any around.
You can see some examples here.

And there is SOX! Just generate a Flac file of any PCM test signal at wish, i.e. with Goldwave equation evaluator. Daphile audio player can play and up-convert it to DSD of different flavours/orders on the fly with the C-3PO plug-in. All for free!

At the moment it becomes Audiophile fashion to up-convert everything to DSD on the fly for playback with software like HQPlayer, Roon, Daphile (SOX). So the question of TS becomes also relevant regarding this. Quite often the dac playing DSD is then hooked to a class-D amp. Here a link to a Pure DSD dac for diy: https://www.diyaudio.com/community/threads/signalyst-dsc1.254935/
 
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At the moment it becomes Audiophile fashion to up-convert everything to DSD on the fly for playback with software like HQPlayer, Roon, Daphile (SOX).
Really?? "fashion"???

Do the "Audiophiles" you referred properly understand what would be HiFi HiRes audio? Only for two-channel stereo playback??
If so, none of the DSP tuning can be applicable in digital domain, right?
Or, do they convert all the DSP-processed digital HiRes PCM signals into DSD to feed into stereo DAC capable of DSD playback?

The said "fashion" gives me an impression that they are intentionally adding UHF (ultra-high frequency, above say 25 kHz) quantization noises to the music tracks with no sound quality improvement at all in audible (15 Hz - 22 kHz) Fq zone.

One typical analogy would be...
You can of course up-sample a low-resolution jpg photo into higher-resolution jpg, but no improvement of image quality at all!
 
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