mav52 Posted July 22, 2018 Share Posted July 22, 2018 Interesting unit, which has over 1 million TAPS. https://chordelectronics.co.uk/product/hugo-mscaler/ https://darko.audio/2018/07/chord-electronics-announce-hugo-m-scaler/ So why does it matter how many TAPS ( FIR's) you have and why is it mo better. https://dspguru.com/dsp/faqs/fir/basics/ I mean when I listen to a Halo LV 3 in NOS or the PS Audio Direct Stream DAC they sound pretty good. Not sure how many of these TAPS ( FIRS) they have. The Truth Is Out There Link to comment
Popular Post rayl1234 Posted July 23, 2018 Popular Post Share Posted July 23, 2018 Someone posted a video of the presentation from CanJam where Rob Watts explains the importance of transients being reconstructed accurately: But as a (still, despite having paid 2x for the tech) happy owner of the Blu2 with M-Scaler, I created an alternate illustration to explain to my friends what it is all about. Below is an illustration of a perfect sine wave sampled and then reconstructed with various interpolation methods. I included as first illustrations zero order hold and connecting the dots, before moving onto different FIR interpolation filter lengths. I used numbers that match 1/1000 of 44.1kHz (to speed up calculation) and also included illustrations matching DAVE so scaled (164) and MScaler (1016). These examples are more extreme than Rob's bec they were selected for illustration. His point (my interpretation) is that for less extreme examples, the misalignment due to different rise times from imperfect interpolation will still cause audible artifacts. (I've always called that "upsampling+interpolation" or even just simply "upsampling", but folks here on CA have corrected me by saying that isn't the audiophile definition... It has been suggested that I mean "resampling"... but I stick to my Signals & Systems course terminology and did not change it.) I am also including a link to the Octave(free) or Matlab code generating the samples: https://pastebin.com/F7709pkG Hope this helps. semente, look&listen, buonassi and 3 others 5 1 Link to comment
mansr Posted July 23, 2018 Share Posted July 23, 2018 Rob Watts is technically correct in that a longer filter gives a more accurate reconstruction. However, I'm certain that nobody can hear the difference between 100k and 1M taps. Conferring such great importance on the filter length falls, in my opinion, in the same category as talking about skin effect at audio frequencies: real phenomena of limited or no relevance to audio applications. esldude 1 Link to comment
mav52 Posted July 23, 2018 Author Share Posted July 23, 2018 2 hours ago, mansr said: Rob Watts is technically correct in that a longer filter gives a more accurate reconstruction. However, I'm certain that nobody can hear the difference between 100k and 1M taps. Conferring such great importance on the filter length falls, in my opinion, in the same category as talking about skin effect at audio frequencies: real phenomena of limited or no relevance to audio applications. If taps are so "important to chord, I wonder why the other DAC manufacturers don't list their TAP quantity. OR is this Marketing at it best. The Truth Is Out There Link to comment
ElviaCaprice Posted July 23, 2018 Share Posted July 23, 2018 Proof is in the pudding. Tested by others and found to make a big difference, number of taps. So big, that it has been recommended by those using the Blu mscaler and now the Hugo mscaler, to be a far more important SQ upgrade over any server tweaks upstream. Basically a must for those with the coin and in the Chord system build. Anyone else that says otherwise is obviously talking from back seat science and not from actual listening. Thus disregard them. Albrecht 1 (JRiver) Jetway barebones NUC (mod 3 sCLK-EX, Cybershaft OP 14) (PH SR7) => mini pcie adapter to PCIe 1X => tXUSBexp PCIe card (mod sCLK-EX) (PH SR7) => (USPCB) Chord DAVE => Omega Super 8XRS/REL t5i (All powered thru Topaz Isolation Transformer) Link to comment
Popular Post mansr Posted July 23, 2018 Popular Post Share Posted July 23, 2018 54 minutes ago, mav52 said: If taps are so "important to chord, I wonder why the other DAC manufacturers don't list their TAP quantity. OR is this Marketing at it best. All manufacturers seem to have some pet feature they emphasise the importance of above all else. For Chord, it's absurdly long linear phase filters. For Ayre it's rather shorter minimum phase filters. For Schiit it's the avoidance of sigma-delta modulators. There are many other examples. The interesting thing is that not only do these approaches differ in the weights they give to different aspects, they are in direct conflict. If two men say they're Jesus, one of them must be wrong. As I said already, Rob Watts is technically correct, but I still think he's overdone it to an extent that is hard to justify. esldude, semente, Jud and 3 others 4 2 Link to comment
Popular Post ecwl Posted July 23, 2018 Popular Post Share Posted July 23, 2018 11 minutes ago, mansr said: As I said already, Rob Watts is technically correct, but I still think he's overdone it to an extent that is hard to justify. I have listened to DAVE with and without Blu2 M-Scaler for the past 14 months. I can tell you the sonic difference is not subtle. I completely agree with you that conceptually it seems hard to justify 1 million taps vs 164,000 taps. But it’s definitely something you should try to listen to some day. ElviaCaprice, The Computer Audiophile and beautiful music 2 1 Link to comment
rayl1234 Posted July 23, 2018 Share Posted July 23, 2018 2 hours ago, mav52 said: If taps are so "important to chord, I wonder why the other DAC manufacturers don't list their TAP quantity. OR is this Marketing at it best. Different designers/listeners have different areas of focus. What is true is that for the most accurate reconstruction from a digital sampling perspective, the closer to the infinite sinc, the better. However, some folks prefer the sound of certain other filter approaches, some people prefer to emphasize the noise shapers, R2R, voltage regulation, etc. Chord's direction has been towards infinite sinc and with pulse arrays -- part of the synergy of the latter is they believe they can build medium power (like in the tens of watts per channel) "digital amplifiers" that can be fed directly from a 16FS digital signal (or perhaps better stated -- high power DACs minus the primary reconstruction filter). Folks who do software upsampling to output to a DSD DAC are similar but, of course, DSD is 1 bit whereas the Chord's pulse arrays are 4 or 5 bits (I forget) to speed up the error factor correction delay.... beautiful music 1 Link to comment
Popular Post Miska Posted July 24, 2018 Popular Post Share Posted July 24, 2018 On 7/23/2018 at 4:41 PM, mansr said: Rob Watts is technically correct in that a longer filter gives a more accurate reconstruction. However, I'm certain that nobody can hear the difference between 100k and 1M taps. Conferring such great importance on the filter length falls, in my opinion, in the same category as talking about skin effect at audio frequencies: real phenomena of limited or no relevance to audio applications. Funny thing is that computers doing stated upsampling to 16x with 1M taps doesn't even break a sweat. It is so easy to pull it to completely different figures with computer, if one wants.... Now I would like to see Rob Watts (Chord) and Bob Stuart (Meridian/MQA) arguing about this filter length stuff, they are in completely opposite camps... Both of them fixing timing problems... I personally prefer to sit somewhere between and find other aspects of filter design more interesting than number of taps, which is as meaningless figure talking about filters as MIPS is when talking about computer performance. bhobba, semente, The Computer Audiophile and 4 others 6 1 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
Popular Post Miska Posted July 24, 2018 Popular Post Share Posted July 24, 2018 20 hours ago, rayl1234 said: Folks who do software upsampling to output to a DSD DAC are similar but, of course, DSD is 1 bit whereas the Chord's pulse arrays are 4 or 5 bits (I forget) to speed up the error factor correction delay.... Which doesn't really make practical difference compared to the modulator design. Chord uses so simplistic modulator that the noise floor looks more like something between DSD128 and DSD256 even though it runs at 100 MHz. I'm more into how many clock cycles you have per output sample. Current CPU's have 186 clock cycles per DSD512 output sample, because they run at 4.2 GHz. Whitigir, jabbr, auricgoldfinger and 2 others 2 1 2 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
rayl1234 Posted July 24, 2018 Share Posted July 24, 2018 59 minutes ago, Miska said: Funny thing is that computers doing stated upsampling to 16x with 1M taps doesn't even break a sweat. It is so easy to pull it to completely different figures with computer, if one wants.... Now I would like to see Rob Watts (Chord) and Bob Stuart (Meridian/MQA) arguing about this filter length stuff, they are in completely opposite camps... Both of them fixing timing problems... I personally prefer to sit somewhere between and find other aspects of filter design more interesting than number of taps, which is as meaningless figure talking about filters as MIPS is when talking about computer performance. Sometimes we pay for the packaging... E.g., Build me an upsampler with a USB in/out so I can play Tidal through it, and I am will gladly try it. Though I could write equivalent code (since he disclosed that he ended up at equivalent of sinc, so the calculations are public domain) to upsample my stored .wav/.flac files, that is < 5% of my listening, so I need a solution that works with streaming.... so, an early adopter I became... The Computer Audiophile 1 Link to comment
ecwl Posted July 24, 2018 Share Posted July 24, 2018 2 hours ago, Miska said: Which doesn't really make practical difference compared to the modulator design. Chord uses so simplistic modulator that the noise floor looks more like something between DSD128 and DSD256 even though it runs at 100 MHz. That’s an excellent point and thanks for sharing your Mojo measurements in the past. That said, I think in terms of noise shapers: Mojo/Hugo: 5th order noise shaperHugo 2: 11th order noise shaperHugo TT 2: 12th order noise shaperDAVE: 17th order noise shaper so maybe it’s better now? ecwl 1 Link to comment
The Computer Audiophile Posted July 24, 2018 Share Posted July 24, 2018 Hi Guys - Thanks so much for the discussion thus far. It's really cool to see people offer opinions about sound quality after listening and for others to offer their expertise based on what they know about the technology. All information is welcomed. Since there's no free lunch, I'd love to know about the Pros and Cons to the Chord approach. asdf1000 1 Founder of Audiophile Style | My Audio Systems Link to comment
rayl1234 Posted July 24, 2018 Share Posted July 24, 2018 I will say, after experiencing the Chord stack at RMAF 2017, my immediate reaction was to set about selling my DAC stack and acquiring a Chord Blu2/DAVE one and am happy despite paying the early adopter tax. To me, even under show conditions, it was convincing enough to set a plan into motion. For me, the big con, besides $ (much reduced now with standalone M Scaler), was conceding that I am OK with focusing energy on a PCM-centric system w/o MQA (and w/DSD as a second class citizen). I have not personally experienced MSB -- that's the big comparison unknown for me. (Not that I would spring for a Select 2 given the price, but a Reference would be within the ballpark.) Link to comment
Popular Post ecwl Posted July 24, 2018 Popular Post Share Posted July 24, 2018 3 hours ago, The Computer Audiophile said: Since there's no free lunch, I'd love to know about the Pros and Cons to the Chord approach. Chord DAC designer Rob Watts posts lots on Head-Fi detailing his approach. And we are fortunate to have great people at CA forums like Barrows and Miska to offer their perspectives. Their perspectives are often quite technical so I admit I don't always understand everything. My undergrad degree is in Math and Chemistry. But I do have to simplify scientific knowledge and understand it in my mind and explain it to others for work (even though the simplified version may not be an accurate representation of reality). Anyway, this is how I would simplify the Rob Watts approach/explanation/philosophy. A DAC is actually made up of two parts, one part that converts the original digital samples/signals into a different sampling rate/bit depth (upsampling/filtering) and another part that's truly the digital-analog convertor. There are different philosophies on how to do the upsampling/filtering and there are primarily 3-4 approaches to the true DAC part. For the upsampling/filtering, let's take two extremes: Some people feel that no-oversampling (NOS) is the best approach so if you recorded in PCM 44kHz 16-bit, it's best to play the music back using R2R DACs because you're playing the original digital signal. If you recorded in DSD64, it's best to use a DSD DAC to play back the original signal. Rob Watts approach is the other philosophical extreme. He feels that even though Nyquist theorem says the digital samples from PCM has enough information to represent the original analog waveform, the goal of the DAC is not to play back the original digital samples like an NOS R2R DAC, but to generate the original analog waveform as close as possible. In order to be able to do that, you need to upsample the signal say 44kHz 16-bit to a higher frequency sample say 705.4kHz so that you're filling in the gaps between the original samples to regenerate the original analog waveform. And you need lots of taps to get close to the sync function. You can see the difference in the waveform that is generated from @rayl1234's beautiful diagrams. The cons of this approach of lots of real taps is that you need a lot more computing power (and memory management because you also need more and more signal data, e.g. M-Scaler uses 1.26s of digital samples at a time to generate the accurate samples). You would also cause some audio delays of 0.63s. An intermediate approach as pointed out is to use a different upsampling algorithm/filter. But whatever filter is chosen, regardless of tap length, would have specific impacts in the time domain and the frequency domain. Some people feel that as long as the filter looks good in the frequency domain, we are good. Rob Watts claims that he has tried a variety of filters that looks great in the frequency domain but he can hear the difference. He feels it is important to use as many taps as possible to mimick the sync filter to ensure the filter is accurate in the time domain and frequency domain to ensure the best sound. I suspect MQA is a compromise filter between the time domain and the frequency domain but I admit I don't fully understand MQA (other than the lossy compression part). Another aspect that's important to remember is that many non-Chord filters with long tap length algorithms are actually the same filter applied multiple times. Meaning that it could be a 100 tap length filter used to upsample from 44kHz to 88kHz and then the same filter applied again to 176kHz then to 352kHz then to 705.4kHz. The problem with that approach according to Rob Watts is that you are only using 0.00126s of digital samples to get to 705.4kHz so the analog waveform that you end up generating are still not going to be close to the original sync function and would have compromises in the time and frequency domain. Of course, 16x (16fs) upsampling is only the first step as there has to be an additional step to upsample to a higher frequency and then to a lower bit depth with noise shaping, since most DACs actually operate at much higher frequencies. I would post next on Chord Pulse Array DAC. But I'd welcome people who actually understands the technical sides better to correct my simplifications and where I get things wrong. beautiful music, mozes, Currawong and 2 others 1 1 3 Link to comment
Popular Post ecwl Posted July 24, 2018 Popular Post Share Posted July 24, 2018 3 hours ago, The Computer Audiophile said: Since there's no free lunch, I'd love to know about the Pros and Cons to the Chord approach. Now on to what I consider the "true" DAC part. This I even understand less so if I'm wrong, please correct me. If you want a simple approach, there are two choices, R2R DACs and DSD DACs. DSD DACs are pulse-width modulation so single-bit at high frequencies, usually 2.8MHz or multiples, up to 44.8MHz (Emm Labs) modulating at 1-bit (on or off). R2R ladder DACs basically have different resistors to generate varying signal levels running at anywhere between 44.1kHz to 384kHz. The problem with R2R ladder DACs is that it is very hard to match all the resistors to the same level. That's why we often see R2R DACs with multiple DAC chip sets to average out the signal for better matching. Another problem is that if the DAC can only run at a maximum of 384kHz, you're never going to get a waveform that gets close to the original analog waveform. There is another problem that is universal to DSD and R2R DAC designs that Rob Watts likes to talk about. Even though mathematically, DSD designs seem simple, the reality is that 1's and 0's are not the same in DSD DACs. Meaning if you imagine you have the following DSD signal 1010101010101010 vs 1100110011001100. You would think that at 2.8MHz or 44.8MHz, they should give the same sound. The problem is that switching between on to off and off to on would inevitably generate different amounts of noise in the system. That's why the signal you get in DSD for two theoretically mathematically similar signals would contain differing amounts of noise. What this results in are a few things: jitter (because the noise messes with the clock), signal-dependent noise, and noise floor modulation. It is noise floor modulation that Rob Watts liked to focus on in the past when discussing his pulse array DACs and it is a characteristic of DACs that most review measurements do not take into account of. Conceptually, what happens is that when the DAC is producing a 1kHz signal at -60dB, the noise floor of the DAC is say -140dB but when the DAC is producing a 1kHz signal at -30dB, the noise floor of the DAC is now -120dB. Rob Watts believes the noise floor modulation is what makes digital sound harsh and not analog/vinyl sounding. Unfortunately, sometimes noise floor modulation actually makes music sound more exciting and since most of our DACs have noise floor modulation, I would argue that we are used to listening to DAC noise floor modulation sound more than live sound and I've seen many people prefer it. Of course, many would say we can't hear -120dB vs -140dB so it doesn't matter but I have to say comparing even older Pulse Array DACs to other DACs, I think it does. Moving on, Rob Watts also feels DSD can have another problem, more so at 2.8MHz and less so at 44.8MHz. If you imagine music suddenly starts with a drum or cymbal strike or guitar pluck, in analog waveform and PCM, the signal jumps up slightly and continues. But if you are trying to get it in DSD, you would have to content with a bunch of 1's and 0's until they modulate to a point where it resembles that initial increase in amplitude of that note/strike/pluck. So it is harder for DSD to capture small amplitude transients accurately as it needs to go through the noise shaper. This is the reason why DSD generally has a softer sound (which may be pleasing to people). Of course, if you run DSD at a much higher frequency, say 44.8MHz, you would run less into these issues. This is the reason why Rob Watts prefers the thermometer code approach which is basically having multiple elements of PWM/DSD DACs running parallel to solve the DSD timing issue. He prefers using discrete elements, like dCS, because he thinks there is simply too much noise that would be present on DAC chips (which he used to design). This would technically solve the small amplitude transient problem of DSD and the level matching problems of R2R DACs. But more importantly, running multiple elements (5-bit 2.8MHz or 5.6MHz like dCS), still would not solve the noise floor modulation and jitter problem completely. Because the amount of switching activity would still be signal dependent. As a result, Rob Watts developed Pulse Array DAC using thermometer code with constant switching activity. Because the elements are constantly switching, the switching no longer is signal-dependent so it significantly reduces signal-dependent noise, noise floor modulation and makes the DAC design much more jitter-immune. At least this is my understanding of what Rob Watts is saying. I'm sure I got some things wrong and would appreciate people to correct me. Ultimately, having read a lot of reviews and forum posts, what I find is that people like the DACs that they like. What is theoretically (according to Rob Watts) a more accurate representation of the analog waveform, may not actually be more pleasing for individual audiophiles with their personal sonic preferences and audio system synergy also matters a lot. My philosophy is always that if you listen to a DAC and you like the sound, kudos to you. You're spending your money. You should pay for what you like. ecwl, mozes, Currawong and 2 others 1 1 2 1 Link to comment
Miska Posted July 24, 2018 Share Posted July 24, 2018 4 hours ago, ecwl said: That’s an excellent point and thanks for sharing your Mojo measurements in the past. That said, I think in terms of noise shapers: Mojo/Hugo: 5th order noise shaperHugo 2: 11th order noise shaperHugo TT 2: 12th order noise shaperDAVE: 17th order noise shaper so maybe it’s better now? Order is just as useless number to describe a modulator as is number of taps to describe a filter. Higher order modulator makes the increasing noise slope steeper, which in turn puts more demand on the analog filter. So just blindly increasing number of taps or modulator order is not necessarily best solution... Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
Miska Posted July 24, 2018 Share Posted July 24, 2018 6 hours ago, rayl1234 said: Sometimes we pay for the packaging... E.g., Build me an upsampler with a USB in/out so I can play Tidal through it, and I am will gladly try it. That's what I've been doing with software upsampling to DSD512 (of course one can do any PCM rate too). Playing Tidal doesn't require USB though. auricgoldfinger, ecwl and jventer 1 2 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
Miska Posted July 24, 2018 Share Posted July 24, 2018 16 minutes ago, ecwl said: This is the reason why Rob Watts prefers the thermometer code approach which is basically having multiple elements of PWM/DSD DACs running parallel to solve the DSD timing issue. He prefers using discrete elements, like dCS, because he thinks there is simply too much noise that would be present on DAC chips (which he used to design). This would technically solve the small amplitude transient problem of DSD and the level matching problems of R2R DACs. But more importantly, running multiple elements (5-bit 2.8MHz or 5.6MHz like dCS), still would not solve the noise floor modulation and jitter problem completely. Because the amount of switching activity would still be signal dependent. As a result, Rob Watts developed Pulse Array DAC using thermometer code with constant switching activity. Because the elements are constantly switching, the switching no longer is signal-dependent so it significantly reduces signal-dependent noise, noise floor modulation and makes the DAC design much more jitter-immune. And the DSC1 DAC design I published for playing up to DSD512 (24.576 MHz) uses 32 discrete elements with constant switching... dCS has 24 and ESS Sabre has 64. Anyway, there's really no timing problem and reason for using multiple elements is completely different from any timing stuff. ecwl 1 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
ecwl Posted July 24, 2018 Share Posted July 24, 2018 11 minutes ago, Miska said: And the DSC1 DAC design I published for playing up to DSD512 (24.576 MHz) uses 32 discrete elements with constant switching... dCS has 24 and ESS Sabre has 64. Anyway, there's really no timing problem and reason for using multiple elements is completely different from any timing stuff. I think you’re the expert on the matter. You’re right, my understanding for multiple elements is to avoid slight mismatch between each element. But I am not sure if Rob Watts description of constant switching is the same as yours. I was under the impression that in dCS, ESS, and presumably DSC1, you’re using thermometer code or something similar to do DEM (dynamically select elements) so that you’re not switching the same elements each time to avoid element mismatch. But you would still get signal dependent switching. Whereas Rob Watts describes matching elements that switch so that the same number of elements would always switch at the same time all the time so that you always get the same amount of noise from switching. But I might have understood it wrong. ecwl 1 Link to comment
Miska Posted July 24, 2018 Share Posted July 24, 2018 45 minutes ago, ecwl said: I think you’re the expert on the matter. You’re right, my understanding for multiple elements is to avoid slight mismatch between each element. But I am not sure if Rob Watts description of constant switching is the same as yours. I was under the impression that in dCS, ESS, and presumably DSC1, you’re using thermometer code or something similar to do DEM (dynamically select elements) so that you’re not switching the same elements each time to avoid element mismatch. But you would still get signal dependent switching. Whereas Rob Watts describes matching elements that switch so that the same number of elements would always switch at the same time all the time so that you always get the same amount of noise from switching. But I might have understood it wrong. Multiple elements are used for other reasons, and DEM is used to decorrelate mismatch between the elements. Some DEMs may not always switch all elements, but DSC1 does switch all elements every time (although it is easy to modify not to). ecwl 1 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
ecwl Posted July 24, 2018 Share Posted July 24, 2018 3 minutes ago, Miska said: Some DEMs may not always switch all elements, but DSC1 does switch all elements every time (although it is easy to modify not to). Very neat. I’ve always wanted to listen to HQPlayer playing through the DSC1. And you’ve finally inspired me to understand how this constant switching Rob Watts talks about works. And it made me realize even though Mojo/Hugo are 4-element DACs, they are really functioning like a 1-bit/element DSD DAC at 104MHz. Although I’m guessing this is how Rob Watts does it as he was never explicit about it. Basically, I think he uses the elements to represent either 1 or 3 and nothing else That means when turned on, lets say we start with 0001 or 0010 or 0100 or 1000 And if the next signal is also 1 then he switches 2 elements to still stay at a sum of 1 e.g. 0001 to 1000 or 0100 or 0010 And if the next signal is 3 then he switches 2 elements to get to 3 e.g. 0001 to 1101 or 1011 or 0111 And if the next signal is 3 then he switches 2 elements to stay at 3 e.g. 0111 to 1011 or 1101 or 1110 So the DAC modulates between 1’s and 3’s (equivalent to 0’s and 1’s) to generate the signal. But then the DAC is constantly only switching 2 elements all the time. I have to admit, I don’t fully understand how DSC1 works as I don’t fully understand circuit diagrams since I never took electrical engineering. But it’s impressive how you manage to do something similar with 32-elements and switching all of them at the same time. ecwl 1 Link to comment
Miska Posted July 24, 2018 Share Posted July 24, 2018 Regardless how many values you want to represent, this is generally nice property of thermometer code (scrambled unary coding) used with SDM; you can represent the same value in many ways. Because all bit positions have same weight. This is something that is not possible with PCM/R2R. In addition, with PCM/R2R weight of the element error increases exponentially towards MSB. Modern DAC technology allows a lot of flexibility on how one does the conversion, from DSP to the actual conversion stage. ecwl 1 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
Jud Posted July 25, 2018 Share Posted July 25, 2018 1 hour ago, Miska said: Order is just as useless number to describe a modulator as is number of taps to describe a filter. Higher order modulator makes the increasing noise slope steeper, which in turn puts more demand on the analog filter. So just blindly increasing number of taps or modulator order is not necessarily best solution... Is there any significance to even vs. odd order modulators? ecwl 1 One never knows, do one? - Fats Waller The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature. Link to comment
Miska Posted July 25, 2018 Share Posted July 25, 2018 Just now, Jud said: Is there any significance to even vs. odd order modulators? No, it is the the same as analog filter order and such, it doesn't really matter... Jud and ecwl 1 1 Signalyst - Developer of HQPlayer Pulse & Fidelity - Software Defined Amplifiers Link to comment
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