DIY OPEN SOURCE BUILD WIPES FLOOR WITH MORE EXPENSIVE COMMERCIAL DSD DACS
I received a coveted not easy to find product recently. It is a Signalyst (HQPlayer) true DSD DAC. Click here for description. Or, perhaps Converter is simply the best thing to call it, since, it really has no DAC in it, which is one of the points in DSD's favor; simplicity of conversion. IF one choses to make it simple. That tends NOT to be the case with many DSD DAC's, with their digital volume controls, so called 'DSD-Domain' processing, re-modulation, etc.
I recently made a blog post on how pure DSD works. (Click here to read the post). The purest DSD conversion is still something of a holy grail, involving only an analog filter, such as a RC filter, and never having more than the 1 bit bitstream. The more realistic, and better performing pure DSD conversion still uses just an analog filter, but it's a FIR filter, which is more complex than the RC and other analog alternatives. But a FIR filter, which is usually implemented via digital means, can be done 100 percent in the analog realm, either on silicon or via discrete analog components. The beauty of the Signalyst DSC DAC is it is made of all discrete analog components once you are past the USB input stage.
The DSC DAC is the hardware version of what we call 'open source' software, so its design is open to some tinkering and modifications. My version is known as the DSC2, and is a fully differential balanced output that uses output transformers for final analog stage. Output transformers are a bit controversial in modern solid state equipment, because they have higher harmonic distortion than many find unacceptable, nor are they perfectly linear. The worst distortion is found at the low frequencies, and it lessens as frequency increases. The output transformer will impart something of a tube like sound. However, this does not have to be the analog output stage for this DAC. Going back to the 'open source' nature, it could be anything of the designer's choosing.
Anyone read the current controversy regarding the uber expensive PSAudio Directstream DSD DAC MK2? And its 'high' noise levels and 'high' distortion levels? Well, most of its measurement issues center around the use of output transformers.
I am making NO statement whatsoever on the efficacy of the PSAudio implementation or its actual sound. I have never heard it. What I DO know, is my DSC2 Converter, which costs a tenth as much (that is, if you can find one or the parts to build it. It is unfortunately NOT a commercial product.) significantly outperforms the Directstream in every measurement of the DSC2 I have made, while only needing a simple software such as Roon to listen to PCM files, and needing NO conversion to listen to native DSD files. (HQPlayer, though, is recommended as it and the DSC2 converter are designed to work in tandem.) It is bitperfect with DSD, something the Directstream cannot claim.
The 'worst' measurements will be the THD+N, which inverted is called SINAD. Anything over 90db SINAD is certainly decent enough performance, and 100db SINAD is probably more than good enough for most ears. All of this, however, is the consequence of the use of output transformers at the output stage in the DSC2. I can gladly report, though, that the transformers used are quite linear, for transformers. It stays above 90db SINAD from approx. 50hz to over 15khz. Not only that, the NOISE floor is well below -120db at virtually all frequencies. The low level amplitude linearity is ASTONISHINGLY good for a 1 bit DAC. And jitter? Not even a factor. Less than 20 picoseconds. The worst thing the the lower SINAD will do is impart a sweet, tube like sound, which is exactly what it does, but NEVER at the expense of treble extension, detail or transient response. It shifts a little to the warm side of neutral, and as is the case with many a piece of kit, will need careful equipment matching. Yet overall its sound is both state of the art digital and beguilingly 'analog like' at the same time. Its a beautiful sound that you will just want to listen to all night long, and not feel guilty about the internet mob mocking the noise floor, low linearity, or jitter while you fall asleep, because there is nothing there for the absolutist zealots to mock.
Yet, I am still hesitant to post too many measurements, by experience I have become acutely aware of how important clean power is. I am currently running the DSC2 on a variable output Switch Mode Power Supply that is of unknown noise quality. I will be more comfortable when the XPower 9 volt arrives. Apparently it was out of stock on Amazon and is delayed to a Saturday delivery. More refined measurements and graphs to come.
UPDATE 7/2/2023- I now have a 9v iFi Xpower SMPS powering the DSC2 DAC. It did exactly as expected, lowered the 'hash' and anomalous 'stuff' in the noise floor. See the new graphs below.
The resolution of the DSC2 is exceptional. Both the noise floor, and the linearity. (Both of these are required for true high resolution. Often we focus on the excellent in-band noise floor of DSD, but ignore the low level resolution that is exposed in a linearity test)
No worries here. As I said, the linearity of the DSC2 is truly first class.
All these measurements are at DSD256.
At -100db, linearity is virtually perfect. deviations are LESS than 0.1db. Much less.
At -110db, we are STILL at <0.1db deviation. This beats out some DACs and come close to others made by that other company with the letters DSC in it, lol.
At -115db, we finally hit some loss of resolution, with a deviation of an 'entire' lol -0.5db. Wow.
-119db we finally have a deviation of -2.5db.
So with my measurements we have practically perfect 18 bits of resolution, still very, very linear to 19 bits of resolution. Linearity doesn't quite make it to a 'perfect' 20 bits, but for a 1 bit converter this is exceptional. Heck, for any converter it is impressive.
Couple that with the actual noise floor that is lower than -120db, I cannot help but be impressed by the measured performance here.
SINAD (THD+N) in my latest, better calibrated measurements stays in the mid 90db range, approaching 100db at times.
Even deep into the low frequencies, it only drops once you get under 50hz, and more like 30hz. Our ears are not as sensitive to the distortion that low anyway. Once again, I am thinking of a $8000 DSD transformer DAC that does not come close to this level of performance.
And on jitter... my final measurements at DSD256, with a 48khz base rate, (48k x 256 or 12,288,000hz), I got a low result of a mere 18 picosecond jitter.
I will remeasure everything once I get the XPower 9V power supply. I am not sure the numbers will change much; I do expect the FFT noise floor to be cleaner however, in my graphs.
As with the Topping E70 ESS chipset vs E70V AKM chipset comparo, the iFi comparo will have to come out to you in multiple parts. There is just too much info for 1 review post. Be on lookout for the official iFi NEO DAC/Headphone Amplifier this week. I have gathered lots of data and am in the process of sorting what data makes it and what gets left behind... At this point all I will say is, the iFi series of kit is truly amazing stuff. Lots of overlap in their different categories as well, meaning that deciding exactly what to buy isn't just a cut and dry affair. Hopefully the data Euphonic Review has gathered can help anyone still on the fence sort it out.
DSD can be like TNT in the audiophile world. Even as accepted as it is these days, it still evokes controversy in a lot of circles. Less so now than in the 'Great DSD Wars' of the previous decade. But there indeed was a time when having a strong opinion one way or the other about DSD was a path to anathema from someone or some group. No one was pleased. In those days and in the days since, I have studied and studied the pros, cons, ins and outs of Direct Stream Digital. (1 bit PDM, or is that 1bit PCM? Depends on who you talk to I suppose). I have learned enough to be very dangerous. Which means I have crossed the point where I know how truly complex this subject is, and I know I am nowhere near an expert. At the same time, there are a few things that I do know fairly well, with decent confidence. So let us start there as we continue to stir up a little controversy in the name of FUN here at Euphonic Review.
How do you like your DSD?
Personally, I will take mine rare, as straight from the cow as possible, presuming the germs are all killed. I prefer "NATIVE DSD", but what does that even mean?? It seems every product maker has their definition, usually all in the name of selling products over accuracy. If you poll 1000 audiophiles, you will probably get 10,000 different answers. Not even the actual experts can agree, either.
What is MY definition?
Native DSD (not counting the production side, that is another can of worms) is when a DAC maintains the native, bit-perfect 1-bit signal, and the only conversion occurs at the end of the chain via a filter that leaves behind the music and cuts out the noise.
This can be divided into two camps, but only one of them seems to be realistically achievable, although it hasn't stopped many from trying. The 'purest' form I speak of, which is still out there like the Holy Grail waiting to be found, looks something like this (poorly put together) graph.
This extremely minimalistic design has only one bitswitch, which is usually the DSD output of a USB audio processor. Amanero Asynchronous USB input card is a common choice when attempting to build one of these 'dac-less' DACs. This 1 bit 'logic' output, can be treated the same as the unfiltered analog DSD square wave. It is of low signal level, but can be filtered by a RC analog filter to remove the carrier frequency and some of the noise-shaping. A good result will be more achievable with a very high DSD sample rate due to presumably less ultrasonic noise, but as of 2023, I know of no one who has been able to make this work well enough to produce a commercial example. I believe Lampizator came up with a version than used a very particular style of antiquated FM radio to do something similar, but it was too noisy to be viable.
The next 'camp' of Native DSD conversion is ubiquitous in modern Digital to Analog Conversion. I am sure earlier versions of this exist, but the Burr-Brown DSD1700 chip is what I think of as ushering in the SACD age.
This is what you will find in Native DSD products by iFi. It is the technique used by ROHM chips in DACs such as the S.MS.L D300. AKM and Cirrus/Wolfson chips use a similar technique in their Native DSD 'Bypass' Modes. It is also how the Signalyst DSC1 DAC works.
NO, it is not what ESS uses. ESS in this case reminds me of that old Sesame Street song, 'One of these things is not like the other'. More on ESS later. Back to our first two Native DSD 'camps'.
How are they different from one another? In principle, not very. In practice, however, yes, there is a difference, and whether or not this is 1-bit or multi-bit conversion is probably a fruitless argument, because in this configuration, the 'multi-bit' properties exist post coefficients, and are fully in the analog domain of this 'hybrid digital/analog' Fixed Impulse Response (FIR) filter. The output of the filter isn't a digital word, rather, it is an electrical current, whether that be from the switches/coefficients or at/after the accumulator. Again, we are not in the 'standard digital' realm.
FIR filter. There you have it. That is the difference. Rather than a single bit switch and an RC filter, the 1 bit DSD goes through what is a digital FIR filter, however it is implemented in the analog domain, either partially or fully. This FIR filter IS the Digital to Analog conversion. Pretty cool huh? Filter and converter all in one step. Here we still have the simplicity of no DSP, a 1-bit signal all the way until the very end when it reaches the Fixed Impulse Response 'Moving Average' filter. The 1-bit signal moves though a delay line, and the taps/coefficients ARE the bitswitches themselves, their outputs being summed up as the analog signal output that goes to your ears. (Note many of these type DACs also have a secondary RC filter to help deal with the ultrasonic noise).
Note, the graph is crude, but the basic concept of how it works is here, so don't go too hard on me...
So what is next?? What about ESS? Well, ESS is super quiet about DSD. They protect their proprietary info. It is VERY closely guarded, kind-of like a Fort Knox. And that is not a terrible exaggeration. All I can say, it is NOT QUITE like what you read above in the 'analog' FIR filter example. But we can make some educated guesses based on measured behavior, and the experiences of those who work with the chip in their kit.
What about AKM in non Native DSD mode? Cirrus? Well, they all use some variations on the same theme. 1 bit DSD is internally converted to a multi-bit signal. Sometimes it may be decimated to PCM, sometimes it may maintain the same sample rate. It is VERY common to convert 1-bit DSD into a multi-bit 'DSD derivative' via a digital filter.
However, unlike the FIR filter described above, this Filter is in 100 percent digital domain. These fully digital DSD filters are excellent and precise at filtering out the unwanted ultrasonic noise, and due to their multi-bit nature, and being in fully digital domain, other DSP is easily applied in this state. I.E. Sample Rate Conversion, Volume Control, further filtering such as IIR etc. One thing that MUST occur, though, is a re-modulation to some form of Delta-Sigma before the output. ESS would call this their Hyperstream DAC. DCS has a 5 bit (32 levels?) Delta Sigma modulator before its "Ring DAC", which is simply their take on a thermometer/unary code converter using scramble code/dynamic element matching. AKM has various types of Delta Sigma modulators in their various chips. Same for Cirrus/Wolfson and others.
I personally am not as big a fan of this because of all the DSP. Yes, it is true that PCM often goes through lots of DSP, too. But I am personally consistent in not liking that either. I like my PCM at high resolution with Non-Oversampling. Also, simplicity of conversion is one of the 'promises' of DSD. When you bring lots of DSP into the equation, it isn't so simple anymore. Transient response can suffer. The original 1-bit input and whatever is at the output are very, very different. This CAN be a good thing, though, as advanced Delta-Sigma modulators like the ESS Hypersteam can and often do convert DSD with a lower noise floor, lower distortion, and better linearity.
As I always say, in the end, let the ear be your guide!!
For a visual example of these types of DACs, I have gone old school, and will take a stab at showing how Sony converts 1 bit DSD into their 8 bit PCM 'DSD-WIDE'.
I hope you found this trip through DSD land edifying. Thank you as always for reading Euphonic Review!
Until next time,
As the reviews of Topping's AKM based E70 Velvet start to flood the internet, I have seen a few errors here and there. One of the more well known boards, which I will not name, gets something that is important very, very wrong. At the very least, I can set the record straight for my readers.
I thought about naming names, but, I decided intrepid readers can figure it out.
(By the way, if you are one of my readers, I express great gratitude to you. I never thought anyone would really read this new audio blog/review site, but the many thousands of you that have in just a few short months, well, you have overwhelmed me with gratitude for your time and attention.)
The incorrect information comes from a reseller product representative, not so much from the author. The salesman claims by using fixed output as opposed to variable output, the Topping E70V will use bypass mode for Native DSD.
This is incorrect. The E70V never uses the DSD bypass mode, set in the logic as DSDD= '1'. The E70V always, regardless of mode or volume, is set as DSDD='0'. See the diagram below for visual clarification.
There is only ONE DSD filter available in the E70V, and it has a 19khz fc at DSD64. This filter feeds the volume control DSP as a multibit 'intermediary' and is converted to lower bit Delta Sigma via the modulator.
The second DSD filter available with the AKM chipset is a 39khz fc filter at DSD64, and is only available with the 'bypass mode'. None of this is available on the Topping E70V, and this is confirmed by Topping representatives at the source. Not from a sales guy.
Here is the AK4191 diagram that helps spell it out...
Again, there is no alternative filter choice. Whether the Topping E70V is measured in Fixed output or Variable output, there is a single DSD filter with no ability to change it. This option IS available to any DAC that uses the AK4191+AK4499, but Topping does not provide this option in the E70V. Measurements confirm the 19khz fc, with full stopband around 30khz at DSD64. FC (frequency cutoff) will double as the DSD rate doubles. (38khz at DSD128, 76KHZ at DSD256)
It seems some people are confused by my wording and I apologize. To put it as simply as possible, AKM offers two possible DSD paths. A 'direct' path, using DSDD bit "1", that bypasses all Digital Signal Processing and Modulation. The DSD signal is filtered at the output stage with a 'analog' FIR filter, built of shift registers for the delay line and bitswitches for the taps, followed by a analog 'accumulator'.
There is a 'DSP' path that allows for Volume Control, using DSDD bit "0". In this path, the 1-bit DSD signal is first greeted by a digital FIR filter, the output of which strips the majority of the high-frequency noise shaping. The output of any such filter is multi-bit, which allows for the next stage, which is volume control. THEN the multi-bit signal, which has a longer wordlength than can be processed by the output stage, is converted to a lower bit depth at the Delta Sigma Modulator (the same one used for PCM, yes).
Running the Topping E70V in fixed output mode, disabling the volume control, does NOT switch the onboard DSDD logic from bit '0' to bit '1". The Topping E70 never makes use of the direct DSD mode. It is always in logic mode '0', regardless. Fixed output mode simply locks the volume control at 100 percent. The output path of Digital FIR and Delta Sigma Modulator is still followed, emulating what DACs such as ESS do. Why? Because this is more than just a technique to apply volume control to DSD. It can provide for a very precise conversion of DSD, with lower noise levels and taking full advantage of the advanced Delta Sigma output stage. Others, like me, still prefer to keep DSD conversion 'simple'. Keep it at 1 bit all the way until the final FIR filter converts it to analog.
Which is why I prefer DACs made by iFi, which is 1 bit native all the way to the analog FIR output stage, RME, which uses AKM like the Topping E70V, but has the user selectable option to use native DSD Bypass mode. Other DACs off the top of my head that are true native DSD... usually anything made with Burr-Brown DACs, T+A uses either Burr-Brown or their own proprietary output filter for true Native DSD, there seem to be some DIY and Chi-Fi options out there, and I know I am forgetting a bunch.
No ESS DACS don't make that cut. There is no 'bypass' option. All DSD is re-modulated in their Hyperstream converters. Indeed, the sound excellent, but they are NOT Native DSD converters in the strict sense.
In a couple weeks I will be a proud owner of a DSD only DAC built around the Signalyst discrete native DAC-less converter. I cannot wait to give it a spin!!
Also, I am re-evaluating some early scores in reviews that reflect measured performance. The emergence of superior evaluation technology and software skills require some rating changes, to be fair and consistent. Apologies for this, however I feel it is the best way forward for our fledgling publication, as we grow our identity and solidify our standards.
Ahhhh. There is nothing more satisfying than fixing, or rather, bypassing a bug in the system. You may have noted the issue in the white noise filter frequency response graphs. Something about the internal tone generator in the ARTA software creates a 'hiccup' at the Nyquist limit. It does not affect the accuracy of the measurements, at least those 'outside' the hiccup. It is almost as if the white noise started imaging above the original sample rate Nyquist limit as one looks at the oversampled filter plot. The filter quickly brings things back to normal, but that little anomaly shouldn't be there. There should be a smoothly rolling off plot.
Bypassing the ARTA internal tone generator with external tones fixed the issue. Along with fixing that issue, I changed the color patterns quite drastically, which, in my opinion, is much more pleasing to see. The white background with red and blue data plots simply looks better than the default green over black. At least here at Euphonic Review. Opinions vary of course.
See below the results. The GTO filter finally looks like it should. It resembles the MQA filter. No surprise since iFi collaborated with Meridian on this filter. It is personally my filter of choice, and Thorsten Loesch (when he was still actively the iFi engineer) explains it well in the tech notes, which you can access here by clicking. (GTO FILTER PDF)
Below are the iFi NEO DAC fw 3.33 filter profiles. Once again, the only anomaly here is the filter called 'Minimum'. One would likely interpret that as meaning 'Minimum Phase'. It is not a Minimum Phase filter, however. The measurements show it to be a low tap, short linear phase filter.
I hope you like the new filter graph format. Without any further delay, here are what the iFi NEO DAC 3.33 firmware filters look like.