As the proud owner of one of the best turntable values on the market, the Rega RP3 (ca. 2015 vintage), I am always on the lookout for potential performance modifications. These range from highly effective mods (if indeed expensive) such as the NEO MK2 speed controller, Audio-Technica MC line contact cart/stylus, aluminum sub-platter and Rega reference belt, all the way to the least expensive of mods, such as the Hudson Hi-Fi 4mm cork turntable mat. While I have greatly improved the performance of the base Rega RP3 turntable, one annoyance that always seems to rear its ugly head is static. I have invested quite a bit in static control, such as the Milty Zerostat 3 antistat gun, the Audioquest antistatic brush, and even a humidifier for the super dry conditions in winter. I can deal with the occasional pop and click that is inherent to the vinyl itself; imperfections are part of the game. But the constant crackle of static? It just ruins the experience for me. While I was perusing the products offered by Hudson Hi-fi, I came across their "Vinyl Record Cleaning Arm - Anti Static Brush for Vinyl Records." It was a mere $24.99 and was deliverable by Amazon Prime. What could go wrong??? Boy was I pleasantly surprised by the results. So much so, this diminutive, very inexpensive little device is only the second piece of gear to ever receive the Euphonic Review Editor's Choice Award. ------------ The Hudson HiFi Anti-Static Arm Brush is a useful addition to any vinyl record cleaning kit, designed to enhance the quality of record playback by reducing static and dust accumulation. This device automatically cleans records as they play, starting from the outer edge of the LP and moving to the inner grooves, effectively removing dust before it can reach the stylus. Its operation does not affect the tracking force or tonearm operation of the turntable, ensuring seamless playback. The brush itself features carbon fiber bristles at the center, which are known for their anti-static properties, helping to eliminate static and reduce the pops and scratches associated with vinyl playback. Included is a grounding cable. (As my Rega table is grounded via the RCA outputs, I chose to ground the Anti-Static Arm Brush to the ground connector on my iFi ZEN Phono.) Manufactured with a focus on sustainability, the brush bristles are made from goat's hair, which provides wider coverage and is gentler than synthetic materials typically used in other brushes. The Hudson HiFi Anti-Static Arm Brush is praised for reducing wear on records and styluses, thus extending the life of your vinyl collection while enhancing your listening experience. Not only will it reduce wear and tear, STATIC is mostly a thing of the past. The last few days spent listening with the Anti-Static Brush Installed has been pure bliss. I have yet to hear vinyl playback in my home as good as this. NOTE... I received no payments or endorsement benefits of any kind to say nice things about Hudson Hi-Fi. The products I have tried of my own free volition work extremely well, and EUPHONIC REVIEW recommends them HIGHLY.
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Exactly how ESS DAC's process DSD has been kept a bit of a mystery by ESS, and certain players in the industry (Mytek and Benchmark come to mind) get the nature of conversion, especially the nature of the volume control, egregiously wrong. For a BAD example of how the ESS chips implement DSD volume control, Benchmark, who otherwise is so excellent, gives us what sounds like more like a wild guess than reality. from the Benchmark website: "The net result is that the 6-bit converters have the near-perfect linearity of a 1-bit converter while achieving an 18 dB reduction in noise (due to the 64:1 parallel structure). This improvement delivers a 6-bit sigma-delta modulator that has an 18 dB noise advantage over a classic 1-bit sigma-delta converter (such as that used in DSD). The array of 1-bit converters also allows native DSD conversion with digital volume control. This combination of features is very unusual, but the ES9018 provides a unique solution to the DSD volume control problem. Normally, it is very difficult to implement a digital volume control (or any other form of digital processing) in a 1-bit DSD system, but with an array of 1-bit converters, we can set the volume by controlling how many DSD converters are turned on." ( you can click the black text to read for yourself on Benchmark's site). NOPE. Maybe if we wanted a mere 64 level volume control this could work, at the large expense of resolution and linearity. The digital volume control of ESS chipsets is a real 32 bit solution. 6 bits? No, that will not cut it. For 10 years I have searched for the answer to this question. Unfortunately, I have a hard time seeing things right in front of my eyes, tending to pay more attention to the periphery. So, I began downloading every datasheet for every chip ever made by ESS to see if I could find any clue. And then, as if I had been skipping over the answer since 2014, there is was. The function diagram that had alluded me for so long. You can see it at the top of page, and I will repeat it again right here. I posted to Head-fi in the HQPlayer forum this very signal path, along with an explanation how this architecture is actually nothing special compared to what we already know about DSD-wide, and how DSD is processed for digital volume control on Cirrus, AKM chips as well as others. Here is the post I made on head-fi. I hope you find it enlightening. ESS again has been tight-lipped on their method of DSD conversion, but I DID happen to find a diagram from one of their datasheets that answers a LOT of questions.
This confirms that what you may have heard explained over the years by some really high-end, well respected engineers is not really correct, or at least they were obfuscating. The truth seems to be that the DSD path is not that different from what we have come to know about things like Sony's DSD-Wide, or how Cirrus, AKM and others covert DSD with DSP/Volume Control. The Digital Signal Path shown shows what I have suspected all along. DSD hits a FIR low pass filter, which creates a multi-bit signal. The signal is not 'decimated', and remains at the same rate as DSD input. (Decimation can mean more than one thing to me; in this case I say not decimated because no samples are removed, even if there is some redundancy.) The filter output can be just a few bits wide, but IF the volume control is used to attenuate, the output of the FIR filter is multiplied by a 32bit gain control, creating an even wider sample. Due to bandwidth restraints, we are probably not in a DSD unary code; we will be in binary. Next up DSD multibit intermediate is sent to the IIR filter that is user selectable at either 47khz, 50khz, 60khz, 70khz for further noise control before it is sent to the Delta-Sigma modulator. ( Not shown in diagram is the sample rate converter for jitter reduction, but this is likely before modulator as well, although I could be incorrect here). The modulator works as a multi-bit DSM. Considering the DAC itself has 64 unary/thermometer elements per channel, the modulator probably operates at 6 bits (binary). The 6 bit binary output of the Delta Sigma modulator at whatever oversampled speed is used (x128 or x256) is send to a logic system that converts the 6 bit binary into a 64 level UNARY code. ( note, that in unary code, 64 elements would actually mean 65 levels because all elements 'off' is zero. Likewise, 63 elements would be required for 64 (6-bit) levels) Furthermore, ESS uses here what they call the 'revolver' technique. (similar to DCS 'Ring' DAC). Levels are 'scrambled' since unary code is essentially multiple 1-bit signals that when added together equals the correct amplitude. It doesn't matter which element the logic 'shoot's the level to. The output elements are all equal, and this creates exceptional linearity, avoiding element mismatch. Now HERE is where we get conversion that is similar to the DSC1 (DSC2 DSC2.5 etc). DSC1 uses 32 equally matched output elements. Unlike the ESS which receives multi-bit delta sigma with 6 binary bit / 64 individual levels, the DSC1 receives 1 binary bit DSD. It uses shift registers to create 32 1-bit streams. All the same 1-bit stream, but each stream is offset by 1 clock cycle that, along with the output elements, form a FIR filter, which in this case, is a type of very simple moving average filter, a CIC filter. (If you ever see the output of a CIC filter on chart, you will see the null points that give it its comb filter name) Yes, the DSC 1 design has similarities to the ESS in that both use unary coded 1-bit conversion for the DAC. The ESS however is a true multi-bit signal. The DSC1, along with many other brands who do something almost exactly the same, is really just a 1-bit signal as far as actual information is concerned. But the method of conversion is very, very similar. The 1-bit way is less complex, and there are those who prefer as little complexity and as little DSP as possible. I am not here to say which is better, or if one party is right and the other is wrong. From my PERSONAL perspective, I would use the DSC2 with HQplayer any day. Thankfully I am lucky enough to have one of the very rare in the wild DSC2 converters. NEXT UP... FULL REVIEW OF THE SIGNALYST DSC2 OPEN SOURCE HARDWARE PURE DSD DAC with HQPLAYER4/12/2024 I have had the Signalyst DAC for sometime; actually I am enjoying Akiko Suwanai playing the Sibelius Violin Concerto in D minor op.47 in Native DSD64 (well, oversampled to DSD256 for the slightly long 32 tap (for a DSD Moving Average/CIC FIR) filter. I would go all the way to DSD512 but my PC cannot handle the conversion.
With the arrival of the E1DA Scaler, there is no longer an impedance mismatch between the output transformers of the DAC and the E1DA ADC, so its time for some 'real' measurements to see how it performs on paper. I can already tell you how it performs to the ear. It is magnificent. That is all that matters, but curiosity always gets the best of me. Besides, I want to see what the (hardware) filter's performance parameters actually are... approximate -3db cutoff and rolloff. I am really looking forward to this one! 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. This is lifted directly from the manual-- * VARIABLE ---- The output is volume controlled. (all formats) * FIXED ---- The output 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! eXCELLENT BREAKDOWNS OF FILTERSĀ in simple, easy to understand terms as they pertain to dsd8/26/2023 FIR Filter: 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.
IIR Filter: 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."
Summary:
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.
In summary:
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?
Simple Analogy: 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. )
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