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Denon DCD-SA1 Review (CD/SACD Player)

So which type of slow filter is AL32? Minimum phase or linear phase slow roll-off?
 
One of the post that justifies the time spent here.
Thanks for this!

So,where this DSD-can't-be-edited myth comes from?
The rarity of the gear?Their price?

Perhaps I can address this question.

Jean-Luc Ohl, a well-known French pro, has pointed out on a French forum an old open access paper from James A. Moorer, Sonic Solution, whom I already mentioned in my post #105.

This paper is still online and accessible for free, contrary to its AES version that is behind paywall or subscription, but it's the same content : Breaking the Sound Barrier : Mastering at 96 kHz and beyond.

The publication date of the AES paper is 6th November 1996.

Yet, at this early stage, Mr Moorer already described ways (and challenges) to implement digital signal processing on DSD streams to adjust volume or realize equalization. But he stated that the required computational power of available DSP chips of the times (he mentioned the Motorola 56000 family) were insufficient to obtain the required precision for pro quality standard and that special hardware had to be built (hence the development of a Sony DSP processor, obviously the CXD2926). At several instances, he wrote about the impracticability of signal processing on 1 bit stream because of the insufficient available processing power at the time of writing his article and the necessity to develop special hardware tools.

I suspect that the myth about the non-feasibility of editing DSD comes from this state of affair at the time, ie that no off-the-shelf processor were able to do that and impracticability in Moorer's text has became impossibility in the mind of some. That myth has been perpetuated throughout the following years among the general public even when more and more powerful hardware became available.

That being said, putting aside most of the mathematics used by Mr Moorer in his article, which are mostly way over my head, there are also some descriptions of practical processing on one bit streams that are easy to understand. Mr Moorer explained in relatively simple terms how to change the gain on a 1 bit stream in the digital (discrete) domain in page 10 of his article. Suffice to say that it must be understood that a 1 bit stream consisting of 1s and 0s must be looked as symbols whose meaning are actually values of 1 and -1. Hence, it is easy to scale those values by an arbitrary factor, for instance 0.5 to bring the gain down by 6 dB. The very same process is described in Ayataka Nishio's presentation of his chip. It is a straightforward process. If one has to remodulated back to a 1 bit stream (to produce a release format for SA-CD distribution, for example), the difficulties bare on the shoulders of the following delta-sigma modulator for reasons explained by Mr Moorer in his article, but the overall process is really easy to understand*. It is also possible to not remodulate back to 1 bit and convert the results of the gain adjustment computation directly in analogue with a multibit converter. That's what is done in some Cirruc Logic digital to analogue audio converter chips such as the CS4365, CS4385, CS4398 ... with additional tricks, because Cirrus Logic combines the function of both gain adjustement and low-pass filtering of the DSD shaped quantization noise to comply with the recommended 50 kHz filter of the Scarlett book, but to my mind the principle of the gain adjustment function remains fundamentally the same.

In a David Walstra's article in Studio Sound (DSD: Where are We Now?, October 1998, page 106), it is written that a working DSD editing system designed jointly by Sony and Sonic Solution has been demonstrated at the 104th AES convention in Amsterdam in 1998. Obviously, DSD were already editable before SA-CD discs and players were put on the consumer market.

*Note : to my understanding, the whole process in fundamentally identical to the technique Sharp used to use to adjust the volume on DSD streams in his memorable "1 bit" amplifiers, such as the SM-SX1, SM-SX100, SM-SX200... In this integrated amplifiers, the DSD stream, once converted back from discrete values to analogue in the form of square signals is simply pass through... a potentiometer wired as an attenuator ! The square signals whose voltage amplitude has just been reduced by the potentiometer is modulated to another 1 bit stream by a delta-sigma modulator (7th order) to drive the switching transistors of the class D output stage. This process to adjust the gain of the bitstream in the analogue (continuous time) domain is really comparable to the process in the discrete (digital) domain : the potentiometer is just replaced by the computation with the chosen scaling factor applied on the discrete 1 bit data. One can follow closely the signal path from the output of the Sony SA-CD decoder in the DX-SX1 transport associated to the SM-SX1 amplifier up to the delta-sigma modulator chip in the service manual of each device to figure out the technique : Sharp DX-SX1 (from page 25 from output pins 47 and 48 of the Sony CXD2751 decoder) and Sharp SM-SX1 (from page 21, "SOC101" 1 bit terminal).
 
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Perhaps I can address this question.

Jean-Luc Ohl, a well-known French pro, has pointed out on a French forum an old open access paper from James A. Moorer, Sonic Solution, whom I already mentioned in my post #105.

This paper is still online and accessible for free, contrary to its AES version that is behind paywall or subscription, but it's the same content : Breaking the Sound Barrier : Mastering at 96 kHz and beyond.

The publication date of the AES paper is 6th November 1996.

Yet, at this early stage, Mr Moorer already described ways (and challenges) to implement digital signal processing on DSD streams to adjust volume or realize equalization. But he stated that the required computational power of available DSP chips of the times (he mentionned the Motorola 56000 family) were insufficient to obtain the required precision for pro quality standard and that special hardware had to be built (hence the development of a Sony DSP processor, obviously the CXD2926). At several instances, he wrote about the impractibility of signal processing on 1 bit stream because of the insufficient available processing power at the time of writing his article and the necessity to develop special hardware tools.

I suspect that the myth about the infeasiblity of editing DSD comes from this state of affair at the time, ie that no off-the-shelf processor were able to do that and impractibility in Moorer's text has became impossibility in the mind of some. That myth has been perpetuated throughout the following years among the general public even when more and more powerful hardware became available.

That being said, putting aside most of the mathematics used by Mr Moorer in his article, which are mostly way over my head, there are also some descriptions of practical processing on one bit streams that are easy to understand. Mr Moorer explained in relatively simple terms how to change the gain on a 1 bit stream in the digital (discrete) domain in page 10 of his article. Suffice to say that it must be understood that a 1 bit stream consisting of 1s and 0s must be looked as symbols whose meaning are actually values of 1 and -1. Hence, it is easy to scale those values by an arbitrary factor, for instance 0.5 to bring the gain down by 6 dB. The very same process is described in Ayataka Nishio's presentation of his chip. It is a straightforward process. The difficulties bare on the shoulder of the following delta-sigma modulator for reasons explained by Mr Moorer in his article if one have to remodulated back to a 1 bit stream (to produce a release format for SA-CD distribution, for example), but the overal process is really easy to understand*. It is also possible to not remodulate back to 1 bit and convert the results of the gain adustment computation directly in analogue with a multibit converter. That's what is done in some Cirruc Logic digital to analogue audio converter chips such as the CS4365, CS4385, CS4398... with additionnal tricks, because Cirrus Logic combines the function of both gain adjustement and low-pass filtering of the DSD shaped quantization noise to comply with the recommended 50 kHz filter of the Scarlett book, but to my mind the principle of the gain adjustement function remains fundamentally the same.

In a David Walstra's article in Studio Sound (DSD: Where are We Now?, October 1998, page 106), it is written that a working DSD editing system designed jointly by Sony and Sonic Solution has been demonstrated at the 104th AES convention in Amsterdam in 1998. Obviously, DSD were already editable before SA-CD discs and players were put on the consumer market.

*Note : to my understanding, the whole process in fundamentally identical to the technique Sharp used to use to adjust the volume on DSD streams in his memorable "1 bit" amplifiers, such as the SM-SX1, SM-SX100, SM-SX200... In this integrated amplifiers, the DSD stream, once converted back from discrete values to analogue in the form of square signals is simply pass through... a potentiometer wired as an attenuator ! The square signals whose voltage amplitude has just been reduced by the potentiometer is modulated to another 1 bit stream by a delta-sigma modulator (7th order) to drive the switching transistors of the class D output stage. This process to adjust the gain of the bitstream in the analogue (continuous time) domain is really comparable to the process in the discrete (digital) domain : the potentiometer is just replaced by the computation with the chosen scaling factor applied on the discrete 1 bit data. One can follow closely the signal path from the output of the Sony SA-CD decoder in the DX-SX1 transport associated to the SM-X1 amplifier up to the delta-sigma modulator chip in the service manuals of each device to figure out the technique : Sharp DX-SX1 (from page 25 from output pins 47 and 48 of the Sony CXD2751 decoder) and Sharp SM-SX1 (from page 20, "SOC101" 1 bit terminal).
Amazing post,thank you!
ASR is probably the place that myths die thanks to users like you and some others.
 
Looking at the DSD1792 datasheet it looks like the chip is converting DSD into PCM on the fly before playing it back through multi-bit delta sigma D/A converters. It doesn't have a "native" DSD DAC (which is basically an op-amp in comparator mode driven by a pulsetrain.
 
Looking at the DSD1792 datasheet it looks like the chip is converting DSD into PCM on the fly ..
To my mind, this DAC don't [01/04/2025 edition: see confirmation in post #177 below].

Would it that we would see the same kind of frequency response that we can see on this measurement of the DCD-SA1 taken by NTTY when the user defined function of conversion to PCM from DSD is selected (see first page of this thread for complete measurements) :

index.php


Please not the downward slope of the DSD shaped quantization noise above about 35 kHz. This is due to the fact that when converting DSD to low rate PCM, any frequency above the Nyquist frequency of the desired PCM rate (half the sample rate) have to be filtered out to avoid aliasing of the frequency content above the Nyquist frequency in the pass-band.

This other measurement of the DCD-SA1 courtesy of NTTY better shows the point :

index.php


Also note mentioned on the graph, the DSD pink noise track was obviously not converted to PCM by the Denon for the purpose of the measurement contrary to the music track (blue).

By the way, the datasheet of the Burr Brown PCM/DSD1792 (as well as datasheets of all the other DACs of the same family beginning with the PCM1738) says nothing of the kind of a conversion from DSD to PCM. In fact, the datasheet is rather sketchy about the exact nature of the work onboard this chip. Hence the necessity to measure with sufficient bandwidth and interpret the measurements. The datasheet states that DSD is low-pass filtered by a built-in "analogue low pass filter", which is a rather vague statement. Either this chip function the same way the very first Burr Brown DSD converter works, the DSD1700, ie with an analogue FIR filter (excerpts from the relevant datasheet) :

DSD1700_DSD_Filter.jpg


Or Burr Brown has gone full digital, as Cirrus Logic dual purpose PCM/DSD DACs do. One can argue without end that the multibit output signal of a digital comb filter, IIR filter or FIR filter is no longer a DSD signal but a PCM signal. For my part, I call that the debate about the sex of angels :). But to my mind, the point is moot. I may be wrong, but as long as there is no decimation process (throwing away computed samples to produce a signal sample rate lower than the input sample rate, for instance discarding 63 samples out of the digital filter and keep only one sample to produce a wide bit-width/44.1 kHz sample rate PCM signal), the original Nyquist frequency of the input DSD signal is left unchanged and thus no aliasing can occur.
 
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This is a review and detailed measurements of the Denon DCD-SA1.

View attachment 391583

As I already wrote in my previous reviews, I like testing CD Players, especially older ones. This is one more proof. My firs review of the Onkyo C-733 here contains information about my measurements which I try to align with the AES standard. It means that, over time, you can compare the devices I reviewed.

This is my first review of an SACD player, but really my interest is into CD Players, so that’s an exception I made as I needed to know what SACD (and its DSD format) had to offer.



Denon DCD-SA1 - Presentation

Released in 2005, nearly 20 years ago, it was top of the line CD/SACD player at a staggering price (7’000€+ in Europe). This player is one of those “statements” from the big Japanese names, as they like to produce them from time to time.

The conversion is performed by two BurrBrown DSD1792 stereo DACs which were state of the art converters at the time (PCM 24bits and DSD compatible).

Being a Denon, you already spotted the in-house digital filter in the “Advanced AL24 Processing” version here. This is the internal oversampling filter from Denon. It is to note, though, that the AL24 is deactivated when playing SACD (DSD) disks, since it works only with PCM data, of course. It is of equal importance to know that the player offers a mode where we can convert DSD to PCM and so benefit again from AL24 processing, which is interesting.

View attachment 391584

As an SACD player, we can play CDs, of course, and the Denon DCD-SA1 adds to that two digital inputs to benefit from its internal DAC. I like to see that possibility on CD players.

RCA and XLR are present, as we would expect.

Weighting an incredible 22kg (48.5lbs), this alone talks about the build quality. The top cover weights 3.5kg (7.5lbs) and is made of a top aluminum thick layer with copper bottom plates, both rubber-isolated in between, and all of it is of course decoupled from the sub-frame via several silent blocks :eek:

View attachment 391585

A special attention was given to power supply and isolation from vibrations. I let you have a glimpse at the inside and you can also look up on the web for more details about the engineering that went into this player:

View attachment 391586

The two power transformers are placed in an aluminum housing filled with resin in a two-steps process so that they are decoupled from their respective casts. Additionally, the two aluminum housings are again decoupled from the chassis using rubber material of different resonance to kill mechanical vibrations:

View attachment 391636

It is an absolute delight to use this player. Each button on the front face reminds you of the price you paid. The drive is also very fast which I like a lot.

View attachment 391587



Denon DCD-SA1 - Measurements (Analog outputs - From CD)

From now on, I will be consistent with my measurements as I described them on the Onkyo C-733 review. So over time, this will help comparing the items I reviewed.

From RCA, the Denon DCD-SA1 outputs 2.134Vrms, that is 0.6dB above the usual 2Vrms. The XLR output is very close at 2.09Vrms. There was a channel imbalance of around 0.16dB. The single-ended outputs invert absolute polarity; balanced outputs are non-inverting.

Let's start with the standard 999.91Hz sine @0dBFS (without dither) from my test CD (RCA out):

View attachment 409865

Left and right channels are shown but only one gets evaluated in that view. Both channels have the same performances, though.

THD+N is only limited (and so is the SINAD) by the CD Audio format. It's the best we can get.

And by the way, proof of that is if I measure the XLR out, and because of 1dB difference in the input gain, THD+N is now 93dB:

Basically, all measurements are nearly identical between RCA and XLR, so I won’t make a difference between them in the rest of this review.

As with my other reviews I add a measurement of 999.91Hz @-6dBFS to show if THD improves:

View attachment 409866

This is an overlay of the left and right channels and we see that THD goes down to -114dBr, which is good news and very good performances.

Other results (not shown) are:
  • Crosstalk : -133dB at 1kHz and 10kHz (thanks to conversion in mono-mode)
  • IMD AES DFD "Analog" : -92.2dB (18kHz & 20kHz 1:1 @-3.02dBFS)
  • IMD AES DFD "Digital" : -95.8dB (17'987Hz & 19'997Hz 1:1 @-3.02dBFS)
  • IMD AES MD : -100dB (41Hz & 7993Hz 4:1 @-1.68dBFS)
  • IMD CCIF : -95.8dB (19kHz & 20kHz 1:1 @-3.02dBFS)
  • MD DIN : -85.7dB (250Hz & 8kHz 4:1 @-1.68dBFS)
  • Pitch error : 19'997.15Hz (19'997Hz requested) : 0.00075%
If you compare with the Denon DCD-900NE, you see that we are very close. Basically that is nearly the best we can get from CD Audio.

I suppose you saw a very silent power supply above, so all the efforts from Denon there paid off. RCA only showed a very small power supply related spike at 50Hz (I live in Europe) and at a very low -125dBr. XLR outputs showed nothing, which means better rejection of power supply related noise (1kHz with dither):

View attachment 409867

Bandwidth measurements, now measured from white periodic noise (400+ averages and smoothed), showed a rapid roll-off at 20kHz (-2dB):

View attachment 409870

This is close to the Denon DCD-900NE, which means it's wanted and is an effect of the Denon AL24 oversampling filter. I’ll come back to that later.
I was disappointed to see a 0.16dB difference between the two channels (same with RCA and XLR).

Multitone test showed no issues:

View attachment 409874

CD Audio content is more than safe from distortion, no surprise considering the previous results.

Jitter test (16bits/44.1kHz) is nailed too, exhibiting a beautiful trace:

View attachment 397108

Red trace is what is on the test CD (digital output), it can’t be better. The Denon (blue trace) does not add any jitter.

Started with the Teac VRDS-20 review, and on your request + support to get it done (more here), I'm adding now an "intersample-overs" test which intends to identify the behavior of the digital filtering and DAC when it come to process near clipping signals. Because of the oversampling, there might be interpolated data that go above 0dBFS and would saturate (clip) the DAC and therefore the output. And this effect shows through distorsion (THD+N measurement up to 96kHz):

Intersample-overs tests
Bandwidth of the THD+N measurements is 20Hz - 96kHz
5512.5 Hz sine,
Peak = +0.69dBFS
7350 Hz sine,
Peak = +1.25dBFS
11025 Hz sine,
Peak = +3.0dBFS
Denon DCD-SA1
-33.6dB​
-27.6dB​
-18.3dB​
Yamaha CD-1 (Non-Oversampling CD Player)
-79.6dB​
-35.3dB​
-78.1dB​
Onkyo C-733
-79.8dB
-29.4dB
-21.2dB
Denon DCD-900NE
-34.2dB​
-30.4dB​
-19.1dB​

I kept some references and will keep the same for other reviews, so you can quickly compare. The results of the Denon DCD-SA1 mean the oversampling filter does not have headroom to prevent intersample-overs. The Yamaha CD-1 shines here because it's old enough not to have an oversampling filter.


And I forgot to add one of my favourite measurements, the THD (excluding noise) vs Frequency at @-12dBFS:

View attachment 391596

The Denon DCD-SA1 had no issue here, it is the best trace that I have in my collection of this specific measurement. This an easy test for advanced 1bit DACs. I like this measurement because it shows lack of linearity already at this level with older R2R architectures, and some lower resolution 1bit architectures too, that I enjoy testing.


Denon DCD-SA1 – AL24 measurements

As with the Denon DCD-900NE, I think the Denon proprietary oversampling filter deserves a specific section, as it finds its roots back 3 decades ago.

As @bolserst wrote some time ago ago about Denon filtering, the first iteration of ALPHA processing featured an automatic filter selection based on LSB toggling, and which I could replicate too. Subsequent version of ALPHA processing included further intelligence in terms of filter selection.

I'll try to keep this section as simple as I can, but it's a challenge (again). The situation is actually the same as with the Denon DCD-900NE. So, I’ll use a different standardized test, to show the same behaviour.

First, this is the filter response (from white noise) overlaid with the standard CCIF IMD test (19kHz+20kHz 1:1) which a lot of reviewers like to use:

View attachment 391598

At the moment, please forget about the filter response (in blue) between 60k and 72kHz.

Now, those of you used to perform and/or look at these tests will see an impossibility here. It is an obviously slow filter response (as we can see from the white noise), and so it's not logical to see total absence of aliases of 19kHz and 20kHz which would show up at 25.1kHz and 24.1kHz respectively.

Well, that is because the Denon (its AL24 filter) recognizes the typical test tones and switches to a sharp filter in that case, which would make people like me (theoretically) happy. Fail :)

To counter the test detection by the filter, it is enough to add a third test tone with this standard CCIF test. So adding a 9kHz test tone defeats the detection of the AL24 filter, and here below we get what we should:

View attachment 391599

Very much different, indeed! This time we see what's logical with a slow filter response: aliases of 19kHz and 20kHz replicate around 22.05kHz. So you find them at 25.1k and 24.1k respectively. The same goes with other standardized tests such as AES, DIN. When “detected”, the AL24 filter switches to a sharper mode, to exhibit what testers like to see : absence of aliases out of band.

Also, as with the first iteration of ALPHA processing, it detects square signals and switches to NOS (Non Oversampling) mode in that case. This allows Denon to show perfect square waves. When looking at the same in frequency domain, we get this:

View attachment 391600

This is beautiful and could be used by a teacher at the university to talk about D/A conversion and its effect on creating aliases, enveloped into a sinc function. This garbage is on purpose, only to show perfect square waves, when requested.

The two filter modes I showed, Sharp and NOS, are not activated during music playback. Their purpose is only to shine during very specific tests.

As I commented about the Denon DCD-900NE, this has been ongoing for decades and continues today. If you want to go deeper into this, I recommend you to read the Denon DCD-900NE review, as it is exactly the same, and I already compared this DCD-SA1 with the DCD-900NE.

About the situation between 62kHz and 72kHz, that has minimum effect when playing music. Again, please check the Denon DCD-900NE review, I’ve put measurements, they are the same with the DCD-SA1. Even if it's not clean, we shall not worry too much about that specific noise (it will remain at very low level, ie -100dBr).


Denon DCD-SA1 - Measurements (Optical Out - From CD)

Some of you like to know if the player can be used as a transport. So I measured the digital output, from my test CD.
The below view shows what's on the test CD (1kHz @0dBFS):

View attachment 391601

It can't be better than that, this is what's recorded on my test CD.

Equally, and again about the digital output, I already suggested here the use of an undithered 1kHz sine at -90.31dBFS to verify the quality of the drive, should we have doubts. With 16bits, the signal should appear (on a scope) as the 3DC levels of the smallest sign magnitude digital signal, which is what we get with the Denon:

View attachment 391603

This, and other measurements I performed on its digital outputs, made me confident that the Denon is a prefect transport for those who want to use it with an external DAC (for CD audio only, of course).


Denon DCD-SA1 - Measurements (Digital In - RCA out)

Very quick feedback about using the digital inputs of this player with higher resolution PCM input.
The THD does not change (of course) and the noise improves because of the bit depth increase. Unfortunately, with "only" 2Vrms output, my interface (Motu ultralite mk5) reaches rapidly its limits (because of no auto ranger like an AP, it would be much more at ease with 5Vrms at least) and so I essentially measure its noise floor:

View attachment 391605


The noise floor is lower, but again I can’t measure it reliably. What I see, from playing with dither at lower levels, is that the noise is below -105dB, and so that is very good for the time. Power supply spike appears from the reduced noise floor (and because of 256k FFT length + 32 averages) at below -130dBr (I say negligible :) ).

One thing I did not mention, because I’m keeping it for the SACD measurement section, is that below full scale (0dBFS), the THD performance increases. Let’s have a look.


Denon DCD-SA1 - Measurements (Analog outputs - From SACD)

For these tests, I used the Denon Audio Check SACD. There are only so few test tones on that disc, but it’s informative anyways.
The test SACD of Denon contains test tones at -16dBFS. It’s far from the 0dBFS we (and I) are used to use. That does not help for comparison, so I had to adapt to that.

First our regular 1kHz test tone at -16dBFS (XLR out):

View attachment 391607

Pay attention to the fact that the trace in dB relative (dBr). Since we are at -16dBFS, this has the effect to visually increase the noise floor by 16dB.

At this much lower level than full scale, we see that THD remains below -100dBr, which is very good.

The calculated noise floor is around -105dB which I’m sure is the noise floor of my measurement interface. Indeed, because the Denon output “only” 2Vrms, I have to increase the input gain of my interface by 15db! This is to respect my own protocol of measurements to get the ADC of the interface close to 0dBFS when capturing data. This allows me to get dashboard measurements from REW software that are correctly calculated, especially the ENOB. Since I’m almost only measuring CD Players, this limitation is not a problem, as a 16bits test tones with dither (which I use) have a noise floor of -93dB. Of course, here the SACD (DSD) lower bit depth puts my interface in an uncomfortable zone. On one side, that is good news for the Denon DCD-SA1.

If I go more into SACD testing, I will need to figure out a way to increase the voltage before feeding my interface. I suppose I could use an extremely transparent Preamplifier to do that. I’ll report here if I get there.

All that said, and since the Denon DCD-SA1 offers the possibility to convert DSD to PCM on the fly, and therefore benefit from the AL24 filtering, let’s see if that changes something;

View attachment 391608

Well, no. But that’s not all.

Let’s play a nasty game. I remember at the time of SACD release, I’ve been told that the extended bandwidth of DSD was a major improvement, along with the increased bit depth. If I can already testify on the later, not yet on the former.

Because of its nature (1bit stream) the DSD generates a lot of mathematical errors, say imprecision. Noise shaping comes into play to reduce these errors and to reject them out of audio band (beyond 20kHz).
So, let’s have a look at a measurement of the same 1kHz sine (still at -16dBFS) but using a wider bandwidth, up to 48kHz, allowing the software to calculate noise and distorsion up to 48kHz:


View attachment 391611

Hmmm, what we see here is a lot of noise created as soon as we go over the 20kHz theoretical limit of our ears. This is an effect of Noise Shaping. So what’s preventing a lot of noise in audio band is creating a lot of noise out of audio band. And that’s really a lot, because if you look at the dashboard, and I left it calculate the noise and distortion up to 48kHz, this is a huge difference. The calculated ENOB is now 10.5bits, ouch...

This typical noise accumulation, out of band, can be found on all tracks of the Denon Audio Check SACD.

Now, let’s try when converting DSD to PCM before feeding the DACs, which means the AL24 filter kicks in too. Here you go:

View attachment 391612

That view demonstrates that the noise generated by the noise shaping technique is part of the test Denon test SACD, it’s not generated by the player. When converting DSD to PCM, the AL24 improves the filtering by removing this out of band noise present on the track (not all, but some of it). ENOB improves by 2bits.

Last and not least, let’s feed the same test tone, but this time from my interface (24bits/96kHz), directly into the digital input of the Denon (and therefore in PCM, of course):

View attachment 391618

And voilà, despite noise included up to 45’600Hz, we keep very close performances to what we initially saw when limiting the calculation at 20kHz (again I’m nearly sure that Noise is limited by my interface).

Is that a fair test? Yes and I no. Again, I remember the promise to get more than the CD limitation at 20kHz from SACD. But if it is to get that amount of noise, I wonder.

Because again, when playing pink noise or music from the Denon Audio Check SACD, I see the same noise on all tracks. This is below an analysis of musical content (1700+ FFT averages over more than 5min) with Pink Noise:

View attachment 391620

The track (#27) in the Denon audio check SACD is Mahler: Symphony No. 2 in C minor "Resurrection" - 5th movement closing part (F.!) (Conducted by Wenceslav Neumann, Czech Philharmonic Orchestra, the choir, Gabriela Benachikova-Chapova (soprano), Eva Landova (alto))

I overlaid the SACD layer (green), CD Layer (red) and conversion from DSD to PCM (blue) of the same track to show the differences. This is in Linear Frequency Scale for a better view. Here we see that the only potential advantage of SACD is between 20kHz and 25kHz where noise is still below musical content, From 25kHz on, the shaped noise of the SACD track takes over.

Of course, that is from Denon test SACD. But isn't it supposed to be a disc containing “reference” material? If yes, can we get better than that?


Conclusion

That was a long review! Thanks for those of you who made it!
How to conclude?
  • Used as a CD player, in audio band, these are very good results for a 20 year old player, nearly best in class. Distortion could have been better, yes, for those who can hear -100dB distortion relative to 0dB signal.
  • Used as CD transport, it is flawless, no surprise (only CDA).
  • Best performances are obtained from digital inputs, thanks to Denon to let us benefit from the DAC’s full performances as we want. No DSD input though, but I guess the DSD to PCM convert will make you think about the real need for that feature.
  • The behaviour of the AL24 filtering is funny, designed to shine under measurements. It's been ongoing for a long time. The Denon DCD-S10 was already including some tricks and the very new Denon DCD-900NE carries that heritage.
  • This is an incredibly well-built CD/SACD player, I love it, one we see once in a decade. It’s still alive and kicking!
  • SACD measurements left me wondering what was into this move from Sony and Philips. I did not see anything tangibly good for me, especially knowing that I used a referenced SACD Test disc for the measurements, which I suppose has been created with great care(?).
  • Noise shaping techniques have improved, allowing the theoretical 16bits of CD Audio to be actually much lower. What's left to SACD when compared to CDA is that extended bandwidth which I did not find here, at least from the Denon Audio Check SACD.
  • Anyone who participated in the engineering, product design and approval of such a piece of art, shall be proud and must be thanked.
I hope you enjoyed the long review and, as usual, let me know how to improve and if you have questions. I have recorded all the 44 measurements for both XLR and RCA outputs (and much much more). And if you want me to publish others or run one of your choice, feel free to ask.

--------
Flo

This is a review and detailed measurements of the Denon DCD-SA1.

View attachment 391583

As I already wrote in my previous reviews, I like testing CD Players, especially older ones. This is one more proof. My firs review of the Onkyo C-733 here contains information about my measurements which I try to align with the AES standard. It means that, over time, you can compare the devices I reviewed.

This is my first review of an SACD player, but really my interest is into CD Players, so that’s an exception I made as I needed to know what SACD (and its DSD format) had to offer.



Denon DCD-SA1 - Presentation

Released in 2005, nearly 20 years ago, it was top of the line CD/SACD player at a staggering price (7’000€+ in Europe). This player is one of those “statements” from the big Japanese names, as they like to produce them from time to time.

The conversion is performed by two BurrBrown DSD1792 stereo DACs which were state of the art converters at the time (PCM 24bits and DSD compatible).

Being a Denon, you already spotted the in-house digital filter in the “Advanced AL24 Processing” version here. This is the internal oversampling filter from Denon. It is to note, though, that the AL24 is deactivated when playing SACD (DSD) disks, since it works only with PCM data, of course. It is of equal importance to know that the player offers a mode where we can convert DSD to PCM and so benefit again from AL24 processing, which is interesting.

View attachment 391584

As an SACD player, we can play CDs, of course, and the Denon DCD-SA1 adds to that two digital inputs to benefit from its internal DAC. I like to see that possibility on CD players.

RCA and XLR are present, as we would expect.

Weighting an incredible 22kg (48.5lbs), this alone talks about the build quality. The top cover weights 3.5kg (7.5lbs) and is made of a top aluminum thick layer with copper bottom plates, both rubber-isolated in between, and all of it is of course decoupled from the sub-frame via several silent blocks :eek:

View attachment 391585

A special attention was given to power supply and isolation from vibrations. I let you have a glimpse at the inside and you can also look up on the web for more details about the engineering that went into this player:

View attachment 391586

The two power transformers are placed in an aluminum housing filled with resin in a two-steps process so that they are decoupled from their respective casts. Additionally, the two aluminum housings are again decoupled from the chassis using rubber material of different resonance to kill mechanical vibrations:

View attachment 391636

It is an absolute delight to use this player. Each button on the front face reminds you of the price you paid. The drive is also very fast which I like a lot.

View attachment 391587



Denon DCD-SA1 - Measurements (Analog outputs - From CD)

From now on, I will be consistent with my measurements as I described them on the Onkyo C-733 review. So over time, this will help comparing the items I reviewed.

From RCA, the Denon DCD-SA1 outputs 2.134Vrms, that is 0.6dB above the usual 2Vrms. The XLR output is very close at 2.09Vrms. There was a channel imbalance of around 0.16dB. The single-ended outputs invert absolute polarity; balanced outputs are non-inverting.

Let's start with the standard 999.91Hz sine @0dBFS (without dither) from my test CD (RCA out):

View attachment 409865

Left and right channels are shown but only one gets evaluated in that view. Both channels have the same performances, though.

THD+N is only limited (and so is the SINAD) by the CD Audio format. It's the best we can get.

And by the way, proof of that is if I measure the XLR out, and because of 1dB difference in the input gain, THD+N is now 93dB:

Basically, all measurements are nearly identical between RCA and XLR, so I won’t make a difference between them in the rest of this review.

As with my other reviews I add a measurement of 999.91Hz @-6dBFS to show if THD improves:

View attachment 409866

This is an overlay of the left and right channels and we see that THD goes down to -114dBr, which is good news and very good performances.

Other results (not shown) are:
  • Crosstalk : -133dB at 1kHz and 10kHz (thanks to conversion in mono-mode)
  • IMD AES DFD "Analog" : -92.2dB (18kHz & 20kHz 1:1 @-3.02dBFS)
  • IMD AES DFD "Digital" : -95.8dB (17'987Hz & 19'997Hz 1:1 @-3.02dBFS)
  • IMD AES MD : -100dB (41Hz & 7993Hz 4:1 @-1.68dBFS)
  • IMD CCIF : -95.8dB (19kHz & 20kHz 1:1 @-3.02dBFS)
  • MD DIN : -85.7dB (250Hz & 8kHz 4:1 @-1.68dBFS)
  • Pitch error : 19'997.15Hz (19'997Hz requested) : 0.00075%
If you compare with the Denon DCD-900NE, you see that we are very close. Basically that is nearly the best we can get from CD Audio.

I suppose you saw a very silent power supply above, so all the efforts from Denon there paid off. RCA only showed a very small power supply related spike at 50Hz (I live in Europe) and at a very low -125dBr. XLR outputs showed nothing, which means better rejection of power supply related noise (1kHz with dither):

View attachment 409867

Bandwidth measurements, now measured from white periodic noise (400+ averages and smoothed), showed a rapid roll-off at 20kHz (-2dB):

View attachment 409870

This is close to the Denon DCD-900NE, which means it's wanted and is an effect of the Denon AL24 oversampling filter. I’ll come back to that later.
I was disappointed to see a 0.16dB difference between the two channels (same with RCA and XLR).

Multitone test showed no issues:

View attachment 409874

CD Audio content is more than safe from distortion, no surprise considering the previous results.

Jitter test (16bits/44.1kHz) is nailed too, exhibiting a beautiful trace:

View attachment 397108

Red trace is what is on the test CD (digital output), it can’t be better. The Denon (blue trace) does not add any jitter.

Started with the Teac VRDS-20 review, and on your request + support to get it done (more here), I'm adding now an "intersample-overs" test which intends to identify the behavior of the digital filtering and DAC when it come to process near clipping signals. Because of the oversampling, there might be interpolated data that go above 0dBFS and would saturate (clip) the DAC and therefore the output. And this effect shows through distorsion (THD+N measurement up to 96kHz):

Intersample-overs tests
Bandwidth of the THD+N measurements is 20Hz - 96kHz
5512.5 Hz sine,
Peak = +0.69dBFS
7350 Hz sine,
Peak = +1.25dBFS
11025 Hz sine,
Peak = +3.0dBFS
Denon DCD-SA1
-33.6dB​
-27.6dB​
-18.3dB​
Yamaha CD-1 (Non-Oversampling CD Player)
-79.6dB​
-35.3dB​
-78.1dB​
Onkyo C-733
-79.8dB
-29.4dB
-21.2dB
Denon DCD-900NE
-34.2dB​
-30.4dB​
-19.1dB​

I kept some references and will keep the same for other reviews, so you can quickly compare. The results of the Denon DCD-SA1 mean the oversampling filter does not have headroom to prevent intersample-overs. The Yamaha CD-1 shines here because it's old enough not to have an oversampling filter.


And I forgot to add one of my favourite measurements, the THD (excluding noise) vs Frequency at @-12dBFS:

View attachment 391596

The Denon DCD-SA1 had no issue here, it is the best trace that I have in my collection of this specific measurement. This an easy test for advanced 1bit DACs. I like this measurement because it shows lack of linearity already at this level with older R2R architectures, and some lower resolution 1bit architectures too, that I enjoy testing.


Denon DCD-SA1 – AL24 measurements

As with the Denon DCD-900NE, I think the Denon proprietary oversampling filter deserves a specific section, as it finds its roots back 3 decades ago.

As @bolserst wrote some time ago ago about Denon filtering, the first iteration of ALPHA processing featured an automatic filter selection based on LSB toggling, and which I could replicate too. Subsequent version of ALPHA processing included further intelligence in terms of filter selection.

I'll try to keep this section as simple as I can, but it's a challenge (again). The situation is actually the same as with the Denon DCD-900NE. So, I’ll use a different standardized test, to show the same behaviour.

First, this is the filter response (from white noise) overlaid with the standard CCIF IMD test (19kHz+20kHz 1:1) which a lot of reviewers like to use:

View attachment 391598

At the moment, please forget about the filter response (in blue) between 60k and 72kHz.

Now, those of you used to perform and/or look at these tests will see an impossibility here. It is an obviously slow filter response (as we can see from the white noise), and so it's not logical to see total absence of aliases of 19kHz and 20kHz which would show up at 25.1kHz and 24.1kHz respectively.

Well, that is because the Denon (its AL24 filter) recognizes the typical test tones and switches to a sharp filter in that case, which would make people like me (theoretically) happy. Fail :)

To counter the test detection by the filter, it is enough to add a third test tone with this standard CCIF test. So adding a 9kHz test tone defeats the detection of the AL24 filter, and here below we get what we should:

View attachment 391599

Very much different, indeed! This time we see what's logical with a slow filter response: aliases of 19kHz and 20kHz replicate around 22.05kHz. So you find them at 25.1k and 24.1k respectively. The same goes with other standardized tests such as AES, DIN. When “detected”, the AL24 filter switches to a sharper mode, to exhibit what testers like to see : absence of aliases out of band.

Also, as with the first iteration of ALPHA processing, it detects square signals and switches to NOS (Non Oversampling) mode in that case. This allows Denon to show perfect square waves. When looking at the same in frequency domain, we get this:

View attachment 391600

This is beautiful and could be used by a teacher at the university to talk about D/A conversion and its effect on creating aliases, enveloped into a sinc function. This garbage is on purpose, only to show perfect square waves, when requested.

The two filter modes I showed, Sharp and NOS, are not activated during music playback. Their purpose is only to shine during very specific tests.

As I commented about the Denon DCD-900NE, this has been ongoing for decades and continues today. If you want to go deeper into this, I recommend you to read the Denon DCD-900NE review, as it is exactly the same, and I already compared this DCD-SA1 with the DCD-900NE.

About the situation between 62kHz and 72kHz, that has minimum effect when playing music. Again, please check the Denon DCD-900NE review, I’ve put measurements, they are the same with the DCD-SA1. Even if it's not clean, we shall not worry too much about that specific noise (it will remain at very low level, ie -100dBr).


Denon DCD-SA1 - Measurements (Optical Out - From CD)

Some of you like to know if the player can be used as a transport. So I measured the digital output, from my test CD.
The below view shows what's on the test CD (1kHz @0dBFS):

View attachment 391601

It can't be better than that, this is what's recorded on my test CD.

Equally, and again about the digital output, I already suggested here the use of an undithered 1kHz sine at -90.31dBFS to verify the quality of the drive, should we have doubts. With 16bits, the signal should appear (on a scope) as the 3DC levels of the smallest sign magnitude digital signal, which is what we get with the Denon:

View attachment 391603

This, and other measurements I performed on its digital outputs, made me confident that the Denon is a prefect transport for those who want to use it with an external DAC (for CD audio only, of course).


Denon DCD-SA1 - Measurements (Digital In - RCA out)

Very quick feedback about using the digital inputs of this player with higher resolution PCM input.
The THD does not change (of course) and the noise improves because of the bit depth increase. Unfortunately, with "only" 2Vrms output, my interface (Motu ultralite mk5) reaches rapidly its limits (because of no auto ranger like an AP, it would be much more at ease with 5Vrms at least) and so I essentially measure its noise floor:

View attachment 391605


The noise floor is lower, but again I can’t measure it reliably. What I see, from playing with dither at lower levels, is that the noise is below -105dB, and so that is very good for the time. Power supply spike appears from the reduced noise floor (and because of 256k FFT length + 32 averages) at below -130dBr (I say negligible :) ).

One thing I did not mention, because I’m keeping it for the SACD measurement section, is that below full scale (0dBFS), the THD performance increases. Let’s have a look.


Denon DCD-SA1 - Measurements (Analog outputs - From SACD)

For these tests, I used the Denon Audio Check SACD. There are only so few test tones on that disc, but it’s informative anyways.
The test SACD of Denon contains test tones at -16dBFS. It’s far from the 0dBFS we (and I) are used to use. That does not help for comparison, so I had to adapt to that.

First our regular 1kHz test tone at -16dBFS (XLR out):

View attachment 391607

Pay attention to the fact that the trace in dB relative (dBr). Since we are at -16dBFS, this has the effect to visually increase the noise floor by 16dB.

At this much lower level than full scale, we see that THD remains below -100dBr, which is very good.

The calculated noise floor is around -105dB which I’m sure is the noise floor of my measurement interface. Indeed, because the Denon output “only” 2Vrms, I have to increase the input gain of my interface by 15db! This is to respect my own protocol of measurements to get the ADC of the interface close to 0dBFS when capturing data. This allows me to get dashboard measurements from REW software that are correctly calculated, especially the ENOB. Since I’m almost only measuring CD Players, this limitation is not a problem, as a 16bits test tones with dither (which I use) have a noise floor of -93dB. Of course, here the SACD (DSD) lower bit depth puts my interface in an uncomfortable zone. On one side, that is good news for the Denon DCD-SA1.

If I go more into SACD testing, I will need to figure out a way to increase the voltage before feeding my interface. I suppose I could use an extremely transparent Preamplifier to do that. I’ll report here if I get there.

All that said, and since the Denon DCD-SA1 offers the possibility to convert DSD to PCM on the fly, and therefore benefit from the AL24 filtering, let’s see if that changes something;

View attachment 391608

Well, no. But that’s not all.

Let’s play a nasty game. I remember at the time of SACD release, I’ve been told that the extended bandwidth of DSD was a major improvement, along with the increased bit depth. If I can already testify on the later, not yet on the former.

Because of its nature (1bit stream) the DSD generates a lot of mathematical errors, say imprecision. Noise shaping comes into play to reduce these errors and to reject them out of audio band (beyond 20kHz).
So, let’s have a look at a measurement of the same 1kHz sine (still at -16dBFS) but using a wider bandwidth, up to 48kHz, allowing the software to calculate noise and distorsion up to 48kHz:


View attachment 391611

Hmmm, what we see here is a lot of noise created as soon as we go over the 20kHz theoretical limit of our ears. This is an effect of Noise Shaping. So what’s preventing a lot of noise in audio band is creating a lot of noise out of audio band. And that’s really a lot, because if you look at the dashboard, and I left it calculate the noise and distortion up to 48kHz, this is a huge difference. The calculated ENOB is now 10.5bits, ouch...

This typical noise accumulation, out of band, can be found on all tracks of the Denon Audio Check SACD.

Now, let’s try when converting DSD to PCM before feeding the DACs, which means the AL24 filter kicks in too. Here you go:

View attachment 391612

That view demonstrates that the noise generated by the noise shaping technique is part of the test Denon test SACD, it’s not generated by the player. When converting DSD to PCM, the AL24 improves the filtering by removing this out of band noise present on the track (not all, but some of it). ENOB improves by 2bits.

Last and not least, let’s feed the same test tone, but this time from my interface (24bits/96kHz), directly into the digital input of the Denon (and therefore in PCM, of course):

View attachment 391618

And voilà, despite noise included up to 45’600Hz, we keep very close performances to what we initially saw when limiting the calculation at 20kHz (again I’m nearly sure that Noise is limited by my interface).

Is that a fair test? Yes and I no. Again, I remember the promise to get more than the CD limitation at 20kHz from SACD. But if it is to get that amount of noise, I wonder.

Because again, when playing pink noise or music from the Denon Audio Check SACD, I see the same noise on all tracks. This is below an analysis of musical content (1700+ FFT averages over more than 5min) with Pink Noise:

View attachment 391620

The track (#27) in the Denon audio check SACD is Mahler: Symphony No. 2 in C minor "Resurrection" - 5th movement closing part (F.!) (Conducted by Wenceslav Neumann, Czech Philharmonic Orchestra, the choir, Gabriela Benachikova-Chapova (soprano), Eva Landova (alto))

I overlaid the SACD layer (green), CD Layer (red) and conversion from DSD to PCM (blue) of the same track to show the differences. This is in Linear Frequency Scale for a better view. Here we see that the only potential advantage of SACD is between 20kHz and 25kHz where noise is still below musical content, From 25kHz on, the shaped noise of the SACD track takes over.

Of course, that is from Denon test SACD. But isn't it supposed to be a disc containing “reference” material? If yes, can we get better than that?


Conclusion

That was a long review! Thanks for those of you who made it!
How to conclude?
  • Used as a CD player, in audio band, these are very good results for a 20 year old player, nearly best in class. Distortion could have been better, yes, for those who can hear -100dB distortion relative to 0dB signal.
  • Used as CD transport, it is flawless, no surprise (only CDA).
  • Best performances are obtained from digital inputs, thanks to Denon to let us benefit from the DAC’s full performances as we want. No DSD input though, but I guess the DSD to PCM convert will make you think about the real need for that feature.
  • The behaviour of the AL24 filtering is funny, designed to shine under measurements. It's been ongoing for a long time. The Denon DCD-S10 was already including some tricks and the very new Denon DCD-900NE carries that heritage.
  • This is an incredibly well-built CD/SACD player, I love it, one we see once in a decade. It’s still alive and kicking!
  • SACD measurements left me wondering what was into this move from Sony and Philips. I did not see anything tangibly good for me, especially knowing that I used a referenced SACD Test disc for the measurements, which I suppose has been created with great care(?).
  • Noise shaping techniques have improved, allowing the theoretical 16bits of CD Audio to be actually much lower. What's left to SACD when compared to CDA is that extended bandwidth which I did not find here, at least from the Denon Audio Check SACD.
  • Anyone who participated in the engineering, product design and approval of such a piece of art, shall be proud and must be thanked.
I hope you enjoyed the long review and, as usual, let me know how to improve and if you have questions. I have recorded all the 44 measurements for both XLR and RCA outputs (and much much more). And if you want me to publish others or run one of your choice, feel free to ask.

--------
Flo


Thanks a lot TTY for this mindfulness explanation.
I'm new here and give a reply on your test.
I understand now that the al24 (or al32) filtering is only actived on PCM and not with SACD, i didn't know about that.
The noise floor above 20khz when using SACD is also striking.

I own a Denon DCD-1600NE and only playing CD's on it and the unit is analog attached to a Audiolab 6000A.
Also i have attached the coax output of the 1600NE on the coax input (DAC) on the 6000A (so i can switching when i play a CD).

Now there is a huge difference in hearing when using the analog output and the coax output.
The analog output sounds much better than the coaxial output.
The difference is so big that I don't think a better DAC can beat it.

So despite the high noise level of the SACD mode, it sounds much better aurally.
Measuring is knowing, but the human ear is the ultimate goal for experiencing the music.

My question to you is:
Do you have the same hearing experience with this? so switching between the analog and digital output of the SACD player?

Anyway, what you do is very, very interesting, so keep on going this good work, it's all an eye-opener and keeps us sharp! :)
 
Thanks a lot TTY for this mindfulness explanation.
I'm new here and give a reply on your test.
I understand now that the al24 (or al32) filtering is only actived on PCM and not with SACD, i didn't know about that.
The noise floor above 20khz when using SACD is also striking.

I own a Denon DCD-1600NE and only playing CD's on it and the unit is analog attached to a Audiolab 6000A.
Also i have attached the coax output of the 1600NE on the coax input (DAC) on the 6000A (so i can switching when i play a CD).

Now there is a huge difference in hearing when using the analog output and the coax output.
The analog output sounds much better than the coaxial output.
The difference is so big that I don't think a better DAC can beat it.

So despite the high noise level of the SACD mode, it sounds much better aurally.
Measuring is knowing, but the human ear is the ultimate goal for experiencing the music.

My question to you is:
Do you have the same hearing experience with this? so switching between the analog and digital output of the SACD player?I think
It seems you are a little confused.

If you are only playing CD on your Denon, the performance in SA-CD mode is irrelevant for you because you have not any SA-CD to play. SA-CD is a different type of disc than CD, it is not a different operating mode of the player. If you only play CD on your Denon, you only get the kind of analogue output measured with CD source, ie without the shaped quantization noise typical of SA-CD replay above somewhat 25 kHz that bothers you.
 
Thanks a lot TTY for this mindfulness explanation.
Thanks!
I'm new here and give a reply on your test.
I understand now that the al24 (or al32) filtering is only actived on PCM and not with SACD, i didn't know about that.
Correct.
The noise floor above 20khz when using SACD is also striking.
That is part of the DSD signal and its specific conversion to analog. It’s the case when playing an SACD only.
I own a Denon DCD-1600NE and only playing CD's on it and the unit is analog attached to a Audiolab 6000A.
Also i have attached the coax output of the 1600NE on the coax input (DAC) on the 6000A (so i can switching when i play a CD).
I’d like to test a 1600NE, it’s certainly an excellent player.
Now there is a huge difference in hearing when using the analog output and the coax output.

The analog output sounds much better than the coaxial output.
The difference is so big that I don't think a better DAC can beat it.
Probably because the analog output voltage of the Denon is higher than the one of the internal converter of your amplifier. Even if it is 1dB more, that’s sufficient to make you perceive a lot more (because it plays slightly louder) and get that impression it’s better.
So despite the high noise level of the SACD mode, it sounds much better aurally.
Measuring is knowing, but the human ear is the ultimate goal for experiencing the music.

My question to you is:
Do you have the same hearing experience with this? so switching between the analog and digital output of the SACD player?
Yes if I don’t adjust the gain between the Denon analog output and the external converter, else no.
Anyway, what you do is very, very interesting, so keep on going this good work, it's all an eye-opener and keeps us sharp! :)
Thanks!
 
It seems you are a little confused.

If you are only playing CD on your Denon, the performance in SA-CD mode is irrelevant for you because you have not any SA-CD to play. SA-CD is a different type of disc than CD, it is not a different operating mode of the player. If you only play CD on your Denon, you only get the kind of analogue output measured with CD source, ie without the shaped quantization noise typical of SA-CD replay above somewhat 25 kHz that bothers you.
Thanks Scytales for your response.
I understand what you are saying, that is clear to me.
But listening a CD over a SACD player through the analog output is 10x better then listening over PCM (the digital output) or another normal CD-player, despite the SA-CD mode is irrelevant in this case.
Is there a technical explanation for it ?
If you or someone else have a SACD player you could do this test too, i'm curious about the results.
 
NTTY has provided you with a possible explanation to your subjective assessment : "Probably because the analog output voltage of the Denon is higher than the one of the internal converter of your amplifier. Even if it is 1dB more, that’s sufficient to make you perceive a lot more (because it plays slightly louder) and get that impression it’s better."*

To check if this probability is the correct explanation, I invite your to procure yourself a test CD with fixed tone frequencies (for instance : ask NTTY for the files of his own TEST CD).

Disconnect your speaker from you amplifier to avoid any accident due to the high level of some test tones.

Select whatever track with a fixed sinusoidal test not to high in frequency, ie 1 kHz or lower, and sufficient level (at least -20 dBFS or higher).

Measure the output of your preamplifier at an RCA jack or the output of your amplifier at the speaker binding post with a suitable voltmeter. Absolute accuracy of the voltmeter is not critical, because the purpose of the experiment is just to check that output levels are the same between two modes of operation: what interested us is to know if the voltmeter measure the same level or not from one mode of operation to the other.

Play alternatively the chosen track through the analogue output of the CD player to the analogue input of the preamp ad through the digital output of the player to the digital input of the preamp. Note the measured voltages in each case.

If there is a significant difference between the two voltages, something like 0,2 dB or more, it is likely that your preference is caused by a simple difference of level.

If not... well, we'll see.
 
NTTY has provided you with a possible explanation to your subjective assessment : "Probably because the analog output voltage of the Denon is higher than the one of the internal converter of your amplifier. Even if it is 1dB more, that’s sufficient to make you perceive a lot more (because it plays slightly louder) and get that impression it’s better."*

To check if this probability is the correct explanation, I invite your to procure yourself a test CD with fixed tone frequencies (for instance : ask NTTY for the files of his own TEST CD).

Disconnect your speaker from you amplifier to avoid any accident due to the high level of some test tones.

Select whatever track with a fixed sinusoidal test not to high in frequency, ie 1 kHz or lower, and sufficient level (at least -20 dBFS or higher).

Measure the output of your preamplifier at an RCA jack or the output of your amplifier at the speaker binding post with a suitable voltmeter. Absolute accuracy of the voltmeter is not critical, because the purpose of the experiment is just to check that output levels are the same between two modes of operation: what interested us is to know if the voltmeter measure the same level or not from one mode of operation to the other.

Play alternatively the chosen track through the analogue output of the CD player to the analogue input of the preamp ad through the digital output of the player to the digital input of the preamp. Note the measured voltages in each case.

If there is a significant difference between the two voltages, something like 0,2 dB or more, it is likely that your preference is caused by a simple difference of level.

If not... well, we'll see.
Thanks Scytales.

I know what you mean with differences in the output volume and i am familiair with that phenomenon, but never the less to test both outputs is a good idea. I have the Denon test-cd with 1 kHz and a suitable voltmeter. I will try this out and will give feedback about this.

Good to mention is that i calculate, design and build my own speakers for many years now and ofcourse heared a lot of speakers, my hearing is a sort of "absolute", like something is out of phase or distortion of sounds my ears are straight irritated :)
 
Besides functionalities, I don’t see how it could best the 900NE, and what for.
No USB nor digital inputs, no balanced out, same AL32, for $3k?
I expect more from Denon at that price point.
 
Sometimes I am wondering... if it is better to just buy some CD player used as CD transport and to take some Denon AVR with AKM DAC - would it be better?
Even like that the price saved will be 2-3 times.. Just thinking
 
To my mind, this DAC don't.

Would it that we would see the same kind of frequency response that we can see on this measurement of the DCD-SA1 taken by NTTY when the user defined function of conversion to PCM from DSD is selected (see first page of this thread for complete measurements) :

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Please not the downward slope of the DSD shaped quantization noise above about 35 kHz. This is due to the fact that when converting DSD to low rate PCM, any frequency above the Nyquist frequency of the desired PCM rate (half the sample rate) have to be filtered out to avoid aliasing of the frequency content above the Nyquist frequency in the pass-band.

This other measurement of the DCD-SA1 courtesy of NTTY better shows the point :

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Also note mentioned on the graph, the DSD pink noise track was obviously not converted to PCM by the Denon for the purpose of the measurement contrary to the music track (blue).

By the way, the datasheet of the Burr Brown PCM/DSD1792 (as well as datasheets of all the other DACs of the same family beginning with the PCM1738) says nothing of the kind of a conversion from DSD to PCM. In fact, the datasheet is rather sketchy about the exact nature of the work onboard this chip. Hence the necessity to measure with sufficient bandwidth and interpret the measurements. The datasheet states that DSD is low-pass filtered by a built-in "analogue low pass filter", which is a rather vague statement. Either this chip function the same way the very first Burr Brown DSD converter works, the DSD1700, ie with an analogue FIR filter (excerpts from the relevant datasheet) :

View attachment 413440

Or Burr Brown has gone full digital, as Cirrus Logic dual purpose PCM/DSD DACs do. One can argue without end that the multibit output signal of a digital comb filter, IIR filter or FIR filter is no longer a DSD signal but a PCM signal. For my part, I call that the debate about the sex of angels :). But to my mind, the point is moot. I may be wrong, but as long as there is no decimation process (throwing away computed samples to produce a signal sample rate lower than the input sample rate, for instance discarding 63 samples out of the digital filter and keep only one sample to produce a wide bit-width/44.1 kHz sample rate PCM signal), the original Nyquist frequency of the input DSD signal is left unchanged and thus no aliasing can occur.
For the sake of completeness, I wish to share some information I was able to found again that I think definitively dispel the various claims that the Burr Brown DACs of the PCM1738 and PCM/DSD179x family internally convert the DSD modulation in a PCM modulation.

The aforementioned claims is mostly, if not solely, based upon the following block diagram out of this Burr Brown family of DAC chips datasheets (the DSD1792A datasheet in this case):

DSD1792_Block_Diagram.png


This block diagram shows a common digital PCM/DSD interface followed by a an 8x oversampling filter. That can lead one to believe that DSD is decimated into a low sample rate PCM modulation prior to being over-sampled again.

But here is the actual block diagram of the first DAC of this family (the PCM1738, AKA the Sony CXD9657), taken out of the AES presentation of this chip by the Japanese designers, that clearly shows the two different signal processing paths between DSD and PCM signals [1]:

Actual_PCM1738_block_diagram.png

It is clearly shown that DSD is processed by a separate digital filter and that no decimation of the DSD stream takes place (as demonstrated by NTTY measurements among others).

The DSD filter is unquestionably digital: it produces a digital code that drives the current DAC stage the same way the entire digital signal processing applied on any PCM modulation produces a digital code that drive the same DAC stage (consisting of equally weighted current sources capable of producing up to 67 different levels of current when combined together).

The base structure of the DSD filter is most probably one of those illustrated in an invention patent application by two of the DAC chip designers [2]. It is an FIR filter:

DSD_FIR_Filter.png


In the case of the PCM1738 (or PCM/DSD179x), k=h with DSD input, since the actual multi-bit current DAC stage can operate at least at 64 FS, which is the default rate of the third order delta-sigma modulator in PCM mode. The fact that kFS is equal to hFS is actually emphasized in the patent application as a novelty at the time.

This other diagram out of the same patent application illustrates the difference between the new chip and the former DSD1700 analog FIR filter structure, with the clear separation between digital (a simple delays line) and analogue functions as defined by the designers:

DSD1700_FIR_Filter.png


[1] Shige Nakao, Hitoshi Terasawa, Fumitaka Aoyagi, Norio Terada, and Toshi Hamasaki, "A 117-dB D-Range Current-Mode Multi-Bit Audio DAC for PCM and DSD Audio Playback", 109th AES Convention, Los Angeles, California.
[2] Shige Nakao and Toshi Hamasaki, "Method and apparatus for digital-to-analog converting a signal modulated in the frequency domain", US20020018012A1.
 
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