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About romaz

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  1. Much has been said about how differences among servers are perhaps greater than difference among DACs. In my own experience, I both agree and disagree with this observation and I will attempt to explain. In my view, the DAC is clearly the more important component and why some DACs sell for >$100k. Because this thread was never meant to discuss DACs, I have shied away from commenting but given Moussa's post, I figured I should comment although this will represent my last post on DACs on this thread. In fact, this post will represent the beginning of my exit from posting in forums in general. Life has become too busy. When putting together an audio system, people will have their priorities. I have already stated mine and they are simply (1) resolution and (2) transparency. My reference isn't vinyl or tape or the million dollar setups one can hear at RMAF, Axpona or Munich, my reference is the live music I am often exposed to. Most of what I listen to is unamplified acoustical music, whether it be large orchestral, small ensemble, choral music, or solo instrumental (especially organ but also piano and guitar). When I am at an acoustical performance, whether it be classical or jazz, the first thing I notice is the acoustics of a venue and the resonances that venue provides. The natural reverb and decay of instruments and voices are quite evident and from the stalls to the balcony or from one venue to another, they will vary. It has been stated that the reverberation time in a large venue like Carnegie Hall measures between 1.8 to 2 seconds. At the Alice Tully Hall in the Lincoln Center just a mile away, this more intimate arena has a shorter reverberation time of 1.4-1.5 seconds. Which is preferable depends on whether I am listening to a solo guitar, four string quartet or a full orchestra but regardless, I very much enjoy hearing the acoustics of a great building and never would I prefer to hear music in an anechoic chamber. This is where most DACs stumble and where I find the DAVE excels. This is also where I find PCM superior to DSD. DSD provides you an expansive and a soft "tube-like" sound but this softness, which can be a wonderful way of masking the harshness of many chip DACs also results in a diffuse and imprecise presentation with respect to depth and timing and my careful A/B of my own recordings has convinced me of this. As someone who values the accurate spatial portrayal of a live musical performance, I have found that a good music server can provide much but a good DAC can provide more. When talking about resolution, as we look at our PCM files, we are provided 2 types of information: (1) bit-depth and (2) sampling rate. For Redbook, this means 16/44 which translates to 16-bits of dynamic range and a sampling rate of 44 kHz. While DR is important, I contend that sampling rate is much more important with respect to a DAC's abilities. When people talk about dynamic range, most people think about how loud and dynamically a DAC can play when really, it's about how quietly a DAC can perform that is important. With regards to DAC performance, Rob Watts equates DR to the "hiss level" of the DAC and the greater the DR, the less likely you are going to hear "hiss" when no music is playing. There is a DAC (that I will not name) that sells for >$100k and boasts a DR of 173dB (or 28.8 bits of dynamic resolution) as if we should be impressed by this. For those that know better, this performance metric is useless since most believe most humans are incapable of hearing beyond 21 bits of dynamic resolution. Just as important, most ADCs are also limited to about 21-bits of DR and so when people talk about 24-bit recordings, they often don't contain a true 24-bits of dynamic range. Even at 24-bits (or 144dB) of dynamic resolution, for those who choose to look at DR in the traditional way of how loudly something can play, listening to any sound at 144dB SPL would be considered lethal. Now this is what people fail to realize -- as soon as you connect DAVE (or any DAC) to an outboard headphone or speaker amp, you now have thrown away the DR capabilities of your DAC because now, you've buried the DR performance of your DAC into the much higher noise floor of your amplifier. For those who use an outboard amplifier with their headphones or speakers (this means most everybody who do not own a Chord DAC), you're basically listening to the much more limited dynamic range of your amplifier which is typically between 16-18-bits. With regards to sampling rate, I will explain why I consider this to be the more important spec with regards to DACs and this is why most DACs cannot match the performance of the very best turntables. Sampling rate gives you a measure of timing resolution and this provides you not just spatial information such as depth but also timbre accuracy and the layering of fine detail. With analog sources, you are hearing a continuous waveform and SQ is limited only by the quality of the gear that transmits this waveform. As such, it is generally easier to get great sound from an analog setup such as a turntable. With digital, an ADC is responsible for sampling the analog waveform a specific number of times per second and the larger the number of samples that are taken, the fewer the gaps of missing information there are and the more fluid or "analog" the recording sounds. In theory, a waveform that is sampled 176,000 times per second (hi-res PCM) will sound better than a waveform sampled only 44,000 times per second (Redbook). If that waveform is sampled an infinite number of times, then from a mathematical standpoint, your digital file becomes equivalent to your original analog waveform but as we know, infinite sampling is not possible based on the technology we have today and so this would suggest that digital can never truly equal analog. However, there is the practical matter of the limitations of human hearing that potentially make it possible for digital to equal analog. Most scientists agree that the human brain/ear has the ability to discern 2 separate sounds if they occur at least 5-7µs (microseconds) apart and so this represents the limits of a human's auditory time resolution abilities. This means that when 2 sounds occur 10µs apart, as an example, we can hear 2 discrete sounds but when these 2 sounds only occur 4µs apart, instead of hearing 2 discrete sounds, we hear only one blended sound. This is the rationale for why digital sounds "discrete" and why analog sounds "continuous." With Redbook, as previously stated, sampling occurs 44,000 times per second and this equates to a time resolution of 20.8µs. Anyone comparing a CD to vinyl in a resolving setup should easily be able to discern that with a CD, information is clearly missing. As you sample more often, let's say 96,000 times per second, time resolution improves to 10.4µs and while this represents a significant improvement, most ears will likely still be able to detect that an analog source provides more information. When you use an ADC to sample a file 192,000 times per second, time resolution now improves to 5.2µs. In theory, at this sampling rate, a digital file should sound virtually indistinguishable from the original analog wave form and so this is the basis for why hi-res files were created. This would suggest a 24/192 hi-res PCM file should sound equivalent to the original analog waveform. For those who have done careful listening, however, with most DACs, 24/192 does not equal analog and even DXD or DSD256 files still can't match the resolution of the very best analog setups. At most audio shows you attend, when you ask a certain exhibitor to give you their very best presentation, if they have a turntable or a reel-to-reel present, quite often they will switch to their analog source and, in fact, I have witnessed this many times. As a further example, having visited the Magico factory in Hayward, CA recently, they have arguably the finest listening room assembled in the world today. This room cost them $250k to build and has the equivalent of a floating floor and no parallel walls to avoid standing waves. Short of an anechoic chamber, it perhaps has the lowest noise floor of any listening room and they use this room as their lab. In fact, it is how they voice their speakers including their $600k Magico Ultimates and their soon to be released $175k M6. Here is a photo of that room: Because Berkeley DACs are the local favorite, they use a Berkeley Reference 2 DAC (Berkeley is headquartered nearby) fronted by a Baetis Reference server. However, when they wish to present their very best, they revert to their turntable. The reason is not so much because this sampling theory is faulty but because ADCs have limitations. It is the reason why such technologies like MQA were created and why many DACs oversample. Those in the NOS (non-oversampling) camp suggest that NOS DACs sound more natural but NOS strives only to reproduce the best that the ADCs can offer, warts and all. Oversampling is much more ambitious and strives to overcome the limitations of the ADC by interpolating the missing bits of information through the use of sophisticated mathematic filters. If the oversampling is done perfectly, a 16-bit Redbook file originally sampled at 44kHz per second should be audibly indistinguishable from the original analog waveform and this is the basis for the long tap-length filters that Rob Watts has been championing for decades but also the basis for what HQPlayer tries to accomplish. As to who does it better, I will leave it for others to decide for themselves but having listened to both approaches, I much prefer Rob's approach. As to the benefits of oversampling to DSD vs PCM, people will have their preferences, I have already stated mine. Regarding why some people fail to recognize great differences between DACs, I hear this all the time and I believe there are several reasons. As both a headphone and a speaker listener, I have found both types of listening to have their advantages. Headphones have the ability to portray fine detail better while speakers can image and soundstage better. DAVE is unique because its headphone output doesn't utilize a separate headphone amp. When you plug a headphone into DAVE's headphone jack, you are actually listening to the DAC itself. This means your headphone is tapped to DAVE's full bandwidth, ultra low noise floor (-180dB), dynamic range, and time resolution. Moreover, what is unique about DAVE is it has no noise floor modulation and so whether you listen to music at low levels or at DAVE's peak levels, noise floor remains at the same ultra low levels. There is simply no cleaner, clearer, more transparent way of listening to music than this. The problem with headphone listening is that headphones do not portray depth well, certainly not as well as speakers and so to this degree, a lot of DAVE's performance cannot be fully realized through headphones alone. The problem with listening to speakers with DAVE (or any DAC) is that DAVE's performance is largely buried in the amplifier you use to drive your speakers. While DAVE's performance still shines through, its performance is blunted as you end up inheriting many of the limitations of even the finest speaker amplifiers. Just like with outboard headphone amps, no speaker amp can match the performance characteristics of your DAC and so what you get with even the finest amps is a diminished photocopy of the original. Throw in a preamp, no matter how good, and this further adds to a loss of transparency. That is just the nature of adding components to your analog chain. Unless you are using a preamp for sound tuning (ie tube linestages), or you have an amp that demands a certain preamp to function optimally or unless you have multiple sources you need to switch among including a turntable, with DAVE, the very best preamp is no preamp at all. Just like with amplifiers, no preamp can match DAVE's performance with respect to distortion characteristics, noise floor, speed, dynamics, or time resolution. Even more, Rob programmed into DAVE the ability to attenuate down to whisper levels with absolutely no loss in resolution. That means that as you attenuate DAVE to its lowest level (-75dB), DAVE is still outputting full resolution, something that no preamp can match. In the photo below is VAC's very highly regarded Master preamp (about $30k): Kevin Hayes, VAC's designer, was kind enough to allow me to compare my DAVE driving his wonderful VAC tube amplifiers both with and without his Master preamp: It was the forgone conclusion of most people in the room that the sound through the attached Harbeth speakers would be vastly better with the Master preamp in the chain. They were surprised when this was not the case. Here is another example of a dealer's DAVE driving an $11k Constellation Inspiration Stereo Amplifier both with and without Constellation's $9k preamp. To both the dealer's and my ears, SQ was better without the preamp and so when this dealer sells a DAVE, he no longer tries to promote the sale of a preamp: And so what Moussa and ElviaCaprice are hearing is something that is very unique. Through their high-efficiency speakers, they are hearing the full potential of their Chord DACs limited only by their choice of cabling and speakers. With either the Omegas or the Voxativs I am using, I am hearing every bit of detail that my best headphones can provide while also the imaging and soundstage that only speakers can provide without the resolution and transparency robbing impact of an outboard preamp or amplifier. At the present time, I am trying out a pair of $25k Martin Logan Renaisssance Hybrid Electrostatic speakers in my large listening room, which I find to be very resolving and transparent. These speakers are currently being driven by a pair of Pass Labs XA60.8 class A monoblocks ($13.5k for the pair). While I cannot deny how wonderful this sounds when fronted by my DAVE, compared to DAVE directly driving my more modest pair of Omegas, this latter setup still sounds more resolute and more transparent. This is possible only with Chord DACs because only Chord DACs (as far as I'm aware) have output impedances that are low enough to directly drive speakers. In the case of DAVE, which has an output impedance of 0.055 ohms, this equates to a damping factor of 145, which is stellar. Soon, Rob Watts will be introducing amplifiers that will connect to his DACs via digital interconnects (not analog ones) and will have the same resolution and transparency characteristics as DAVE directly driving speakers. Essentially, these amplifiers will be "invisible" meaning they will have no character of their own. They will have class A output and the first amplifiers will output either 20 watts stereo or 70 watts in monoblock form. This technology is supposed to be scalable where 200 watts of amplification will be possible. Furthermore, as I have alluded in other posts, I have added Rob's new M-scaler to my DAVE. This is incorporated into Chord''s new Blu Mk 2, which is a CD transport that also includes a USB and BNC SPDIF input. This increases DAVE's TAP resolution to just over a million TAPS. This is a milestone that suggests Redbook is now completely indistinguishable from the original analog waveform and Rob didn't believe it would ever be achieved when he first conceptualized it back in the 80s but because of the rapid advancement of FPGA technology, this indeed has been achieved. Practically speaking, this results in a massive improvement in DAVE's resolution, so massive that the collective impact of my server mods which includes 8 clocks being replaced pales in comparison to what Blu Mk2 provides. For those of you who own a Chord DAVE, I would suggest you prioritize getting a Blu Mk2 beyond anything else discussed on this thread. Combined with Chord's upcoming "digital" amplifiers, there will be no more resolute or transparent way of listening to a digital file. Despite all of this, I am finding, however, that the quality of the music server still matters.
  2. Sorry for my delay in posting. Hopefully, I can post my more definite findings soon. Just to add to the discussion on clocks, it is unclear what clock frequency Ayre uses in its AX-5 Twenty but I suspect it won't be a 10MHz clock and you cannot compare phase noise measurements between clocks if they operate at different frequencies. This is according to Chris Peters, CEO of Mutec. When I asked him to compare for me the differences between the clock used in their well-regarded MC-3+USB vs their REF10, here is what he had to say: "The problem is that comparing the output signals of the REF10 and MC-3+USB is nearly impossible because of the different clock frequencies of both devices. The phase noise depends directly on the clock frequencies. We have measured the MC-3+USB with the nearest clock frequency of 11.2896 MHz to the REF10. The difference is approx. 30 dBc. That means the phase noise of the REF10 is approx. 30 dBc lower as this one of the MC-3+USB." When I asked him why I have been underwhelmed by my experience with certain "atomic clocks" in the past, here is what else he had to say about the REF10 and atomic clocks: "The phase noise difference e.g. between the REF10 and the Antelope 10M or the 10MX is approx. 40 dBc. That is the reason why people do almost not hear any difference when connecting the 10M to the MC-3+USB. The MC-3+USB is better the so-called Atomic Clock of Antelope." Now is a difference of approx 30dBc audible? Based on my experience in Munich, very much so and enough of a difference that I went ahead and bought the REF10. Here are the phase noise tracings for Mutec's REF10: So, how do you interpret this plot? Not so easy, it seems. There is a knowledgeable and well-respected poster here on CA (who I shall not name) who comments on the importance of phase noise measurements below 1Hz. Well, when I showed these plot tracings to Lee in Munich, he told me the important measurements to look at are between 10Hz and 100Hz and it is between these frequencies that a really good clock will separate itself from other clocks. In this regard, he felt the REF10 was stellar, perhaps the best he had ever seen and so it was clear to me that this is the frequency range he would be targeting with SOtM's master clock. Here is what Chris Peters had to say about this (I have intentionally X'd out the name of the CA poster in question): "I assume you are referring to the discussion in the CA forum regarding the audio-relevant measurement areas. And I think you are referring to "XXX" when talking about the range lower than 1 Hz. I honestly speaking do not know where or how "XXX" got this information. My beta testers have started trying to experience for audio most relevant frequency range since 2013 when we released the first version of MC-3+. So far we experienced audible differences between various oscillators when their specifications where different in a range between 1 Hz and 100 Hz distance from the carrier frequency of 10 MHz. Differences in this frequency range between the oscillators were audible best for my beta testers. So we optimized the REF10 for this frequency range specifically." So, to a large extent, Chris Peters is agreeing with Lee of SOtM. When looking at phase noise measurements of a clock, look at the measurements between 1Hz - 100Hz. What is interesting is that with better, clocks, you definitely hear an improvement in bass definition but improvements are also clearly heard in the midrange and treble. The best explanation I have for this is that if you improve the signal at 20Hz, for example, you would likely also improve its harmonic frequencies as well (ie 20Hz, 40Hz, 60Hz, etc). As an example, I am currently testing Synergistic Research's "Black Box." This is a $2k device that I had once assumed was just a glorified overpriced bass trap but it's impact is so much more. As I am presently testing potential speakers in my large listening room with cathedral ceilings, while my room is not a resonant nightmare, there are clearly nodes in my room resulting in boomy bass in different areas. With the introduction of this small box (which contains specially tuned passive resonators) into my room, not only does the bass boom disappear but midrange and treble clarity are greatly enhanced. This Black Box is definitely staying put. Anyway, back to clocking, as you further assess phase noise measurements, remember that with the REF10, phase noise measurements are taken from the BNC outputs and not from the clock. Taken from the clock, the measurements would probably be even better. I suspect any clock measurements Ayre might report will be provided by the manufacturer meaning the phase noise at the Ayre's outputs will most certainly be worse. It remains unclear what SOtM's new clock measurements are based from although I have posed this question to Lee himself. As a further example of how important this is, I have been testing clock cables of various price points and length. Using various inexpensive DigiKey clock cables from the same manufacturer of various lengths, as you go from 20 to 40 cm in cable length, the SQ degradation is clearly audible. This is why I had to send my gear back to SOtM. Because they didn't have the really short clock cables in stock, they ended up using much longer clock cables in my build which I ultimately deemed as unacceptable. Moreover, as I have tested identical length clock cables with my REF10 from companies like Pasternack ($40), Blue Jeans Cables (<$20), and Black Cat ($250) against the 700 Euro Habst clock cables that I purchased with my REF10, unfortunately, the differences are quite significant with respect to HF harshness and a very flat sound. Not that the cheap cables sound horrible but when you replace them with the Habst, there's simply no wanting to go back to those cheap cables. This is where those external clock doubters have a leg to stand on when they make their claims that external clocks don't add anything. Cable length and cable quality DEFINITELY matters. As to the impact of these clocks to signal timing, I don't believe it has anything to do with it. Referring back to John's Swenson's post on the REF10 thread that Rajiv provided a link to, the function of the clock in devices like the sMS-200, microRendu, Iso-Regen and tX-USB is not to time the signal (which is what a word clock is responsible for) but for "processing" the signal. How I interpret this is these clocks are necessary for the "functioning" of these components and so a good clock allows a component to function better. I'll use my production line analogy once again. When a production line operates smoothly and in a timely manner, less mistakes are made and less time and energy are wasted to correct any mistakes. This would be similar to how Audiophile Optimizer improves SQ, by removing unnecessary background processes, you get fewer software errors and fewer latencies which also translates to less current draw and less noise being generated in the ground plane. Now, does a better clock guarantee better SQ? Not always and if there is SQ improvement, this improvement can be variable. For example, having replaced the stock Crystek clock in the ISO-Regen with the clock in the REF10, is it now on equal footing as the tX-USBultra (which is now also being clocked by the REF10)? The simple answer is no. While the Iso-Regen definitely is improved on the REF10, my tX-USBultra (which is also being clocked by the REF10) is still the better component to my ears. In essence, and I have suggested this before, the best that a clock can do is to allow a component to perform at its very best but a good clock can never make a component perform beyond its physical capabilities. Absolutely, it is the circuit as a whole that matters and the clock plays just a small part of it. To elaborate on this topic further, while a good clock improves how a component operates, clocks contribute more than just timely functioning, they also contribute noise which can be measured in dB. Since there is no such thing as a perfect clock, all clocks contribute some degree of noise, it's just a matter of how much. When one bad clock in your digital pathway can contribute more than 30dB more noise compared to the REF10, imagine the cumulative noise impact of 8 noisy clocks in that same pathway. To some extent, you can mitigate that noise by throwing an sMS-200ultra into the pathway between your server and your DAC but as many of us have experienced, this doesn't completely fix the problem since adding a reclocking switch before the sMS-200ultra and adding a tX-USBultra after it results in further improvement. As I have now replaced all the clocks in my digital upstream from router onward, it is amazing how each clock replacement adds improvement but what I have also found is that the subsequent clock replacements seem to have less impact than before. What I am trying to say is that with the clocks replaced in my server, the impact of the ISO-Regen and tX-USBultra is now considerably less. I know there are people out there who are critical of the "spaghetti solution" that we call the SOtM trifecta. As an owner and originator of this trifecta, I have to agree it is a bit unsightly and cumbersome, especially when you incorporate the very inflexible SOtM dCBL-CAT7 into the mix (which I find to be indispensable). This is one reason I went away from it but what I can say is that it doesn't have to be this way for those who are opposed to it. With a low noise server (meaning a low power CPU, minimal RAM, avoidance of noisy storage drives, replacement of noisy clocks and driven by a clean, low-impedance PSU), this "spaghetti solution" is not only no longer necessary, but it is actually an inferior approach. Not to suggest that those that have this spaghetti solution should move away from it (because this spaghetti solution probably still sounds better than most things out there) but that there are other pathways to achieving similar (and better) results. Once again, while in Munich, Lee himself told me he considered the sMS-200ultra and tX-USBultra as his mid-level products. He reserved his highest praise for his sMS-1000SQ server and the Ultra version incorporates the sCLK-EX and also his very best card, the tX-USBexp. With this server, you can request to have the system board, Ethernet port and the tX-USBexp reclocked and everything fits nicely in one chassis. Where the sMS-1000SQ probably can be improved (based on what people have told me) is with its PSU. For those looking for an elegant, no-fuss, turnkey solution, I would suggest the sMS-1000SQ Ultra paired with a multi-rail SR7 and either SOtM's new master clock or the REF10. Can this be improved upon? Yes, I know it can and I will provide details of this in a further post.
  3. Yes, I'll post those findings, probably on the Blu2 thread. I will need to get the necessary adapter, however. Blu2 has Rob's improved filters. As you own Blu2, you're already good to go.
  4. I have heard it a few times. Initially, at CES in January when it was first announced and most recently at THE Show in Munich in May. Upon first listen, I was shocked because I thought I was listening to my DAVE. It was only when I directly A/B'd it against the DAVE in the Chord suite at CES did it become apparent that DAVE is still better but the gap is not as large as the price disparity between the two would suggest. I agree with John Darko's assessment, the dynamic slam is not as good as some DACs and Barrows is correct, this is due to the analog output stage that was intentionally compromised to make it portable. Having discussed the Hugo2's tech with Rob Watts while at CES, it incorporates his latest filters that are better than DAVE's. In my view, this DAC will be very hard to beat at its price point and even 2-3X beyond its price point. I believe it can compete against anything, regardless of price, and not get embarrassed. My Hugo2 should arrive next week and I can comment further.
  5. Hi, sorry for the delay in posting. It was always my intention to post my ultimate findings here and that will still happen but it will take a few days. Despite how big this thread is, Rajiv has done an excellent job indexing it and curating it and so much thanks to him!
  6. Jesus and Barrows, I will respond to your questions but this will be my last response. Feel free not to respond back. Chris, if you see fit to remove my post, I'm ok with it. Jesus, it really doesn't matter to me at all whether you like this "spaghetti solution" or not. It has nothing to do with it. As far as attacking you, I'm not sure where your paranoia comes from. As I see it, Sonore are the aggressors here. I merely countered Barrows' narrow minded view about a product he has never even listened to. Trust me, I don't have super human hearing. What I'm hearing is being heard by many, you just refuse to acknowledge there may be something there. It's interesting that you are waiting for measurements from John S. to validate the validity of clocking when you offer no measurements for your products. The only measurements I see come from ASR and we all know what those look like. Regardless, I didn't buy my microRendu or my REF10 because of measurements, I bought them for their potential to cause "goose bumps." As far as I know, there is no lab instrument that can measure for this. Sometimes, you just have to listen. ...if it actually meets the claimed specs??? Barrows, please give this manufacturer the benefit of the doubt that they have integrity in reporting their measurements. At least, they offer measurements. This is what I meant when I said that your bias and your agenda are clear. You say your comments are not meant to be critical and then you follow it with a veiled insult. Barrows, if you are going to extol the virtues of Rob's comments for your personal gain, you need to be prepared to defend his other comments. My point with bringing Rob's comments up is that despite his intelligence, wisdom and experience, even Rob doesn't know everything but at least he is willing to admit it. With regards to the REF10 and how it might impact a DAC or any other digital component, unless you've actually listened to what it can do, please keep your speculative comments to yourself. Yes, that's correct. Rob does not believe the source matters with his DACs. In other words, he believes all sources should sound the same based on the technology he has implemented in his designs and based on his own listening of various sources and so this would apply to this "spaghetti solution" as well. To Rob's credit, however, as you read, he was clear to say he doesn't know everything and so you'll never find Rob coming onto a thread like this and questioning the findings of others. Further to Rob's credit, he is actually willing to listen. He approached me a while back and offered to fly to my home in California later this year to listen to my sources. Obviously, I have agreed to host him as I welcome the opportunity. https://www.head-fi.org/threads/chord-electronics-blu-mk-2-the-official-thread.831343/page-83#post-13679993 No argument here. Sonore makes fine products. Please let them speak for themselves. Now moving back to the REF10, I have received more clock cables to try of varying price. I will report back once I have had a chance to evaluate them...
  7. Your bias and your agenda are quite obvious. Barrows, really, enough is enough. You should know that Rob Watts doesn't think highly of the microRendu at all but at least he is willing to admit he doesn't know everything. Because of the galvanic isolation he has implemented in his DACs and because his pulse array DACs are inherently immune to jitter, he believes all sources sound the same with his DAVE and that the microRendu is no better than a Windows laptop. Here is what Rob shared with me last year shortly after I bought my microRendu and I told him I was hearing an improvement: "Hmm, I am somewhat bothered by the idea that the microRendu is better than a windows (it must be windows) lap-top on batteries, as there is absolutely no explanation for why that may be. But "you know nothing Jon Snow" is my favourite quote for good reason; it reminds me that there are limits to one's knowledge." If you are going to come on to the REF10 thread and publicly question the value of an external clock like the REF10 based purely on your theories, at least have the decency to listen to it first. No one questions the value of using a good internal clock but sometimes, the components that we buy don't have the best internal clocks and that there are external clocks like the REF10 that are better. Your continued bleating about how using an external clock is a bad idea makes you look ignorant when those of us who have experienced the REF10 first hand clearly believe differently. Please, enough with the insults.
  8. As I have reported my findings, I have kept my power supply, speakers, amps and ears a constant. I have never blamed you. Not sure where that is coming from.
  9. Jesus, fair enough. Any measurements John comes up with will be trusted as properly done, however, whether they explain what I am hearing or not is irrelevant. As a proud owner of your products, I will remind you of threads such as these and the conclusions they have drawn based on their measurements: https://audiosciencereview.com/forum/index.php?threads/hardware-review-and-measurements-of-sonore-microrendu-v1-4.1867/ https://audiosciencereview.com/forum/index.php?threads/measurements-of-sonore-microrendu-streamer.577/
  10. This is what is so challenging with digital audio. No one truly understands it all. What I respect about people like John Swenson and Rob Watts is that they fully acknowledge this and that some phenomena are inexplicable but just because certain observations are inexplicable, does't make them invalid. Look how routinely Sonore and Uptone Audio get bashed on ASR based on lack of measurements. It seems some people, purely based on theories or measurements, already know how somethings sounds even before the needle hits the groove. With regards to the clock in a DAC, this appears to be a very complex topic and more than I am willing to tackle with my DAVE. Here is what Rob Watts has to say: "The issue of clocks is actually very complex, way more of a problem then in simply installing femto clocks. People always want a simple answer to problems even if the problem is multi-dimensional and complex. I will give you a some examples of the complexities of this issue. Some years back a femto clock became available, and I was very excited about using it as it had a third of the cycle to cycle jitter of the crystal oscillators we were using. So I plugged it in, and listened to it. Unexpectedly, it sounded brighter and harder - completely the opposite of all the times I have listened to lower jitter. When you lower jitter levels in the master clock, it sounds smoother and warmer and more natural. So I did some careful measurements, and I could see some problems. The noise floor was OK, the same as before, and all the usual measurements were the same. But you could see more fringing on the fundamental, and this was quite apparent. Now when you do a FFT of say a 1 kHz sine wave, in an ideal world you would see the tone at 1 kHz and each frequency bucket away the output would be the systems noise floor. That is, you get a sharp single line representing the tone. But with a real FFT, you get smearing of the tone, and this is due to the windowing function employed by the FFT and jitter problems within the ADC, so instead of a single line you get a number of lines with the edges tailing of into the noise. This is known as side lobes or fringing. Now one normally calibrates the FFT and the instrument so you know what the ideal should be. Now with a DAC that has low frequency jitter, you get more fringing. Now I have spent many years on jitter and eliminating the effects of it on sound quality, and I know that fringing is highly audible, as I have done many listening tests on it. What is curious, is that it sounds exactly like noise floor modulation - so reduce fringing is the same as reducing noise floor modulation - they both subjectively sound smoother and darker with less edge and hardness. So a clock that had lower cycle to cycle jitter actually had much worse low frequency jitter, and it was the low frequency jitter that was causing the problem and this had serious sound quality consequences. So a simple headline statement of low jitter is meaningless. But actually the problem is very much more complex than this. What is poorly understood is that DAC architectures can tolerate vastly different levels of master clock jitter, and this is way more important than the headline oscillator jitter number. I will give you a few examples: 1. DAC structure makes a big difference. I had a silicon chip design I was working on some years back. When you determine the jitter sensitivity you can specify this - so I get a number of incoming jitter, and a number for the OP THD and noise that is needed. So initially we were working with 4pS jitter, and 120dB THD and noise. No problem, the architecture met this requirement as you can create models to run simulations to show what the jitter will do - and you can run the model so onlyjitter is changed, nothing else. But then the requirements got changed to 15 pS jitter. Again, no problem, I simply redesigned the DAC and then achieved these numbers. So its easy to change the sensitivity by a factor of 4 just by design of the DAC itself - something that audio designers using chips can't do. 2. DAC type has a profound effect on performance. The most sensitive is regular DSD or PDM, where jitter is modulation dependent, and you get pattern noise from the noise shaper degrading the output noise, plus distortion from jitter. R2R DAC's are very sensitive as they create noise floor modulation from jitter proportionate to the rate of change of signal (plus other problems due to the slow speed of switching elements). I was very concerned about these issues, and its one reason I invented pulse array, as the benefit of pulse array is that the error from jitter is only a fixed noise (using random jitter source with no low frequency problems). Now a fixed noise is subjectively unimportant - it does not interfere with the brains ability to decode music. Its when errors are signal dependent that the problems of perception start, and with pulse array I only get a fixed noise - and I know this for a fact due to simulation and measurements. 3. The DAC degrades clock jitter. What is not appreciated is that master clock jitter is only the start of the problem. When a clock goes through logic elements, (buffers level shifters, clock trees gates and flip-flops plus problem of induced noise) every stage adds more jitter. As a rough rule of thumb a logic element adds 1 pS of more jitter. So a clock input of 1pS will degrade through the device to be effectively 4 pS once it has gone through these elements (this was the number from a device I worked on some years ago). So its the actual jitter on the DAC active elements that is important not the clock starting jitter. The benefit I have with Pulse Array is that the jitter has no sound quality degrading consequences - unlike all other architectures - as it creates no distortion or noise floor modulation. Because the clock is very close to the active elements (only one logic level away), the jitter degradation is minimal and there are no skirting issues at all. This has been confirmed with simulation and measurement - its a fixed noise, and by eliminating the clock jitter (I have a special way of doing this) noise only improves by a negligible 0.5 dB (127 dB to 127.5 dB). This is true of all pulse array DAC's even the simpler 4e ones. In short the jitter problem was solved many years ago, but I don't bleat on about it as its not an issue and because it's way too complex a subject to easily discuss. Pulse Array is a constant switching scheme - that is it always switches at exactly the same rate irrespective of the data, unlike DSD, R2R, or current source DAC's. This means that errors due to switching activity and jitter are not signal dependent, and so is innately immune from jitter creating distortion and noise floor modulation and any other signal related errors. The only other DAC that is constant switching activity is switched capacitor topology, but this has gain proportionate to absolute clock frequency - so it still has clock problems. I plan to publish more detailed analysis of this, but from memory all of my DAC's have a negligible 0.5dB degradation due to master clock jitter, so its a non issue. And yes you are correct, the absolute frequency is quite unimportant, so forget oven clocks, atomic clocks etc. Also the clock must be physically close to the active elements,with dedicated stripline PCB routing with proper termination. Running the clock externally is a crazy thing to do, as you are simply adding more jitter and noise and an extra PLL in the system."
  11. Barrows, first of all, much respect to you and Sonore. I am well aware of the great things you are capable of and having met you, Adrian, and Andrew at RMAF last year, and as a proud owner of a microRendu, know that I hold Sonore in high regard. Having partnered with John Swenson, you guys have some serious IQ over at Sonore and it shows. Contrary to how things might be perceived, if I have a bias, as an American, it is to see a small American company like Sonore succeed and succeeding, you guys are. Having said that, if my effusive praise of the REF10 seems disproportionate to your perceived value of it relative to other devices in a digital chain, I want to remind you that I am posting in the REF10 thread. If there is a place to be able to speak openly and candidly about the REF10, this is that place. Having clearly stated what DAC I use, people can make up their own mind about how it might translate in their own system. I would like to think that anyone reading this thread who can afford a REF10 and might be interested in a REF10 isn't going to be a fledgling audiophile and isn't going to use it to reclock their iPhone. Absolutely, the REF10 is not the first component anyone should buy but nonetheless, this is one remarkable piece of kit. There are many who jump onto this thread and make comments of what's possible or not possible based on theoretical grounds and have never actually heard the REF10. The perspective that I offer is that of an actual REF10 user. While it appears Sonore has not yet substantiated the impact of clocking in these "spaghetti" devices, I would encourage you guys to give it a go. Where there's smoke, there's usually fire, and so I believe you will find that the many who have gone down this path are not just listening to placebo.
  12. It seems that my statement has been misrepresented. I will quote what I actually stated and I stand by it: "No matter how good a DAC, no matter how many defenses it implements to ward off RF and jitter, it would appear to me that there are huge gains to be had by paying attention to your digital front end." Notice that I made no mention of a $3k clock. If my above statement were not true, then there would be no justification in buying a Sonore Signature Rendu SE (also selling for $3k) -- we should just pour this money into a better DAC. But to borrow your own phrase, Barrows, "everything matters," and my findings with better power supplies, better clocking, etc. with respect to these upstream digital components are just as applicable to my $600 Chord Mojo and $1,300 Oppo BDP-205 as they are to my Chord DAVE. Regarding what a good clock does in these upstream components, I agree with you. It's about noise and not timing. Here is John Swenson's quote: "As discussed in this thread a high frequency clock can have at least three different "uses". 1) integer multiple of a word clock, DACs frequently have two of these, one for each of the sample rate families. 2) clock used for "processing" not related directly to sample frequencies 3) reference for a frequency synthesizer (which may be producing #2 frequencies) Very few DACs actually use word clocks any more. Most of them use some form of #2 clock. The Ref10 is designed to be a #3 source. For most audio equipment it cannot be used directly since not very many boxes have frequency synthesizers built in. #2 are clocks that are fed into things like processors, USB receivers, PCIe boards etc. They are not related to audio sample rate frequencies at all, but observations seem to indicated that they can effect ultimate sound quality. The mechanisms for this are not known. I do have a theory I have been working on that explains this but it is very preliminary at this point and I'm not willing to make it public yet." It's ok if people are skeptical of the impact or the value of clocking in these small "spaghetti" devices. For sure, it shouldn't be anyone's first priority but just to be clear, when I report findings, that's all I'm doing. I have no agenda to promote as I am not trying to sell anything to anyone. I have paid for everything I have and I have no financial relationship with any audio company. Here is an interesting quote from John Swenson regarding the ultraRendu: "2) much lower jitter clock. When the microRendu project was started really good clocks were large and expensive so the choice was made to go with with a good but not best available clock. At this point we didn't understand how critical this next step in lower noise clocks was for many USB input DACs. But now VERY low noise clocks are much smaller and less expensive than they were (still not cheap, but significantly less than before) so it was decided it was very important to use these in this design. This is probably the most important improvement over the microRendu." It seems that with the ultraRendu, the clock does matter. All that the REF10 seeks to accomplish is to improve this clock even further. Regarding the value of a $3k+ external master reference clock like the REF10, value is always relative. As Julian so wisely said, compared to how expensive some external master clocks can get, before having heard it, I found the REF10's asking price to be fair. Considering the REF10 gives you 8 galvanically isolated clock outputs with a choice of 2 different impedances, considering its well implemented linear PSU, considering the very high build quality, but most importantly, considering its stellar performance, I consider the REF10 to be an outright bargain. As I consider the quality of my current music server build and as I compare it to things I have heard in the past including such things as a $17k Aurender W20 or my former $9k CAD CAT, this current build easily supersedes those servers and I attribute this to 3 trump cards: (1) my power supply, (2) the sCLK-EX boards that allow me to reclock almost any device, and (3) the REF10.
  13. I'm back from my travels for now. I also just a few days ago received my gear back from SOtM. I have gone a bit crazy with my build, more than I really needed to but I wanted to answer for myself certain questions about what matters and what doesn't with respect to all this clocking. While I will need more time to draw my definitive conclusions, here is what I can say for now. Back in Munich, I was quite enamored with how the REF10 elevated the SQ of Mutec's very fine MC-3+USB, enough so that I bought the REF10. While I fully believed the REF10 would elevate SOtM's products, I didn't know to what extent and so this was a bit of a gamble. I would have to agree with SwissBear that the REF10's impact on individual Ultra components from SOtM is less prominent when compared against what I heard with the MC-3+USB. Is this because the sCLK-EX's stock internal clock is better than the stock clock in the MC-3+USB? I'm not sure but if someone is going to buy a REF10 for something like the sMS-200ultra and stop there, I'm not sure I would consider a REF10 purchase worthwhile. Where the REF10 earns its keep (and boy, does it earn it) is when its abilities are applied en masse. From one component to the next, I have noticed a variable impact but as you add them together, it becomes an "OMG" moment. As of now, here are the 8 clocks I have replaced using 2 SOtM's sCLK-EX clock boards: 1. Netgear C3000 cable modem/router/switch - 2 clocks. There are more powerful devices like this out there with a richer feature set but I specifically chose this model because it uses low power components. While this device has Wi-FI capability, I don't use it. I use a separate device in a separate room as my Wi-Fi access point for my home and so this device only serves as an internet modem, simple router and 2-port switch. It only draws 1.5A at 12V and so it was my feeling that this would result in better SQ. When I compared it against my much more powerful Netgear Nighthawk Wi-Fi router which draws up to 3.5A, with each powered by their stock SMPS, I found slightly better SQ with this low power C3000. Powering either with a 12V lead from my Paul Hynes SR7 predictably improved SQ but again, I felt SQ was slightly better from the low power C3000. As such, I sent this unit to SOtM and they replaced its 2 clocks and added capacitors. The switching regulators could not be replaced due to excessive heat concerns. 2. DFI BW171 motherboard - 1 clock. This is a mini-ITX motherboard with an embedded Celeron that consumes no more than 8 watts. It is passively cooled, has a very small electrical footprint and when powered by my SR7, SQ was superior to my modified Mac Mini which was also powered by my SR7. This board's 25MHz system clock was replaced. This system clock then serves as a reference clock for many "subclocks" embedded within the motherboard and many of these clocks are not replaceable but there are ways around this. For example, the USB clock on this motherboard can't be replaced although if you use a PCIe USB card, the clock on that card can be replaced. 3. Intel i211AT 2-port LAN board - 1 clock. Because of lack of available PCIe slots, I chose to use the motheboard's integrated LAN board. Each of the 2 ports has its own clock that is replaceable but because these clocks are identical, I was able to use a single clock from the sCLK-EX board to provide clocking to both ports. If I was outputting to an Ethernet endpoint such as an sMS-200ultra or ultraRendu, I probably would have opted for a dedicated PCIe Ethernet card where capacitors can be added and the switching regulators replaced. 4. SOtM tX-USBhubIN - 1 clock. This internal USB clock board is the same board used in the tX-USBultra. Many will be surprised to know that I chose this card not as an output card but as an input card. Many who have followed my build on the other thread know that I noted a nice bump in SQ when I connected my music data drive to an Adnaco fiber USB card instead of one of the motherboard's stock USB ports. With further testing, I found a further bump in SQ with this card over the Adnaco and so this card is used to connect my array of Lexar 512GB compact flash cards for music storage (presently 2TB worth of storage). I have found this solution easily superior to anything I have tried thus far, either direct storage or NAS. 5. SOtM tX-USBexp USB card - 1 clock. This is SOtM's PCIe-based USB card. If I had another available PCIe slot, I would have bought 2 of these instead of the tX-USBhubIN card because according to Lee, SOtM's lead engineer, this card is his finest product. Of interest, while at Munich, he described his sMS-200ultra and tX-USBultra as his "mid-level" products. He suggested his very best product is his sMS-1000SQ and a large reason for this is because of this card. While similar in many ways, the USB chipset, regulator circuitry and the filtration built into this card are his very best. He said with this card in "Ultra" form, there should be no need for an sMS-200ultra. As an owner of an sMS-200ultra, I would have to agree. 6. ISO-Regen - 1 clock. I have to agree with SwissBear, this very fine device is a step down compared to the tX-USBultra with regards to resolution and while I like how it improved the tonal density of my slightly thin SOtM setup, the perceived compromise in resolution made it a "no-go" for me -- unless of course I could replace its clock and so that is what I have done. First off, SOtM will not replace the clock on any ISO-Regens. While they have great respect for Uptone Audio, they feel this device is too close a competitor. Feel free to ask them but don't be surprised if they politely tell you "no." With that said, there are many out there with the skill set to replace the 25MHz clock on the ISO-Regen for those so inclined. Is this modified ISO-Regen now at the level of the tX-USBultra? Not quite. I still prefer the tX-USBultra as my final endpoint before my DAC but the improvement is undeniable. 7. tX-USBultra - 1 clock. This unit served as the host for my 2nd sCLK-EX board and is the endpoint that connects directly to my DAC. What the above setup allows me is that from the beginning of my digital chain up to my Chord Blu Mk2 / DAVE, every bad clock that I can replace has now been replaced to the level of the REF10's OCXO. With the exception of my OS drive and my compact flash USB hubs that I use for storage, everything is powered by independent rails from several Paul Hynes SR7s. What I am saying is that both clocking and power delivery are of an equivalent standard from beginning to end. Now, Chord DAVE is an asynchronous USB DAC and in fact, this is its best input. Rob Watts implemented a floating USB design that provides very effective galvanic isolation to the extent that an Intona made no positive difference. Even the galvanic isolation offered by the ISO Regen results in minimal impact. Chord's new Blu Mk 2 with built-in M-scaler has now been added to my chain and what this effectively does is it increases DAVE's TAP resolution 6-fold to the extent where DAVE now has a TAP-length filter in excess of 1-million TAPS. This device also incorporates galvanic isolation and paired with DAVE, it was assumed by many including Rob Watts that all of my tinkering would have much less impact, if any at all. Despite the gigantic impact of the M-scaler in Blu Mk 2 to my Chord DAVE, my server build with all clocks replaced seems to be making an even larger impact than before. No matter how good a DAC, no matter how many defenses it implements to ward off RF and jitter, it would appear to me that there are huge gains to be had by paying attention to your digital front end. While the REF10s impact on any individual component ranges from small to large, its collective impact on my whole chain cannot be adequately expressed in words because as of now, I have never heard a system quite like this with respect to smoothness, resolution, transparency and dynamics. These are traits generally reserved for analog devices like amplifiers and speakers and so to make this claim with digital gear before the DAC is really quite a statement. Like many of you, I value simplicity whenever I can get it which is one of the reasons I went away from Ethernet endpoints like the sMS-200ultra. As a minimum, I know I could be very happy using a single sCLK-EX board (and its 4 clocks) and forgo reclocking the modem/router/switch, Iso-Regen and tX-USBultra. This means that with a single box server with integrated sCLK-EX board, a 4-rail SR7, and a REF10, I would feel I was in end-game territory but I will say that adding the reclocked modem/router/switch, ISO-Regen and tX-USBultra definitely improves things further. Lastly, with regards to clock cables, I have explored this to some extent although I have a few other clock cables coming in. With the Habst cables against a 0.5m Pasternack RG-59 (about $40) and a 0.5m Blue Jeans Cables RG-6 with Canare BNCs ($17), there's no contest, unfortunately. With these lesser cables, there is an obvious HF glare that is present. I can get used to it and things can sound pretty good but as soon as I swap in the Habst, the improvement in detail clarity and the disappearance of this glare is very apparent. If someone is contemplating an inexpensive clock cable, I would suggest the less expensive Blue Jeans Cables over the Pasternack. Whether this is because RG-6 is better shielded and has less signal attenuation then RG-59, it's hard to say but one seems to sound a little better than the other although the difference is small. Next week, I have a Black Cat Silver Star 75 coming in. It is supposed to play in the league of the big boys at a more affordable price point of $250 and so we'll see.
  14. Here is John Swenson's response to asynch USB DACs: "But what about asynch USB, isn't the DAC in control? Overall yes, the DAC has its OWN FIFO and also checks it, but instead of changing a clock frequency it sends a command back to the computer which tells it to speed up or slow down the average sample rate. So even though the local DAC clock is in ultimate control of the sample rate, as far as the MC3+/USB is concerned the USB data stream is in control, it just passes it on down to the DAC."
  15. This will have to be my last post (probably for the next few weeks). More and more are talking about output impedance these days but apart from Paul Hynes and SOtM, no one else seems to be able to provide measurements. I have to say it is disheartening to have manufacturers talk about how important a certain parameter is and yet they can't provide measurements. It leaves the consumer guessing. Regarding leakage currents, even John has failed to succinctly describe what this is supposed to sound like except that it is often subtle and so I'm pretty sure he isn't talking about the 50/60Hz hum that many of us hear with our amps or subswoofers. My guess is that it's supposed to sound like some sort of a haze or fog. All I can say is that with my SR7 compared against my LPS-1, I have absolutely no sense of such a thing. Thus far, these are the 2 very best PSUs that have come across my system but you may not want to count the sPS-500 out. Regarding how Lee applies filtration, I can't speak directly about the sPS-500 but as you know, Lee has intentionally stayed away from galvanic isolation (to DC) due to jitter issues. With his iSO-CAT 6 LAN isolator, this device is designed to filter from between 10Hz-20kHz. Because we are talking about the audible frequency range, he is unable to apply aggressive filtering without also adversely impacting SQ and so this is my belief why the iSO-CAT 6 LAN isolator has more of a subtle benefit. With the filter block used in his dCLB-CAT7 and now in this USB cable that he made for me, he more aggressively filters the harmonic frequencies (from 100kHz to about 1GHz). As you know, I found the impact of the dCBL-CAT7 to be quite notable when used within the "direct pathway" between server and Ethernet endpoint. With this filter block applied to a USB cable, the impact seems even larger, possibly because the USB pathway is even noisier. While I haven't accumulated enough hours on either this new SOtM USB cable or my Lush cable to draw any definitive conclusions, as of when I left home, I would say that the impact of the SOtM USB cable with the filter block is considerably larger and more desirable than what I was getting from the Lush. With Uptone's USPCB adapter, this is probably as close to an "invisible" USB cable as you can get and so this adapter probably comes closer to "doing no harm" than any USB cable but when compared against SOtM's USB cable with the filter block, this filter block actually seems to be "removing harm" brought about by the preceding component. In this sense, I greatly prefer it to anything I have yet heard. As someone who is already familiar with SOtM's detailed, airy and expansive "house sound" that Lee strives for in each of his components, you might find the sPS-500 very much to your liking. It's clear that Lee has tuned it to especially synergize with his endpoints even though it might not be the final word in low impedance.