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Andrew Allen Ballew

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    Euphonic Review

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  1. No matter what you may consider 'native' DSD, the output must be filtered. Whether this means to you DSD conversion is no longer 'native' is I suppose a matter of perspective. The single element analog filter for DSD is kind of a holy grail. It has been attempted, but never with good results, at least not that I am aware of. I have spoken at length with engineers about this.. the ones at iFi and T+A gave me the most time of day. So most native DSD converters use a filter with multiple elements. Does this make DSD any less native? I would say no. Because this is the only requirement for DSD conversion. A analog low pass filter. It is the the simplest way to convert a digital signal that I know of. Furthermore, it is extremely linear, as all systems use some kind of random/dynamic element matching. Yes, this kind of filter earlier in the signal chain would create a multi-bit signal that doesn't necessarily resemble the original. But it doesn't come early. It is the final action in the chain. The shift registers send the time-offset signals to the switches, which are also the taps! Therefore the digital and the analog become one. The current of each switch/tap are accumulated into the final filtered signal. Almost always then followed by another basic RC or similar analog filter to further shape the remaining ultrasonic noise as the designer/engineer deems fit to best match the analog stage. Finally, DSD speed really has nothing to do with it. All native DSD, from 64fs to 'insert insanely ridiculous speed here', will be converted by a filter at some point in the chain; native DSD doing so at the chain endpoint.
  2. even with my limited experience with HQPlayer, it has I believe confirmed what I have postulated for a long time. That the onboard DSP in DACs is what holds many of them back. The onboard filters, yes, you can hear some difference, but honestly with most DACs they all sound so similar its not worth the trouble. Not the case with HQPlayer. The filters and modulators and their sonic characteristics are much more prominent. So its not too surprising dongles, with virtually all their budget spent on the digital hardware and whatever is left over on analog stage, and after that any actual programming of the say, USB and filter stages, see a major jump in quality when HQplayer is added, especially if you can bypass any hardware DSP. I have already said it a couple times. Of all the money I have spent to improve my head-fi (and hi-fi) this is some of the best well spent.
  3. Yeah, I love the iDSD PRO. But, I also love its little brother, the ZEN DAC (either signature or V2). You give up a ton of features, but the raw performance on test bench equals or exceeds the iDSD PRO! The ZEN series used a lot of what was learned from development of the iDSD PRO. No, it still doesn't SOUND as good, and the Headphone Amp is for sure superior on the PRO, (although the ZEN CAN is pretty darn outstanding even if you didn't know the price. ) I have auditioned a lot of equipment and have a backlog of reviews for my blog called EuphonicReview, but the DAC or DAC/amps I have kept for myself because they sound great and match my personal 'philosophy' on what makes good sound, whatever THAT is worth, lol, are as follows. These are all keepers IMO, SMSL D300 with ROHM chip. Especially if you use ASIO and can say on firmware 1.0. There have been major problems it seems with upgraded firmware that adds DoP support. RME ADI-2 PRO FS R BLACK EDITION AD/DA. Not everyone needs the ADC, so for most the DAC only version (AKM chip if possible) is an amazing DAC. Using volume/modulator bypass mode with DSD makes a pretty stunning sound with HQPlayer. Note that the DSD filters on the updated AK4493 version are not correct. They are not 50khz/150khz at DSD64. They are actually 39khz/76khz, I believe. I use the ADC, however to record in DSD. I once had a TASCAM standalone DSD recorder, and I sold it, and now they are almost impossible to find! But I use Vinyl Studio to record in DSD with the RME. To transfer my vinyl, and yeah, it can be used for other things too. HQPlayer PRO would be nice to have, but I am afraid to even ask the price lol iFi ZEN DAC V2 if you want decent headphone amp attached to excellent DAC, or ZEN Signature for all DAC. iFi iDSD PRO for the sound. It just sounds amazing. Its onboard DSD512/DSD1024 oversampler is also very, very good. I have not done any comparos yet with HQPlayer. But measurements show it gives up no performance to PCM, such as THD, Noise, etc. Except for perhaps a little dip at DSD1024. Which MAY be related to the logic itself, and how well it 'keeps up'. The faster we go with these DACs, the better and more solid their hardware needs to be. Signalyst DSC DAC. 'Nuff said lol. Perfect match with HQPlayer.
  4. Yes, they double with bitspeed. The 13khz at DSD64 is too low, but the filter you mention IMO is also the best choice of DSD filter on the SMSL D300. Which makes the cutoff at 52khz using filter marked 13khz at DSD64. ( I also like DSD256 with 'filter 2' which has FC at 104khz, even though there is an increase of harmonic distortion when filtering at 104kz rather than 52khz) TI as far as I have been able to learn use unequal weights. There are 64+ equally weighted unary code taps/switches used when say, the DSD1793 chip is used in PCM conversion mode. Top 6 bit (64 levels) are converted in unary code, bottom 18 bits in a 24bit word are converted by the DSM. They all combine together and use Dynamic Element Matching for linearity. DSD uses a 8 bit shift register, same unary coded switches. But the 8 taps in the filter use various weightings or numbers or switches to control the analog FIR filter parameters, but all the switches are always used and Dynamic Element Matching still applied. Seems to be the more complicated way to do it. Thank you for the info on the ROHM architecture! Here is the ROHM via the DMSL D300 at DSD64 with filters 2 and 3 (26khz) (52khz)
  5. Today I have been experimenting with the CIC Cascade integrator when oversampling DSD64/128 to DSD256. Here is where my knowledge really starts to get in muddy waters. My understanding is CIC is essentially a FIR Moving Average Filter. But, then again, it isn't. The part that isn't the same.. is it explainable in layman terms? My current understanding is it IS a moving average filter but with built in decimator or interpolator. Today I have been doing most of my listening via the S.MS.L D300 (ROHM segment DAC with native DSD bypass mode). The filters look like moving average FIR as is common with DSD DAC's. But I know the DSC DAC, for instance, primary filter is called a CIC filter. They look the same to me on the Analyzer.. again, what is the major difference? Any answer is very appreciated. Attached below is the S.M.S.L DSD64 13khz FC profile. By just looking, I could tell you its a moving average FIR, but NOT specifically a CIC.. what should one look for?
  6. lol as we say in the southern USA, even a blind squirrel finds an acorn from time to time..
  7. totally understood i was taking the 'bulk of the evidence' present to come to the conclusion. I get that different modulators could lead to different results, for sure. It is was only the 6db loss of dynamic range, it would be suggestive, but would need more evidence to come to a reasonable conclusion. The modulator used in my first example was the onboard FPGA based modulator in the iDSD PRO. Actually I have a suite of test files made with HQPlayer Pro too, but I can't tell you if they are 50% modulation = 0dbfs. I received them very generously from the person from which I acquired my DSC DAC. The DSD measurements done with the iDSD PRO come from its onboard oversampler. With the purposeful lift to equalize DSD and PCM volume level in the analog output stage, and that that difference in noise level is right at -6db, it just seemed like a lot of circumstantial evidence to say Filter 1 or Filter 2. Filter 1 seems unlikely, though, based on other measurements. But, yes, it is a circumstantial case indeed. And could be wrong. :) I agree with Jussi that it was a disappointment when they took the user choice of filter FC away. The old Micro from 2014 as you mentioned could select various of the DSD filters, and there was notable volume change with each. When I asked iFi about this, there only answer was that the filter was at 80khz with a servo based system to match DSD and PCM levels. Of course, they were not talking about the FIR filter I was asking about. That filter is the secondary analog RC filter than follows. When I asked Thorsten directly after he had left iFi, he didn't recall which FIR filter they chose in the end. In the end, it really isn't a big deal lol and I am writing so many words about it :) :) What IS a BIG DEAL is this... I am a classically trained concert pianist by trade (which means mostly teaching these days haha) BUT I am listening to the May 2019 recordings of pianist Marc-Andre Hamelin playing Liszt opera transcriptions using HQPlayer oversampling the 192khz original recording to DSD256, and it is the most realistic reproduction of piano I believe I have yet ever heard on any system of mine. Incredible, and worth every penny of the software. My PC is barely keeping up; its a slightly older HP all-in-one i5 8400T at 1.7ghz--- running at over 90 percent CPU and peaking over 3Ghz clock speed. I will need to upgrade to something with a bit more punch fairly soon. Right now my filter is poly-sinc-gauss and modulator is DSD7 256+FS. Completely satisfied with this combo at the moment. Some of the best spent money ever on my Hi-Fi.
  8. yes. i am simply pointing out the filter behavior of the FIR DSD filter that iFi is using. Jussi said he wasn't sure which one iFi had chosen since they disallowed user control of the DSD filter FC. On the iDSD PRO... the is a servo control that equalizes PCM 0db and DSD 0db. The voltage is the same for each output. It gives a clue to WHICH FIR filter iFi is using, because even WITH the voltage gain via the servo on the DSD signal, the Noise floor is exactly 6 decibels worse. This is what would be expected to happen with the DSD Filter conversion at -6db, which is exactly what FIR filter number 2 on the DSD1793 does. The servo raises the noise floor on DSD by 6db, maintaining a 6db lower dynamic range as it raises noise floor and peak voltage The only other FIR filter close to a output of -6db (as compared to 0dBFS PCM according to datasheet) is filter 1, which is 6.6db lower playback than PCM. But its filter cut is over 180khz, and doesn't seem to match the measured results. I was simply pointing out iFi has likely chosen Filter 2 of the DSD1793 DAC. Filter 2 seems to match what the spectrum shows... more importantly, Filter 2 plays back exactly 6db less than PCM, just as you would expect 'by the book'.
  9. here is DSD on Spectrum Analyzer, not via the iDSD PRO, rather via ZEN DAC V2.. Appears to have a strong rolloff right around 77khz (Filter 2). Again, i could be very wrong, but I don't think the secondary 80khz RC analog filter could itself account for the all the rolloff seen. The DSD128 spectrum seems to confirm as well.
  10. It is number 2 based on my measurements of the 'post' Micro series when the filter choice was fixed. The iFi iDSD PRO shows right a 6db loss in Dynamic range vs PCM when in DSD mode. The iDSD PRO uses a servo to equalize the volume with PCM. I verified equal voltage on the output for both PCM and DSD. yet there was still a 6db loss in Dynamic Range with DSD I suppose it could be 1, but that is off the top of my head FIR filter at 180khz plus. I don't think they would choose that one for noise management reasons. It appears that way in the measurements as well... #2 FIR filter cuts volume exactly 6db, and is followed by a RC analog filter with FC at 80khz. It still COULD be FIR filter 1, but I don't think so. This first set of rightmark measurements are from the headphone out and done with RME ADI-2 PRO with an impedance loads at 32ohm. They are NOT representative of the performance the iDSD PRO is capable of; they were done early early on as I learning to effectively measure hardware. On the left PCM, right DSD 512 from iDSD PRO Engine. almost exactly 6db loss that would be expected with filter 2. I also asked Thor, he didn't seem to recall which one they chose.
  11. Finally.. officially onboard with HQPlayer. Did some experimenting today vs the MUSE system in Roon, which is no slouch, and came away impressed enough to pay up the not insignificant dough. I will need a PC upgrade soon.. as my chosen config, I am running over 90 percent CPU on my about 4 year old HP All-in-One with i5-8400T with 1,7ghz base speed. 12 Gig RAM Win 11. That is just connected locally. I have yet to attempt a connect via NAA through my iFi ZEN Stream over to the iDSD PRO. Here at the desktop, I am greatly enjoying my SMSL D300 with the new ROHM DAC that is 100 percent analog DSD conversion. RME ADI-2 PRO FS R Black Edition is waiting its turn. Finally they will All GIVE way to my Signalyst DSC2 DSD DAC build sitting in the corner waiting patiently haha 😁
  12. On linearity. Yes, 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 transformer 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. However, I still have found some interesting things. 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, it is virtually perfect. deviations are LESS than 0.1db. Much less. At -110db, we are STILL at <0.1db deviation. This beats out $30,000 plus DACs. Like the one if you rearrange the letters a bit in DSC haha. At -115db, we finally hit some loss of resolution, with a deviation of an entire -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 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 18picosecond 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.
  13. Being a transformer based solution, this converter is NOT going to wow anyone when it comes to SINAD. But here is one of those cases where SINAD just does NOT tell the whole story. First of all, not all harmonic distortion is bad. I see it from the perspective of a professional musician... these are overtones. They follow the musical overtone series. (As opposed to inharmonic or non-harmonic distortion, such as clipping). For instance, the second harmonic is the octave. In the right balance, I bet many of you could not hear this distortion at all. Even the third harmonic, (one of those 'bad' odd order harmonics) is in the musical overtone series a perfect fifth. From both the fundamental and the second harmonic. (Yes I know it is no longer a fifth compared to the fundamental, but those of you who know what I mean, you well, know what I mean!) It is called perfect for a reason. When tuning there will be no 'beats'. The mathematical ratio is perfect. Elevated harmonic distortion is what makes tubes sound like they do, and the vintage transformers of those tube amps are part of the equation, too. GRANTED, there is a delicate balance to be found here. Even if you like harmonic distortion, there is definitely too much of a 'good thing' at some point. It can also make for difficult system matching. You have to be careful with your components and how they interact. One thing I CAN say about this DSC2 design, is it doesn't have NEARLY the amount of harmonic distortion as some other transformer DAC designs, including some DSD ones, that cost 7 or 8 times as much. And the DSC2 has a noise floor and dynamic range that is as good or better than many of the best designs on the market today. Not gonna name any names, but again, there are some DACs out there purporting to do similar things at a much much higher price that don't come close to the measured performance of the DSC2. It is my personal opinion, that once you get over 90db SINAD, you are okay, 100db SINAD is better for sure. The DSC2 in the area where our hearing is most sensitive, say, 1khz to 6khz, it gets well over the 90db SINAD mark, and once I get a better PSU, I think it can be coaxed closer to that 100db SINAD mark. At the same time I will say the balance of sound here is very, very good. It isn't so warm as to be too 'tubey' sounding, nor is it dry, thin and boring. Not by any means! Nor it is 'rolled-off' in its sound. It has excellent treble extension and detail, without going so far as to be too sibilant. No, it will not cover up bad recordings. It has plenty of resolving power to display what is really there. Bad for the poor recordings, and truly great for the good ones.
  14. I posted this over in my SMSL D300 DAC thread, but it really belongs here. First, I am not using the cleanest power source yet... iFi X Power 9 volt yet to arrive so I am on a really cheap SMPS of which I do not know the provenance or how noisy it is. Also, the E1DA Cosmos ADC has a very low input impedance, so it isn't a great match for the DSC DAC output transformer's high output impedance, which can make getting enough 'juice' out of the DAC to the ADC difficult. Both of those things admitted, Here is what I did indeed get today attempting a J-test measurement in Multitone. Very, very impressive especially given the less than optimal situation. Cleaner, much much cleaner than some other Native DSD DACs we are very aware of that cost close to 10 Grand ;) The noise floor (note, NOT MUTED) is better than -120db (A)!!!!! well... on to the jitter graph....
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