Sept 25, 1951 through Nov 27 (28), 2023.
End? No, the journey doesn't end here. Death is just another path, one that we all must take. The grey rain-curtain of this world rolls back, and all turns to silver glass, and then you see it.......
White shores, and beyond, a far green country under a swift sunrise.
I have a lot of respect for Paul Miller and the work he does for Hi-fi News and their sister company Stereophile.
However, this is a pretty lazy take on the DSD1793 chip capabilities as installed in the iFi Diablo v2, and any iFi product that uses the 1793, for that matter. Paul is usually much better than this.
NO, the iFi DSD1793 engine is NOT limited to merely DSD64 and PCM 192. Perhaps in yesteryear; however in modern implementations the chip natively decodes DXD x2 and DSD1024.
The limitations exist only in the chip logic and clock. The easiest way to demonstrate this is with 1 bit DSD. The DSD1793 converter is a 1-bit FIR filter. To convert DSD 128, DSD 256, etc, all that is required is a doubling of speed into the converter/filter. The resistors (switches) in the filter know no difference. And later in the data sheet, you will find the DSD1793 bit clock timing runs as fast as 50mhz, which is just enough for DSD1024.
Thorsten Loesch, the designer of the iFi DSD1793 chipset/FPGA, has the following to say:
"Paul read the front page of the datasheet and didn't test.
Can't blame him. It's ultimately a failing by iFi's marketing people.
If you operate something outside standard parameters (which doesn't mean it risks damage or problems) you need to make sure to let people know.
The iDSD diablo 2 uses DSD1793 DAC chips. According to the front page of the datasheet, the Chip is listed as 192kHz and DSD64.
Looking inside the datasheet you can see that there is a "digital filter bypass" Operation mode that allows 768kHz PCM to be input into the chip.
Additionally, while only specified for DSD64, if you look at the DSD bit clock timing specification, it lists 20nS minimum Cycle, which is 50MHz.
Thus the DSD1793 is actually, according to the datasheet and if implemented correctly capable of 32kHz - 768kHz PCM and DSD up to 50MHz or DSD1024."
Bottom line... we love you here at Euphonic Review Paul. Your work stands alone above the rest for many decades. However, here a correction is required.
WOW! Its a true beauty. Extra high quality construction and an aesthetic I was not expecting from this Chinese company at this price point. In depth review with measurements coming soon.. I cannot wait to dive right in!
The S.MS.L D400 PRO DAC is in lab at Euphonic Review and it mighty impressive, I must say.
This will be the second DAC I have reviewed with the latest AKM chipset... the AK4191 + AK4499EX (not to be confused with the older AK4499 chip).
The first DAC I reviewed (which was a very fine DAC) making use of this latest AKM silicon was the Topping E70V 'Velvet'. A fine sounding DAC in its own.
Unfortunately, I started to see lots of really lazy reporting regarding its native DSD, or lack of native DSD implementation. The Topping allows for a fixed output, as do many a DAC. For some reason, otherwise gifted and respected reviewers gloss over the fact this does NOT mean the chipset uses its 'bypass' mode for native DSD.
The Topping E70V offers no choice of DSD filter (it has a single preset, non-changeable FIR filter at 19khz for DSD64 files). Nor does it offer any use of the volume/Delta Sigma Modulation Bypass mode. Simply being in fixed output mode does NOT bypass the Modulator. This was my own observation after studying DSD and its various implementations for years, plus I confirmed this directly with an engineer at Topping. Certainly you can set the volume control at full output for either format as such it has a 'fixed' output. However, for DSD the Delta Sigma Modulator is still in use and cannot qualify the DAC as true native DSD.
Not so the case for the AKM chipset as implemented in the S.M.S.L D400 PRO. It has THREE distinct output modes, as well as access to both 'WIDE' and 'NARROW' DSD filters. According to the manual (which while not great is better than anything I have seen from Topping), the THREE PRE-MODES are as follows.
* VARIABLE ---- The output is volume controlled. (all formats)
* FIXED ---- The output connector volume is fixed. (all formats)
* FIXED DSD BYPASS ---- Open the DSD direct access/bypass function. At this time, the volume of PCM or DSD in NOT adjusted. The output amplitude is fixed to about 3.7VRMS. When this function is opened, the DSD is NOT processed at all, and the output is directly output.
Now that this small detail is settled once and for all (well, it SHOULD be settled, anyway) I am looking forward to a weekend of listening comparing the D400 PRO with its little brother D300 with the ROHM chipset, as well as compare it to the RME ADI-2 PRO, while yes, having the older AK4490 chipset, has a similar activatable DSD modulator bypass mode as well. And finally, I am always interested in how it stands up to my favorite DAC of the last half-decade... the iDSD PRO.
Next item up at Euphonic Review to be thoroughly tested will be a S.M.S.L D400PRO DAC.
This will be a comparison test with the Topping E70V Velvet which uses the same chipset, and I am hoping to see more than single 19khz filter for DSD64, and I may really be strectching my hopes for a DSM 'BYPASS' channel selectable for DSD files.
The brand new AKM dual chip flagship DAC, the AK4191+AK4499EX, sounded extremely promising in the Topping E70V yet had a hint of grain to it.
The AK4191+AK4499EX as implemented in the Topping is still to this day the best measuring piece of kit I have yet seen. Combine that with the fact my measuring rig is more accurate and resolute than ever. I cannot wait to get this one strapped down to the test bench next week, haha!!
First, came the E1DA COSMOS ADC that brought to the masses who were willing to deal with the well, usability issues, very close to the power of the 'big-dogs' like Audio Precision to the home audio lab.
Then, came the APU (Audio Processing Unit), that added an analog 1khz/10khz notch filter to get THD and THD+N measurements even closer to those big dogs.
Finally, now the Euphonic Review lab has the EIDA COSMOS SCALER, which is a buffer and scaler that will further refine our in lab measurements.
I have read from a reputable source that we are talking about accuracy and quality of measurement somewhere between the Audio Precision SYS27XX and APx555, with the addition of the this new auto-scaler with a high enough impedance to accurately measure pretty much any DAC on the planet, is probably closer to the APx555 in performance!!! Not in features, mind you. The Euphonic Review lab will be limited in features, but what features we DO have we consider very, very useful and more than anyone making purchasing decisions really needs.
The GREATER point is, that as good as our measurements have been so far, with the addition of this new Scaler, we will be on an accuracy level that competes with anyone. You name it. Online or magazine. I know that is a big statement, but the reality is just finally here that the hardware has trickled down to the 'little guys' and there is no more monopoly on state-of-the-art audio measurements.
But measurements have NEVER been the primary source of pride here at Euphonic Review. Our source of pride has always been our HONEST reviews. And what we believe to be excellent 'audiophile' ears to go with it. That is an explosive combination, and combined now with impeccable measurement equipment, we think that explosive combination should be going BOOM at any time. We may be the new kids on the block, but we think we have a helluva lot to offer.
Thanks, so many thanks to those who have read from the beginning, suffered through our growing pains (which will surely continue as running a webpage ain't easy), and endured and hopefully enjoyed.
Here's to the rest of 2023 AND to an explosive 2024!
An FIR filter is like a simple sieve at the end of the slide. As the marbles roll down, the sieve lets only the blue marbles through.
An IIR filter is like a smarter, but more complicated, sieve that remembers some of the previous marbles. It might let a blue marble through, then think, "Hmm, the last few marbles were blue, so I'll be a little more lenient with the next one."
Both types of filters are used to process audio signals, either to enhance certain features or remove unwanted noise, much like how our marble sieves are used to sort the marbles.
DO IIR Filters have Delay Lines like FIR filters??
In digital signal processing, the term "delay line" is often associated with FIR (Finite Impulse Response) filters, where the filter uses a series of past and current samples to compute each output. Essentially, the delay line in an FIR filter holds onto past samples of the input signal, which are then used along with the current sample to calculate the output.
IIR (Infinite Impulse Response) filters, on the other hand, use not only past input samples but also past output samples to compute the current output. While they don't have a "delay line" in the same sense as FIR filters, they do have a form of memory that stores past values. This "memory" isn't typically referred to as a delay line, but it serves a similar purpose: it holds past information that the filter uses to calculate its output.
So, in simple terms:
Both kinds of filters can introduce some amount of delay to the signal, but the way they use past information is different. FIR filters rely solely on past and current input values, while IIR filters use both past input and past output values.
What is a Cascade Comb type Integrator vs. a Moving Average Filter explained very simply?
Let's imagine you're trying to figure out how fast a group of people are running. You could use different methods like a Cascade Comb Type Integrator or a Moving Average Filter. Both aim to give you a general idea, but they work a little differently.
Moving Average Filter:
Think of this like taking a quick glance at your stopwatch every few seconds and averaging those speeds. This method will give you a pretty good idea of how fast people are running right now.
Cascade Comb Type Integrator:
Imagine instead you're using a more complex method where you don't just look at the most recent times but also consider times from way back, maybe even from when the race started. This would be like calculating your grade point average (GPA) throughout all your years in school, not just the recent tests.
Both have their pros and cons, and the right one to use depends on what exactly you want to know.
How are Comb Filters used in Audio Processing?
In the context of audio signal processing, a Cascade Comb Type Integrator (often referred to simply as a "comb filter") is a filter used for various applications like reverb, echo, and other time-domain effects. It's different from a moving average filter and has its own characteristics.
How it works in audio:
A comb filter adds a delayed version of the signal to itself, creating peaks and troughs at regular frequency intervals (hence the name "comb"). When cascaded—or chained together—multiple comb filters can provide a more complex and tailored frequency response.
What does it do to the audio?
Imagine you're in a hallway and you clap your hands. The sound of the clap will bounce off the walls and come back to you at different times, creating an echo or reverb effect. Each bounce is like a single comb filter. When you cascade them, it's as if you're in a more complex environment where the sound bounces off multiple surfaces at various distances, creating a more intricate echo or reverb effect.
So, in audio, a Cascade Comb Type Integrator (or cascaded comb filter) is often used for creating complex time-based effects like reverb or echo. It's different from a moving average filter, which is often used to smooth out a signal but doesn't create the time-based effects that a comb filter does.
How are these Cascade Comb Filters used in DSD Processing?
Digital Stream Digital (DSD) is a high-resolution audio format that uses Pulse-Density Modulation (PDM) instead of the more commonly used Pulse-Code Modulation (PCM). In DSD, audio is represented by a series of 1s and 0s, where the density of 1s indicates the amplitude of the signal. It's a bit like morse code for audio; a bunch of 1s close together represent a high point in the audio wave, and a bunch of 0s represent a low point.
A Finite Impulse Response (FIR) comb filter can be used in DSD audio processing to achieve certain effects or improvements in sound quality.
FIR Comb Filter in Simple Terms:
Imagine you are jumping on a trampoline. Each time you bounce up, you touch a hanging bell, which rings. If you bounce higher, you can touch bells that are hanging further away. The first bell represents the immediate sound, and the bells hanging further away represent echoes or repetitions of that sound.
In a similar fashion, a comb filter creates a series of echoes or repetitions in your audio. An FIR comb filter does this by adding delayed versions of the original signal to itself.
Application in DSD:
So in essence, an FIR comb filter in DSD can help in shaping the sound, either to improve quality or to add specific effects. It's like choosing which bells you want to ring as you jump on the trampoline, each creating a unique bounce pattern and sound.
But isn't is still true a Cascade Comb filter is a type of Moving Average Filter?
You're correct that a cascaded comb filter can be viewed as a specialized form of a moving average filter, particularly when it's implemented with Finite Impulse Response (FIR) characteristics. Both are linear filters, and both can be used to smooth out or otherwise modify a signal. However, the way they're commonly used and their resulting effects can be quite different.
Moving Average Filter:
In a moving average filter, you average the past 'N' samples to get the current output. It's like looking back over the last few seconds to see how fast you've been running on average. This kind of filter smooths out sudden jumps in the signal and is often used for noise reduction or signal smoothing.
Cascaded Comb Filter:
A cascaded comb filter, on the other hand, takes the current sample and adds to it a delayed version of itself, possibly scaling it by a factor. When you cascade multiple of these comb filters together, the effect becomes more complex, but the basic operation is similar: current sample plus delayed sample.
How They Relate:
Imagine you're checking the temperature every day. A moving average filter would be like saying, "The average temperature over the past week is 'X' degrees." This gives you a smoothed-out view of what the temperature has been like.
A (single) comb filter would be like saying, "Today's 'feels-like' temperature is a combination of today's actual temperature and the temperature from a week ago." A cascaded comb filter would then involve multiple such 'feels-like' calculations based on temperatures from various past days, leading to a more complex profile.
Now, if you set up a cascaded comb filter in a certain way, it can act similarly to a moving average filter. But they are usually employed for different purposes: moving average filters for smoothing, and comb filters for adding echo, reverb, or other time-domain effects.
So, while you can view a cascaded comb filter as a specialized type of moving average filter, in practice they are generally used for different tasks.
Andrew's comments on the above
The Cascaded Comb filter, a special case of moving average filter, is the type of filter used to filter DSD in native DSD DACs. Multiple streams of the original 1-bit DSD stream are 'cascaded' upon each other, delayed by a sample.
The Signalyst DSC Native DSD DAC has 32 of the 1-bit DSD streams cascaded on one another, offset by a single sample each. This is a 33 level unary/thermometer coded FILTER/DAC. Burr-Brown Native DSD chips like those in iFi products do something very similar, the difference I wish to highlight here is the amount of levels... there are only 8 cascaded/delayed streams, meaning a 9 level unary/thermometer coded FILTER/DAC. Finally, I will mention T+A, the outstanding high end audio hardware producer out of Germany. I can only repeat what I was told by one of their representatives, but their Cascade Comb filter has only 4 offset DSD streams, for a 5 level unary/thermometer coded FILTER/DAC.
Why do the call them 'comb' filters? Because when viewed in FFT , they look like this... a bit like a comb. (this is the S.M.S.L D300 native DSD filter output at DSD64, fc at 13khz. )
I named the S.M.S.L D300 DAC my current King of Budget Esoterica. It sounds way, way better than just another one in a line of nice budget kit, as long as certain conditions are met. Rather than re-hash all of that, I encourage you to click here and check my review if you have not already done so.
The issue isn't with my ears. I can hear just fine even as I approach 50 years. Not that it helps with reviewing audio equipment, it can't hurt that I am a classically educated college level musician with perfect pitch. It's nothing to me to notice overtones, also known as harmonics, both what should be natural to the music, and what is 'added' as harmonic distortion by the 'bad, bad' equipment. I never have really understood the aversion to harmonic distortion. It's, well, HARMONIC. The first harmonic is the fundamental, the second harmonic is the Octave, which means its the exact same note just twice the frequency rate. The 'mean old nasty' third harmonic? Its a perfect fifth above the second harmonic, and therefore is also consonant with the fundamental.. they call it perfect for a reason. Fourth harmonic? Its a perfect fourth. Which guess what? Is the exact same note as the fundamental and the second harmonic! The are all climbing in an octave sequence. Yes, I get that equipment that adds or accentuates these things MAY not represent the musical performance accurately. But I also 'get' that I LOVE the way music sounds on tube amplifiers. I have NO technical or scientific argument or treatise to give you on what makes for the best musical reproduction. All I know is I know what I like when I hear it. It stands out from amongst all kinds of 'copy cats' in a saturated market.
You know what else I like? I am about to welcome you into the cognitive dissonance in which I daily live. I like to know what my hardware is doing on the electronic level. Measurements mean things to me. Surprisingly, they are important. I mean, we have to have SOME BASIC standards, right? I think we passed those benchmarks a long time ago actually, and the crazy low noise and distortion stuff out there is way past our ear's resolution. Yet no one is suddenly claiming digital audio is cured of all its 'digititis'. No one, except maybe a few of that same old cult are proclaiming perfect audio. Same guys who have been yelling 'bits are bits' since the CD hit the scene. (Reminds me of an episode from back in the early 90's when a friend of mine insisted that CD was the thing of the past.. DAT was the real deal and the future... I will let you laugh and clear before we move on. )
So what is my problem?? I am obviously talking about the S.M.S.L D300 DAC. Its PCM measurements are impeccable and it doesn't sound in any way clinical or digital; it frankly sounds amazing.
The PROBLEM came when I started seeing the DSD measurements. The DSD filters are very low, and very slow and gentle. All good so far for the DSD crowd, like me, who wants gentle filters to maximize transient performance and minimize ringing. DSD should be different than PCM. Many DACs just make DSD as much like PCM as possible without actually saying that to the buying public in order to quietly deal with its major issue... all that ultrasonic noise. Ummm... THAT is where we run into the problem with the S.M.S.L D300. Something is off with the analog stage.
The numbers not only are worse than their PCM equivalents, as we go up in oversampling speed, they get much worse. Certain websites would be panning this thing with trophies that have missing parts.
My best guess is the issue is in the secondary analog filter that should come after the initial FIR conversion filter. For instance, iFi has their DSD FIR filter converter followed by a RC analog filter at 80khz to keep the DSD time domain superiority intact while dealing with the remaining ultrasonic noise in a way that adds no new ringing at all, and maintains the signal to noise ratio and harmonic distortion at levels that match its PCM performance. Same with pretty much every other 'native' DSD DAC I have sitting around.
But not this S.MS.L. And damned if it defies the measurements and still sounds amazing anyway, especially at the 'sweet spot' I identified.
What would normally be sacrilege for a 'purist' like me, I caved and am oversampling everything to DSD256 and have chosen to set the filter at 104khz. And it sounds GLORIOUS. And it sounds about as good, as anything I have, including the RME ADI-2 PRO FS R BLACK EDITION in DSD Bypass mode, which not only sounds great, it measures the part. The only native DSD kit I have that really kindof makes quick work of the S.M.S.L is my Signalyst DSC2 discrete DSD converter build, which ironically really doesn't measure that well either since it uses good ol' fashioned transformer outputs like the good ol' tube days.
So with my rant over and my cognitive dissonance as active as ever, I will share with you what the DSD measurements from the S.M.S.L look like on paper. These tests are done either with Multitone's internal modulator that converts all its test tones to DSD, or they are DSD tones tones made by HQPlayer played back via external sources. ADC in use as always is the E1DA Cosmos, so we are in good technical hands here I assure you. Allright.. on with the show.
The rest I will present to you below, so you can see the changes the mostly unseen ultrasonic noise makes inside the 20hz to 20khz band.
I reiterate as bad as some of this might look, it still sounds, well, pretty darn good. There is NEVER anything audible in the noise floor. Never any noted artifacts. Never any noteworthy distortions.
What I believe I CAN surmise that is happening here.... truly excellent time domain performance. Likely very little digital ringing from filters, harmonic distortion that is innocuous or even pleasing to the ear. Masking effects that keep any other nasties or idle tones from being a problem.
And now? I am going to continue to enjoy listening to this merely $400 DAC that has engrossed me with some of the best DSD playback I have heard for the last two hours I have been writing this Blog entry, while I hope you enjoy going through the remaining graphs in the gallery.
Thank you so much for taking time to read my small corner of the interweb here at EUPHONICREVIEW.COM
I thought you might like a look at some data charts pulled from my archives... We all love (or many of us just quite frankly hate) more charts and data! I am starting to develop that love/hate relationship with it... leaning toward the 'hate' part, because every second I am taking measurements or preparing them for publishing is a second that I was at one time and could be at this time actually enjoying the hobby of LISTENING! Maybe someday I can afford what those fancy shmancy sites have.. those people who do things for you for pay. Alas, for now, it is me doing the measuring AND inputing the data.
I came of age with computers as a teenager in the early to mid 1990's. I have been there for it 'all'. Well almost anyway. From the days of 'Quantumlink' dialup on the Commodore 64, to the 'You've Got Mail!' sound every time I logged onto to America Online! from my mid 1990's Macintosh Performa, all the way to the invention of the iPad, Smartphone, etc, and I still feel like a complete idiot around Gen Z who whiz around their smart technology like their smartphone was grown with them in the womb.
That is a long way of saying, yeah, I am more computing challenged than I would like to admit, being a 'pioneer' and all that haha.
Anyway.. all that rubbish said, here is an early preview of the measurements of the iFi iDSD PRO 4.4mm Pentconn output version. Full review finally coming early next week.
Over the past few months I have reviewed several DACs, all of which are DSD capable. Not all of them, however, are what I would consider a 'native' DSD DAC.
'Native' DSD DACs have some differences in their various implementations, but common to ALL of them is digital to analog conversion via an analog filter at the end of the signal chain. I think this rather simple fact is where a lot of people get hung up; however, it is this fact that makes 'native' DSD conversion what it is; an on/off square wave bitstream that reveals the musical signal when band limited by an analog filter.
Unfortunately, this is where what should be an easy conversion to analog becomes somewhat difficult. The DSD ultrasonic noise monster rears its ugly head. The best analog designs will follow the conversion filter with a secondary filter to better manage the noise, because the initial analog filter is virtually always of the 'moving average' type, and is designed to be at its best in the time domain, which is exactly what DSD is - a time-splicing format. Making a filter with lots of taps, or using unequally weighted taps/switches are but a couple of the means used to improve the frequency domain performance, however a balance must be found between time and frequency domain, lest the DSD superior time domain performance be lost. Compromise too much for the sake of frequency domain, and the lines between DSD and PCM really begin to blur.
Finally, one must deal with how any remaining ultrasonic noise affects the 'in-band' audio. This is no trivial matter, because even after the best filtering choices, there can be and there is enough ultrasonic noise left to cause artifacts in-band, exactly where we don't want them! These ultrasonics can and will cause intermodulation distortion, idle tones, and linearity problems.
It is no wonder that the Schiit Audio company in its earliest days made a DSD 'only' DAC because of these various issues. The analog output stage must be optimized for DSD, and here we are with DACs today that use a single analog stage that must somehow effectively passthrough both PCM and DSD formats! Schiit's valid concerns aside, there are indeed many DACs today with a single analog output stage that are very effective passing true 'native' DSD and 'standard' PCM. However, the very issues mentioned by Schiit are exactly why some DSD compatible DAC's do NOT have a 'DSD direct' mode. These 'non-native' DSD converters can optimize the DSD input, add in digital filtering, volume controls, and can re-modulate with their highly linear, high resolution multi-bit Delta Sigma outputs where all is then ready-made for their optimized 'one size fits all' analog output stage. This solves many problems and adds back conveniences that do not exist with 'native' DSD. And indeed does sound quite good if not outstanding in the end. But it isn't 'native' DSD, although many will argue that point. In reality what these DAC types do is actually easier than 'native' DSD strictly defined. It is therefore ironic that the format that can be converted more simply than any other, in the end can become so very difficult.
So why bother with the crusade for the simplest 'native' conversion?
Because: when done well, in my opinion no other other method of DSD conversion sounds better.
Furthermore, if you have been bit by the bug of software conversion to DSD via Roon or HQPlayer, a 'true, fully native' DSD DAC is a MUST. One is especially lucky if he or she has available to them a build of the open hardware Signalyst DAC made of completely discrete parts, that will compete anyday with such commercial discrete converters as those made by DCS.
The DACs I have had in the Euphonic Review lab which convert 'natively' are all iFi DACs with the Burr-Brown DSD1793 chipset, the S.M.S.L D300 DAC with the latest ROHM BD34301EKV chipset, the RME ADI-2 PRO FS R Black Edition AD/DA with the AKM4493 'pre-fire' chipset, and the Signalyst DSD only converter with proprietary chipless converter, which will be looked at at a later date in its own dedicated thread.
All of these DACs use a certain variation on the same theme. The 1-bit signal at the input is completely unadulterated until the conversion filter. These conversion filters are 'moving average' filters, with the slight exception that many of them will use unequally weighted taps/switches that allow for change in frequency cutoff while maintaining the same bitstream speed. A great example of this is the DSD1793 used in the iFi converters. Although iFi in recent years chose to software lock-in a single DSD filter, changed only by bitstream speed, the DSD1793 as implemented in their legacy products can change frequency cutoff regardless of bitstream speed. I have no direct knowledge of how the remainder of these chips select their frequency cutoff and rolloff, but simply observing their behavior, I would say most of them are doing something similar. (The Signalyst, as best I can observe, which will be covered in greater depth in its own future review, uses only equally weighted switches and frequency cutoff can only be changed by a doubling/halving of the bitstream speed.)
Regardless of the finest details, all moving average filters are a form of FIR (Finite Impulse Response) filter, and have a delay line, taps, and accumulator. The 1-bit stream is converted into parallel streams, offset by one sample in time. The taps are the output bit-switches, regardless of whether or not they are resistor or capacitor based. This produces an analog current output that when summed together (and converted from current to voltage) is the filtered analog audio signal. Further filtering can (and likely should) be done by a following analog RC filter to remove more of the ultrasonic noise without having to resort to a steeper roll-off by the initial conversion FIR filter. This allows for higher quality analog output stage performance. This can be even more significant with higher DSD rates, which will have by nature a higher filter cutoff and can possibly allow, perhaps counterintuitively, more ultrasonic contamination in-band. With a double filter arrangement, ultrasonic noise is dealt with in a effective and clever way; the secondary analog RC filter will not add any ringing and allows the primary filter to have gentle roll-off characteristics.
I have made a project of measuring every 'native' DSD DAC I have in my current inventory at Euphonic Review. The results are quite revelatory, and actually in some cases reveal the disconnect that can exist in what we hear and how a device measures. Even the worst measuring DSD signals still sounded very, very good. Would it surprise you that the worst measuring signals often are at DSD512 and higher? That the 'sweet spot' by my best deductions poring over the data MAY be at DSD256? At least in my small sampling of DACs. However small the sample, they do represent a wide range of implementations and hardware. For example, we have the new ROHM chipset, which I don't think S.M.S.L did for its DSD performance the most favors in the output stage. Nevertheless, it still sounds very very good, but certainly could be made to sound even better. Also represented is the DSD1793 which carries the same tech pioneered by Burr-Brown in the mid 1990's for the then upcoming SACD format. AKM is here in the form of the AK4493 DAC chip, with a truly excellent analog implementation by RME. Very, very impressive. And finally, one of the most active companies in the DSD software sphere, Signalyst, is represented in a build of its open source fully discrete moving average filter DAC. (Signalyst results to come at a later date.)
Lets begin with the S.M.S.L DAC. It is the most 'tunable' of the DACs here, with 3 filter choices. However across all possible DSD rates, those 3 filters offer 3 different cutoffs per rate, for a total of 6 possible frequency cutoffs. (Remember, they overlap, adding only 1 new, higher cutoff per rate.) This comes in handy, because the greatest DSD performance swings come with the S.M.S.L DAC.
The following charts are sorted by ACTUAL FIR filter cutoff rate. NOT by the cutoff as labeled on the DAC. For instance, if the Data Rate is DSD128, the filter MARKED at 13khz is actually running at 26khz. If the Data Rate is DSD256, the filter MARKED at 52khz is actually running at 208khz. If the Data Rate is DSD512, the filter MARKED at 13khz is actually running at 104khz. Got it? It may be confusing at first, but once you grab hold of the concept, it's no big deal. DSD 64 cutoffs equal 13, 26 and 52 khz. DSD 128 cutoffs equal 26, 52 and 104 khz. DSD256 cutoffs equal 52, 104 and 208 khz. DSD512 cutoffs equal 104, 208 and 416 khz cutoffs.
So in theory, DSD128 played back with Filter 1 (13khz x 2= 26khz) will have the same filter profile as DSD64 played back with Filter 2 (26khz as marked). Indeed, they do have the same filter profile on the spectrum analyzer, but DSD128 has an advantage. The rise in ultrasonic noise starts later, therefore more of the noise is eliminated by the filter, meaning less distortion 'in-band'.
SINAD is the measurement I have chosen to show these changes in fidelity, and indeed doubling the DSD rate while keeping the filter characteristics the same causes a notable increase in SINAD performance, from 96db at DSD64 (26khz filter) to 99db at DSD128 (26khz filter). As far as SINAD in general is concerned, once the 100db SINAD mark is reached, this is an excellent result. As a matter of fact, the DSD1793 used in iFi products is rated to max out right at 100db SINAD. It is true that the latest, most state of the art chips from ESS, AKM, ROHM, etc can well exceed 100db SINAD. However once you get past the 90db SINAD mark and approach the 100db SINAD mark, you are in true high fidelity land. Beyond that we are in 'space cadet' category where performance metrics start becoming more academic and less practical, especially as we are starting to see SINAD breach the 120db line and steadily marching to the 130db benchmark.
Thankfully the S.MS.L D300 offers many choices of filters, especially if you rely on computer software oversampling to maximize your performance. No, this method of software oversampling is NOT native DSD per se, but it IS the way to get the best performance out of this DAC. I do not hold the ROHM chipset at fault here; I believe the fault is with the S.M.S.L analog implementation. Every other DAC tested here has DSD and PCM performance that match very closely. The S.M.S.L D300 goes down the path of truly exceptional PCM performance, combined with 'merely' decent to very good DSD performance, depending on the oversampling rate and cutoff frequency. I believe the observed phenomenon here is due to the lack of a well implemented secondary RC analog filter, or perhaps none at all. I could be incorrect, and am open to correction, however, something in the analog stage is preventing DSD from reaching its fullest playback potential.
However as was mentioned earlier, there is a notable drop in performance in DSD performance as compared to PCM with the S.M.S.L DAC. SINAD performance with PCM files approaches and exceeds 114db. The best DSD performance, 104db SINAD, is at DSD64 with the 13khz filter, which is quite too low a cutoff for high-fidelity audio. Indeed, the filter is of the very gentle moving average variety; nevertheless by 20khz the rolloff has exceeded 8 decibels! This will cause a very audible treble rolloff. For DSD64 files, surely the 26khz filter and its 96db SINAD is the minimum filter of choice. A far cry from the excellent 114db PCM SINAD, but thankfully it is still good enough that any loss of enjoyment is likely all in your head, because I told you so! If you had never read these measurements, I would be quick to bet if you listened to the D300 you would have heard nothing negative at all. Ah, the power of simple suggestion....
And as we stick to the same 26khz filter with 128x DSD rate (by selecting 13khz DSD filter from the menu), we see that there is a noteworthy performance increase up to 99db SINAD with DSD128 due to the ultrasonic noise rise starting much farther from the in-band frequencies (20hz to 20khz), pushing the noise to a higher level where the filter can more effectively eliminate it. Although DSD64 had the very best SINAD measurement of any at 104db, again the filter rolloff is too low to be considered as high resolution audio. S.M.S.L D300 DSD playback is MUCH improved with the DSD128 and DSD256 formats. I see no need to oversample to the DSD512 rate. The most obvious reason being the poor performance at the two highest filter settings. The next reason? There is no SINAD improvement at all over the DSD256 at 104khz filter (26khz labeled in GUI). It is simply a waste of resources to oversample to DSD512 in this case, since you will not see any performance gains. The filter profiles are the same, meaning you will likely not see any gains at all in the time domain, and the noise/distortion performance in the audible band is identical in both. Here we may be running into a very common issue in DACs and 'ultra high speed' DSD.. the ability of the logic itself to keep up. Faster is NOT always better, and it looks to me that the sweet spot for the S.M.S.L DAC DSD converter is DSD256 at either 52khz or 104khz filter. I personally would go with the 104khz filter, to take advantage of the temporal resolution DSD offers while still having good enough SINAD performance. I can totally understand the thinking that would lead one to choose the 52khz option and get that SINAD up close to 100db. Either way, the DSD sweet spot for the S.M.S.L D300 is DSD256, and choose either filter 1 or filter 2 on the GUI (13khz, 26khz).
As I mentioned before, I prefer 'native' DSD at all sample rates, what one might call 'bitperfect'. However, the S.M.S.L D300 is all over the place DSD performance-wise. I will reiterate what I believe is the source of this 'issue', and that is a lack of a secondary analog RC filter following the DSD FIR conversion filter. Yes, one can achieve 'good enough' performance without a secondary filter, and one can 'tune' in the best performance with multiple DSD filter options available on the D300. However, I think the added expense of a secondary filter (or a better performing one) would have been worth it. Bad call here by the engineers. I am on record calling this little DAC a 'giant killer', and this is still especially true for PCM file playback. Its native DSD playback, in spite of any compromises is still quite good, but as you see the performance of the next couple DACs it will become obvious exactly how good the D300 COULD have been.
Now we move onto iFi, where thankfully we can get the temporal resolution promised by the DSD format.
Unlike the 'dac du jour', Thorsten Loesch and iFi didn't just stick with whatever is currently 'popular', such as the latest ESS or AKM chipset. There is more to overall performance/sound than sub-arctic THD and/or stratospheric SINAD. There is still something to be said about overall architecture, in this particular case the Segment DAC architecture of the Burr-Brown DSD1793 chipset. The DSD1793 uses a 64+ level thermometer/unary code DAC, which allows for the top 6 bits of the 24 bit two's complement PCM to be converted directly via bit-perfect PCM (64 levels), with the remaining 18 bits are converted via a 1 bit Delta Sigma converter. This is a particularly ingenious means of conversion, as it avoids the so called PCM zero-crossing error by keeping the 'lower' 18 bits always at "full volume". Obviously it isn't at full amplitude in the time domain, however in the frequency domain zooming down to the sample level, yes, the 1-bit signal is either full on or full off. The chip will use oversampling/time-splicing for the lower 18 bits of resolution. As has been mentioned, according to Burr-Brown, SINAD is expected to be right at 100db.
Another benefit of this type of conversion is it is ready made for 'native' DSD conversion. The 64 level thermometer/ladder bit switches are all unary coded/equally weighted. The delay line for the DSD conversion is only 8 bits long, however. This allows several different combinations of the 64 switches to create 8 groups of switches that function as the 8 unequally weighted taps. These different combinations can mix and match to change the cutoff frequency and the order (steepness) of the frequency rolloff. All the while the 64 individual switches are equally weighted and allow for dynamic element matching and exceptional linearity.
Note this excellent SINAD consistency continues within the iDSD PRO. At all bitrates and their accompanying cutoff frequency, SINAD measures right at 99db.
In addition to the DSD64, DSD128, and DSD256 rates shown above,
at DSD512 the cutoff frequency is 616khz, and the SINAD stays a consistent 99db.
At DSD1024 the cutoff frequency is an astonishing 1,232khz (1.232 megahertz), and still maintains a SINAD of 99db!!!!
As you can see, the iFi products above have output stages that offer equivalent maximum performance via DSD and PCM. (The DSD1793 chipset, repeated here like a broken record, is limited to around 100db SINAD.) The iFi ZEN and iFi iDSD PRO are not configured to have switchable DSD FIR filters, however, their filter cutoff doubles with each oversampling. For example, DSD64 has a -3db cutoff at 77khz. Anything at DSD64 will always have a cutoff at -77khz. Double the speed to DSD128 however, the cutoff will double to 154khz. And this pattern continues right up to DSD1024 on the iDSD PRO. (Note the DSD1793 does indeed offer switchable cutoffs for each DSD speed rate. In recent years, however, iFi has settled on locking into what appears to be the 77khz FIR filter onboard the 1793 chipset, along with an 80khz Analog RC filter to follow.)
And finally in this particular blog entry we will take a look at the very, very impressive RME ADI-2 PRO FS R BLACK EDITION with the AK4493 chipset (pre-factory fire switched capacitor version.) Just as the iFi with its DSD1793 matches PCM and DSD performance, likewise this much more modern AKM chipset matches its PCM and DSD performance. This MAY be the highest performing true native DSD chipset on the market that is NOT made from expensive discrete parts ala the Signalyst Converter.
Note that the following AKM 4493 based DAC boasts an outstanding 114db measured SINAD with PCM material. The 'native' DSD material comes very close to this level of performance. One of the first things to note, however, is the DSD filter numbers in the GUI are incorrect. They are not 50khz/150khz. This was correct for the previous chipset used by this AD/DA, the AKM4490. The AKM 4493 used in this particular upgraded AD/DA uses DSD64 filters at 39khz/76khz.
What more can be said? Used in DSD Direct Mode (Native DSD), DSD256 using the 'high' filter, in this case at 304khz, while maintaining a 111db SINAD, is truly remarkable performance. This is an astonishingly good choice of DAC for those who use software oversampling. Oversampling in Roon or HQPlayer to DSD256 with the higher of the two filter settings (304khz) with a SINAD over 110db should offer state of the art performance, and then some.
Also, if you are more of a purist, as I am, and wish to have as little DSP touching the DSD stream as possible, The RME ADI-2 PRO FS R BLACK EDITION is also an excellent choice. You can have native rate conversion with every filter save one between 110db and 112db SINAD. The only exception, is the high 76khz filter on DSD64 material. This is not unexpected, as there is a significant amount of ultrasonic noise that escapes the lowpass filter, yet the actual performance penalty is very, very small, as the SINAD is still a high 107db!!
The RME team have built an exceptional native DSD DAC. On an aside and maybe for a future entry, I will publish the RME results with DSD in Volume Control Mode and remodulation to AKM's multi-bit Delta Sigma format. The performance here, even with DSD, is also quite outstanding. The RME continues to impress. Full review coming sometime this year.
MORE TO COME SOON, INCLUDING THE FULL REVIEW OF THE iFi IDSD PRO (part two of the review; part one reviewing the ifi iDSD NEO is currently posted. )
Also a full review of the Signalyst PURE DSD DAC is coming soon! After much listening and mixing/matching amps and headphones, the Signalyst Converter may indeed produce the very best sound I have ever heard.