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romaz

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  1. Hi Larry, During my time on this thread and in forums in general, I have always attempted to be honest and transparent in the reporting of my observations while also being fair and balanced with regards to my opinions and preferences. I recognize that as opinions are expressed, there will always be those who agree and disagree but I have always tried to be respectful of those who have preferences that are different from my own. Having communicated through PMs with you on several occasions, you also know my personal policy to never publicly disparage a manufacturer's product even if I prefer a competing product. With regards to my last post, if it has come across that I was disparaging HQPlayer or dCS, then I apologize. Having benefited greatly from many of your posts over the years, I have no doubt whatsoever that you have a wonderful setup that has been well thought out and carefully put together, one that I'm sure I would enjoy immensely and so if I came across as dismissive of your approach, I apologize. The last thing I wanted my last post here on CA to be was divisive or offensive, especially to someone like you whom I admire and respect greatly. With regards to dCS, they make fine products. Despite what I said, prior to the release of Hugo2, the Vivaldi was my 2nd favorite DAC behind the DAVE. I have not heard a DAC image better than the Vivaldi, better even than the DAVE. I have stated my preferences for Rob's DACs and I have provided my reasons why but they don't take away from the many things that dCS DACs do well. With regards to HQPlayer, as I own 2 copies of this software product for both Mac and PC, I feel I can speak to its capabilities from the standpoint of a user. I respect what it does and what it offers those who have DACs that benefit from them. If I had a point with my post, it is that I don't believe Chord DACs are the best candidates for upsampling with HQPlayer. Of course, people will wonder why and so that was the motivation for including those posts from Rob Watts. As you know, I was very impressed with my experience with the Sound Galleries SGM 2015 while in Munich in May although Ed Hsu even admitted that the Chord DAVE was not the ideal pairing for their HQPlayer-based server. Once again, this doesn't take away from the great options that HQPlayer provides for those who can benefit and based on its reasonable asking price, this should be viewed as a good value. With regards to marketing material from Chord, those quotes I provided are not from Chord nor were they meant to be marketing material. I have no affiliation with Chord, whatsoever. Those missives were made by Rob Watts who is independent of Chord and he speaks only for himself. As you and others have noticed, he speaks in very technical terms and rather than try and convey the technology behind Rob's DACs which I undoubtedly would misrepresent, I have found it easier to quote him and the reason I did so was there appeared to be confusion about how his DACs work. If you have perceived that I have bias towards Chord's products, you would be correct but more accurately, I have bias towards Rob Watts' designs -- they just speak to me in the same way that HQPlayer speaks to you. But I do wish to make it clear that this has nothing to do with any financial relationship I might have with him or with Chord as none exists. I have often reached out to the designers of the products I admire and this would include such individuals as Vincent Brient of TotalDac, Brian Zolner of Bricasti and of course, Paul Hynes. Perhaps I pursue these relationships to a deeper level than most but I have always been this way and so if I seem overly effusive of my praise for an individual's products that I admire, it is purely as an enthusiast. Hopefully, you'll judge me by my body of posts and not solely by that last post. Once again, much respect. Signing off for good this time.
  2. You are confusing air with depth. As I've stated, DSD has a very expansive sound but accurate spatial localization is not as good as PCM. This means a piano that is 5 feet away actually sounds like its 5 feet away. I have done these careful listening tests with a variety of DACs using both DSD and PCM rips from an analog master and PCM is better. With even a simple Mojo, if a recording is flat (as most studio recordings are), then it will sound flat and when there is depth in the recording, it will portray it much more faithfully. I enjoy the Lampizators very much. They premiered their new flagship at RMAF this past weekend and it was one of the few rooms playing digital that did not sound fatiguing. Yes, you get that dimensionality that DSD provides but this also comes from the Lampi's tube stage. There are other ways to create dimension, such as adding reverb effects via DSP but then everything sounds deep and unnaturally dimensional, including close-mic'd studio recordings. There are those that look for their DACs to make everything sound pleasing and this can be useful to cover up for harsh electronics or to make bad recordings sound better but this often comes at a compromise of resolution and transparency. If this is your cup of tea, that's fine. What I have found is that as I have improved my source with better power supplies, better clocks, better cabling, etc., I no longer find it necessary to cover anything up. At this point, my only priority is truth and accuracy.
  3. I had a wonderful time at RMAF this past weekend. It was a genuine pleasure to have met the likes of @kennyb123 and @Always.Learning. Having benefited from their insightful posts over the past couple of years, both on CA and Head-Fi, it was great to finally chat in person. If I am ever in Seattle, we will have to catch a live event together! Of course, there was the man himself, @The Computer Audiophile, who was generous enough to buy everyone a round of drinks during happy hour on Saturday evening. I wish there would have been more time to chat as I wanted to hear more about your "dog" story during your drive home from CES earlier this year. You are a class act, Chris. Last but not least, it was my privilege to have spent some quality time with both @austinpop and @limniscate during dinner on Saturday night and for breakfast each morning. Just a couple of great guys with fascinating backgrounds and true music lovers. I couldn't ask for better company! As RMAF was my 4th audio show this year (including CES, Munich, and the LA Audio Show), there wasn't much here that I hadn't already seen or heard but there was one consistent observation I was able to make and that was the very best sounding rooms were generally spinning vinyl or rolling tape. With very few exceptions, when digital was playing, the harshness and compression was painfully evident. In some rooms, it was bearable for only a few minutes, especially as this annoying glare was being amplified to very loud levels. As an example, within a few minutes of entering the Paradigm room where their nearly $20k top-of-the-line Personas were on display, I looked at Rajiv and we both knew what the other was thinking, that we had to get out of there in a hurry. It wasn't the speakers as I suspect they were very good. It was the digital front end. If there is one thing I have benefited greatly from with all the things we have learned on this thread, it is the reduction of that piercing glare that I once considered tolerable. Having gotten so accustomed to the lack of glare from the digital front end I have now, somehow, I have become all the more sensitive to bright and edgy electronics. There's simply no going back. As life has gotten extremely busy and will get even busier, unfortunately, RMAF will represent my last audio indulgence for the indefinite future. This will also represent my last post here on CA. It has been my pleasure to have been a part of this wonderful community of audiophiles. I will do my best to answer some of the questions that were directed at me. With regards to the sCLK-EX in my server chassis, this clock board is clocking 4 items and not just 3 -- (1) the motherboard's system clock, (2) both of the motherboard's LAN ports, (3) the tX-USBexp, and (4) the tX-USBhubIN. Regarding my choice of motherboard, I believe there are many very good boards out there. While I am very pleased with my DFI board, people should target the feature set they desire more than any particular brand. Having said that, a group of friends and I purchased a specific AsRock RACK motherboard some time ago that has the option of being powered via ATX or 12V. We powered this motherboard 3 ways: (1) directly from a 12V rail from my DR SR7, (2) from the 19V rail of my SR7 using an HDPlex DC-to-ATX converter, and (3) an EVGA Gold class 850 watt ATX PSU. With everything else kept constant, this motherboard sounded best when powered directly via a 12V rail from my DR SR7. The EVGA Gold ATX PSU sounded the worst by a very noticeable margin. This should not be taken as any kind of a sweeping statement against ATX PSUs as there are ATX PSUs that are undoubtedly much better than the EVGA we used but this is my strongest argument for using a low power motherboard. They are easier to power well and I have yet to hear anything that can do what an SR7 can. Regarding oversampling, yes, absolutely, you should oversample and I have stated this preference from very early on in this thread. As far as more companies like Chord finally getting on the oversampling bandwagon that dCS started 20 years ago, this is an inaccurate statement. Rob Watts first postulated his "1 million TAPS" theory while at university in Cardiff during the early 1980s, well before FPGAs were invented. He designed his very first upsampling DAC in 1995 (22 years ago) and all of his DACs moving forward have been upsampling DACs. As his DACs have evolved, they have improved with regards to TAP-length but also filter design and he has now reached the "1 million TAP milestone" with his current M-scaler that he never thought would be reached in his lifetime. dCS lost their chief designer years ago and so their ring DAC technology has stagnated since. The latest dCS Vivaldi Upsampler ($22k) can only upsample to either DXD (24/358) or DSD256. Even HQPlayer can do better. Does the DAVE, Blu2 or Hugo2 upsample to only 705.6 or 768kHz? Absolutely not. This would be just the beginning. DAVE, Blu2 or Hugo2 can accept any signal up to 768KHz PCM and DSD512 but then they upsample this signal much higher and through multiple WTA (Watts Transient Aligned) filter stages. As an example, when DAVE sees a 44.1 or 48kHz signal, the first WTA stage upsamples to 16FS (705.6 or 768kHz). Then the next WTA stage is at 256FS and finally, there is a third stage filter that upsamples all the way to 2048FS. This means a new filtered sample every 9.6nS. While there is no easy way to directly correlate PCM and DSD, this amounts to a degree of oversampling well beyond DSD2048. According to Rob, he upsamples to this degree for several reasons including to reduce the timing of transients uncertainty, to enable his noise shapers to work at 104MHz so that the noise shapers can reproduce depth correctly, and finally to allow no measurable noise floor modulation. What is a TAP and what is its significance? Here is a good article to read: http://www.the-ear.net/how-to/rob-watts-chord-mojo-tech How is a TAP calculated? Number of TAPS = (input sample rate frequency in Hz) * (delay time in seconds) * (oversampling rate) * 2 Are TAPS all that matter? Here is what Rob has to say: "Real tap count is vitally important - and for sure increasing the tap count with everything else the same - algorithm and over sampling rate - will result in much better sound quality. But there is much more too it than that, in that the algorithm is vitally important. The algorithm is the formula to calculate the coefficients for the filter, as an FIR filter is just audio data * coefficients added up to create the intermediate samples. When I first started designing with FIR filters in the 90's I had 2048 taps on an FPGA. Now this used a closed form type digital filter - the coefficients were calculated directly, and the filter guaranteed that the original data was preserved. But I heard this filter, and it was so much better than before. Now I knew that to guarantee 16 bit accuracy for the filter for all the calculated interpolated samples I would need getting on for 1M taps, and that was quite impossible way back then. So I thought can the algorithm allow a better timing accuracy without needing 1M taps? Was there a short cut that would allow perfect performance without using impossible tap lengths? Anyway I spent months thinking about it. I managed to figure out that there was an major measurable issue with filters - aliasing - and that removing this issue would help timing problems. But that would mean the original samples would get changed. And to me this was a very bad idea - it did not feel right, modifying the data. And I delayed trying this out. Now when you design, you are governed by your thinking, so your ideas set how you go about things. The problem is, we all make assumptions - XYZ can't sound good because of ABC - and assumptions that are not based on hard and rigorous listening tests are very dangerous. And I did not like the idea of changing the data - but I knew that aliasing could be something really important. So I eventually went ahead and designed a (pretty much - better than 16 bits) guaranteed non-aliasing filter. And when I heard it, it was amazing. Eventually this became the first version of the WTA filter, and subsequent tests showed that it at least gave a boost of an order of magnitude to the performance - a 256 tap WTA sounded much better than a 2048 tap conventional filter. But getting rid of aliasing was not the whole story, as increasing tap length does make huge differences, even when aliasing (which is vitally important in spite of the poor aliasing performance of MQA and other conventional filters) has been eliminated. So the moral of the story? Don't let seemingly good ideas influence your decisions - test all assumptions with very careful listening tests. Moreover, audio is very complex, and things are never as simple as they might appear. My mantra is "You know nothing Jon Snow..." and that is to remind me that there are very real limits to what I understand, and assumptions must be constantly tested with listening tests. And very big progress can be made by going down avenues that at first sight seem incapable of changing sound quality." Regarding the perfect interpolation filter, here is what Rob has to say: "For the reconstruction filter (the interpolation filter in the DAC) there is only one way of doing it, and that is using an ideal infinite response sinc function - that is if your intention is to perfectly recover the original bandwidth limited analogue signal. Any other type of filter will change the sound by adding uncertainty to the reconstruction of transient timing; and this uncertainty severely damages sound quality. Timing uncertainty due to small tap length filters add a softness, warmth and a bloated bass; some actually like that sound (I do not understand why as it sounds nothing like real un-amplified music) but for sure it is a distortion, or change, from the original. Now theory has nothing to say about how the signal is bandwidth limited; it simply states there must be zero output above FS/2. Actually, I have designed two decimation filters; one that is linear phase but rings before the impulse; one that is non-linear phase (IIR) but has no pre-ringing. Both filters have attenuation at and above FS/2 of 300 dB, so both meets the requirements of bandwidth limiting in terms of aliasing and sampling theory. I will post blind recordings with the different filters so we can all hear the difference." Is there an HQPlayer filter equivalent for Rob's WTA filters? Actually, all of Rob's DACs (including Mojo) upsample to 2048FS but they do so differently and so to say that you can match Rob's algorithms using one of HQPlayer's filters would be a gross overstatement. With Mojo, you have: WTA 16FS > Linear interpolation to 2048FS > Low pass filter > Low pass filter > noise shaper With DAVE, you have: WTA 16FS > WTA to 256FS > Linear interpolation to 2048FS > Low pass filter > Low pass filter > noise shaper Rob found that the addition of the extra WTA filter to 256FS in DAVE got him closer to the ideal sinc impulse response and the closer this gets to ideal, the closer the DAC gets to reproducing the signal in between samples. Regarding his WTA filters, they are proprietary and remain his IP (not Chord's). According to Rob: "The WTA algorithm actually uses my own windowing function of the ideal sinc impulse response. I took an awful lot of time (man years) with this, both in listening tests and in trying to understand what was going on - the understanding being used to allow me to try listening to different things. At the end of the day, the algorithm is fine tuned by listening tests, but you need understanding in order to change the critical parameters - in short knowing what those parameters are." Can a PC upsample to the degree that Rob's DACs upsample? Here is what Rob has to say: "PC's are very restricted in what they can do for real time signals. You simply can't replicate the processing that Dave does in a PC - simply because PC processors are sequential serial devices with a very limited number of cores. When you are doing a doing a FIR filter (a tap) you need to read from memory the audio data; read from memory the coefficient data; multiply the numbers together;then read the accumulated data and add that to the previous multiplication; then save the result. Lots of things to do in sequence. With an FPGA you can do all of these things in parallel at once, so a single FIR tap can be accomplished within a single clock cycle (obviously pipelined) - you are not forced to do things in sequence. With Dave I have 166 dsp cores running, plus FPGA fabric to do a considerable amount of further processing. You simply can't do that in a PC. To give you another example - converting DSD into DoP. You need a quad core processor to do this manipulation in real time - otherwise you get drop-outs - but in a FPGA I could do this simple operation thousands of times over, and at much faster rates than DSD256. What some people do not understand is how capable FPGA's are and how widespread they are used - the backbone to the internet? FPGA's. Search engines? FPGA's. Why? because an FPGA is fantastic at doing fixed real time processing - it takes small die area, and can do complex operations with very low power. Mojo for example has 44 dsp cores, uses sophisticated filtering to 104 MHz, and noise shapes at this rate - but does all this whilst consuming only 0.45 W. There is no way any PC consuming huge amounts of power can do this. Intel last year acquired Altera (an FPGA company) for $16.7 billion because they understand that the future of processing is with FPGA's A second issue is not what you can do but how you can do it - it is not just about raw power, but how the filter algorithm is designed. I have put many thousands of hours and over twenty years improving and understanding how to make a transparent interpolation filter; and I am still learning things today. And a third point is that a DAC is not simply a data processing machine but it has got crucial analogue parts too. If I dropped the WTA requirement, I would still need the same FPGA in order to do the noise shaping and other functions." Over the past few years, upsampling to DSD has become a trend by DAC manufacturers and by adopters of HQPlayer. Here are Rob's comments on DSD (probably not a surprise to many): "I was doing this 25 years ago in an attempt to solve the innate problems of DSD 256 (PDM 256). But I was using multiple, individually dithered noise shapers in an attempt to improve the noise shaper resolution. The benefits of doing this is that it reduces the jitter sensitivity, but does not eliminate correlated jitter problems. It's why in 1995 I invented pulse array as this eliminates correlated jitter (its a fixed switching activity scheme independent upon the output), improves (in the case of Dave) noise shaper resolution by a trillion times and eliminates noise floor modulation - something you can't do with DSD. What was also amusing is that they seem to be using an 8 bit shift register to re-time the outputs. I too used to do this too in the early 90's, but quickly found that using discrete flip-flops gave much better measured and SQ performance. Its due to switching activity on-chip changing the propagation delay of the OP FF - so making signal correlated jitter much worse. Also, power draw on each FF modulates the devices internal power rail, creating distortion. Using discrete flip-flops eliminates the signal correlated jitter issue, and with appropriate low impedance power planes, one can eliminate the PSU induced distortion problems too. This issue of correlated jitter problems can't be solved on-chip. A silicon chip design I worked on over ten years ago had this issue; on chip we had separate clock buffers, custom designed IO buffers, internal separate PSU paths, and all we could do was minimize the issue - it was still there on simulation, and still measurable in reality - although we reduced the problem by two orders of magnitude. But going discrete eliminates the issue. " Here's another comment: "There are actually two independent issues going on with DSD that limits the musicality - and they are interlinked problems. The first issue is down to the resolving power of DSD. Now a DSD works by using a noise shaper, and a noise shaper is a feedback system. Indeed, you can think of an analogue amplifier as a first order noise shaper - so you have a subtraction input stage that compares the input to the output, followed by a gain stage that integrates the error. With a delta sigma noise shaper its exactly the same, but where the output stage is truncated to reduce the noise shaper output resolution so it can drive the OP - in the case of DSD its one bit, +1 or -1 op stage. But you use multiple gain stages connected together so you have n integrators - typically 5 for DSD. Now the number of integrators, together with the time constants will determine how much error correction you have within the system - and the time constants are primarily set by the over-sample rate of the noise shaper. Double the oversampling frequency and with a 5th order ideal system (i.e. one that does not employ resonators or other tricks to improve HF noise) it converges on a 30 dB improvement in distortion and noise. So where does lack of resolution leave us? Well any signal that is below the noise floor of the noise shaper is completely lost - this is completely unlike PCM where an infinitely small signal is still encoded within the noise when using correct dithering. With DSD any signal below the noise shaper noise floor is lost for good. Now these small signals are essential for the cues that the brain uses to get the perception of sound stage depth - and depth perception is a major problem with audio - conventional high end audio is incapable of reproducing a sense of space in the same way one can perceive natural sounds. Now whilst optimising Hugo's noise shaper I noticed two things - once the noise shaper performance hit 200 dB performance (that is THD and noise being -200 dB in the audio bandwidth as measured using digital domain simulation) then it no longer got smoother. So in terms of warmth and smoothness, 200 dB is good enough. But this categorically did not apply to the perception of depth, where making further improvements improved the perception of how deep instruments were (assuming they are actually recorded with depth like a organ in a cathedral or off stage effects in Mahler 2 for example. Given the size of the FPGA and the 4e pulse array 2048FS DAC, I got the best depth I could obtain. But with Dave, no such restriction on FPGA size applied, and I had a 20e pulse array DAC which innately has more resolution and allows smaller time constants for the integrator (so better performance). So I optimised it again, and kept on increasing the performance of the noise shaper - and the perception of depth kept on improving. After 3 months of optimising and redesigning the noise shaper I got to 360 dB performance - an extraordinary level, completely way beyond the performance of ordinary noise shapers. But what was curious was how easy it was to hear a 330 dB noise shaper against a 360 dB one - but only in terms of depth perception. My intellectual puzzle is whether this level of small signal accuracy is really needed, or whether these numbers are acting as a proxy for something else going on, perhaps within the analogue parts of the DAC - I am not sure on this point, something I will be researching. But for sure I have got the optimal performance from the noise shaper employed in Dave, and every DAC I have ever listened too shows similar behaviour. The point I am making over this is that DSD noise shapers for DSD 64 is only capable of 120 dB performance - and that is some 10 thousand times worse than Hugo - and a trillion times worse than Dave. And every time I hear DSD I always get the same problem o perception of depth - it sounds completely flat with no real sense of depth. Now regular 16 bit red book categorically does not suffer from this problem - an infinitely small signal will be perfectly encoded in a properly dithered system - it will just be buried within the noise. Now the second issue is timing. Now I am not talking about timing in terms of femtosecond clocks and other such nonsense - it always amuses me to see NOS DAC companies talking about femtosecond accuracy clocks when their lack of proper filtering generates hundreds of uS of timing problems on transients due to sampling reconstruction errors. What I am talking about is how accurately transients are timed against the original analogue signal in that the timing of transients is non-linear. Sometimes the transient will be at one point in time, other times delayed or advanced depending upon where the transient occurs against the sample time. In the case of PCM we have the timing errors of transients due to the lack of tap length in the FIR reconstruction filter. The mathematics is very clear cut - we need extremely long tap lengths to almost perfectly reconstruct the original timing of transients - and from listening tests I can hear a correlation between tap length and sound quality. With Dave I can still hear 100,000 taps increasing to 164,000 taps albeit I can now start to hear the law of diminishing returns. But we know for sure that increasing the tap length will mean that it would make absolutely no difference if it was sampled at 22 uS or 22 fS (assuming its a perfectly bandwidth limited signal). So red book is again limited on timing by the DAC not inherently within the format. Unfortunately, DSD also has its timing non-linearity issues but they are different to PCM. This problem has never been talked about before, but its something I have been aware of for a long time, and its one reason I uniquely run my noise shapers at 2048FS. When a large signal transient occurs - lets say from -1 to +1 then the time delay for the signal is small as the signal gets through the integrators and OP quantizer almost immediately. But for small signals, it can't get through the quantizer, and so it takes some time for a small negative signal changing to a positive signal to work its way through the integrators. You see these effects on simulation, where the difference of a small transient to a large transient is several uS for DSD64. Now the timing non linearity of uS is very audible and it affects the ability of the brain to perceive the starting and stopping of instruments. Indeed, the major surprise of Hugo was how well one can perceive that starting and stopping of notes - it was much better than I expected, and at the time I was perplexed where this ability was coming from. With Dave I managed to dig down into the problem, and some of the things I had done (for other reasons) had also improved the timing non-linearity. It turns out that the brain is much more sensitive that the order of 4 uS of timing errors (this number comes from the inter-aural delay resolution, its the accuracy the brain works to in measuring time from sounds hitting one ear against the other), and much smaller levels degrade the ability for the brain to perceive the starting and stopping of notes. But timing accuracy has another important effect too - not only is it crucial to being able to perceive the starting and stopping of notes, its also used to perceive the timbre of an instrument - that is the initial transient is used by the brain to determine the timbre of an instrument and if timing of transients is non-linear, then we get compression in the perception of timbre. One of the surprising things I heard with Hugo was how easy it was to hear the starting and stopping of instruments, and how easy it was to perceive individual instruments timbre and sensation of power. And this made a profound improvement with musicality - I was enjoying music to a level I had never had before. But the problem we have with DSD is that the timing of transients is non-linear with respect to signal level - and unlike PCM you are completely stuck as the error is on the recording and its impossible to remove. So when I hear DSD, it sounds flat in depth, and it has relatively poor ability to perceive the starting and stopping of notes (using Hugo/Dave against PCM). Acoustic guitar sounds quite pleasant, but there is a lack of focus when the string is initially struck - it sounds all unnaturally soft with an inability to properly perceive the starting and stopping. Also the timbre of the instrument is compressed, and its down to the substantial timing non-linearity with signal level. Having emphasised the problems with delta-sigma or noise shaping you may think its better to use R2R DAC's instead. But they too have considerable timing errors too; making the timing of signals code independent is impossible. Also they have considerable low level non linearity problems too as its impossible to match the resistor values - much worse than DSD even - so again we are stuck with poor depth, perception of timing and timbre. Not only that they suffer from substantial noise floor modulation, giving a forced hard aggressive edge to them. Some listeners prefer that, and I won't argue with somebody else's taste - whatever works for you. But its not real and it not the sound I hear with live un-amplified instruments. So to conclude; yes I agree, DSD is fundamentally flawed, and unlike PCM where the DAC is the fundamental limit, its in the format itself. And it is mostly limited by the format. Additionally, its very easy to underestimate how sensitive the brain is to extremely small errors, and these errors can have a profound effect on musicality." Rob's quotes above were not placed here to suggest he is right and others are wrong. They were placed here so that people at least understand his perspective as a lot of misinformation is being spread. You can choose to agree or disagree. I suggest you perform your due diligence and let your ears decide and this is what I have tried to do. Do I agree with everything Rob says? No, I do not. For example, Rob believes all sources should sound the same with his DACs. It would appear Chord and their salespeople have bought into this philosophy also but my ears clearly tell me this is untrue and Rob will hear this for himself soon enough. I also wonder what would happen if Paul Hynes were to design a new PSU for my DAVE. Rob doesn't think it would make much difference but I have a gut feeling the difference would be significant, especially as I am using my DAVE to directly drive speakers. With regards to combining HQPlayer upsampling and one of Rob's DACs, I have already reported earlier in this thread that I have done this with my DAVE in the past. Remember that when DAVE (or any of Rob's DACs) sees a signal, whether it be 16/44, PCM 705/768, or DSD512, it takes that signal and upsamples it further. Using my Windows workstation, I ran this exercise with HQPlayer again this evening because I was curious. My new Hugo2 just arrived today and so I tried upsampling from Redbook to both PCM 705.6kHz and DSD512. I did it also with Blu Mk2. Just like before, upsampling to DSD512 with either of these devices resulted in worse SQ. Upsampling to PCM 705.6kHz had better results and perhaps there is an ideal filter/dither combination I failed to try but with the ones I did try, SQ was very acceptable but compared to feeding either Hugo2 or Blu2 the original signal, SQ via HQPlayer was drier and less smooth. With Blu2 combined with either DAVE or Hugo2, the improvement was massive. HQPlayer does not come close to what the M-scaler in Blu2 can do with either Hugo2 or DAVE. For those looking for a reasonably affordable alternative to upsampling with HQP, I suggest you give Hugo2 a try. Combined with even an unmodified NUC and an inexpensive endpoint like the ISO Regen or better yet, a tX-USBultra, I suspect it will give even the very best HQPlayer setups a run for their money. With the brief time I have had it, I am finding Hugo2 to be an excellent upsampling DAC and just like DAVE, it can directly drive high-efficiency speakers as well as headphones. I cannot overstate just how wonderfully resolving and transparent Hugo2 directly driving my Voxativs or Omegas sounds. Moreover, Hugo2 will combine with M-scaler to give you a full 1M taps, something that no other DAC except DAVE can currently provide and so as far as I am aware, when combined with M-scaler and when directly driving speakers, these are the 2 highest resolution and most transparent digital front ends in existence today regardless of price. I wish you all success in your audiophile journeys. Signing out...
  4. Thanks, Mark. Your system is well thought out and so good for you. In my case, the cost was painfully higher mainly because of so many things that I bought that ended up never making the final cut and so I have a lot of money invested in parts that are now just lying around (i.e. motherboards, CPUs, various RAM, various PCIe cards, USB hubs, the Adnaco, network switches, FMCs, routers, etc.). This is the price I've paid for knowledge but at least I've now satisfied just about all of my curiosities.
  5. No, I have never heard of those but I do find ball bearing-type solutions from the likes of Stillpoints and Symposium Acoustics to generally be more effective then the rubber or sorbothane-type footers. It depends. For large orchestral music or music with a lot of atmosphere, I am liking 4D. It provides the most depth. However, when I want tighter imaging, I'll go down to 3C. Signatures 1 and 2 sound too mechanical for my setup. Because I could do so while only utilizing 1 clock from my sCLK-EX board. I don't plan to build 2 servers although I have 2 listening rooms (home office with desktop setup and my main listening room). What I envision eventually is to bridge both LAN ports and then run CAT6/7 cabling from one of these bridged ports directly to my home office where I will install a SOtM trifecta. That way, both systems can benefit from this server. This is where the Adnaco solution could have worked well, however, my server only has 1 PCIe slot. Yes. While it makes no sense, it would appear that what's good for streaming is also good for CD ripping. Whether its due to noise that gets embedded into the stream that makes its way into the rip, I'm not sure. Paul Pang advocates a certain type of DVD-ROM drive for ripping CDs, specifically an ASUS model with a MediaTek chip that utilizes no internal clock of its own (presuming this clock adds noise to the stream). http://ppastudio.blogspot.com/2016/05/cd-ripping.html I bought a portable USB version of one of these these drives and I will run some tests to see if it makes a difference. Yes. As recently posted, I have tried grounding my DAVE via an Entreq Poseidon with Atlantis grounding cables via DAVE's XLR output, a Synergistic Research grounding block with HiDef grounding cable and most recently with Sound Galleries latest D2 grounding blocks and they made no difference. I have 2 of these D2 grounding blocks and have applied them to my music server but also to the spare USB port on my tX-USBultra and they result in improvements in those positions although the bigger difference is on my server.
  6. Yes, you're correct. Lee does not like optical solutions due to high jitter. As it was Lee that personally modified the Adnaco I sent him, he got a chance to listen to it and even with its 3 clocks replaced, he told me he thought it was a step back because It was sounding flat. It was at that point that he encouraged me to try his tX-USBexp. I knew I needed to hear it for myself and so he went ahead and completed the mod of the Adnaco per my request which included the replacing of several capacitors. Once I received the modified Adnaco back, I felt it had definitely improved and I liked the smoother and more solid image it was portraying. Around the time I received the modified Adnaco back, I was also able to audition the ISO Regen and I found the ISO Regen to be so good that in its stock form, it was equivalent to the modified Adnaco. Around this time, I was able to access an SOtM tX-USBexp (non Ultra) and so I compared it to the modified Adnaco. I found Lee to be correct, even the non-ultra tX-USBexp was sounding more impressive to me than the modified Adnaco and when I added the ISO Regen to the tX-USBexp, this combo was easily better than the Adnaco. Better yet, tX-USBexp + ISO Regen only utilized 2 clocks while the Adnaco solution by itself utilized 3 clocks although this combo costs more than the Adnaco. Having said that, the Adnaco definitely has merit and it does some nice things. For those who have a thin sounding system, you might really like what the Adnaco does and regardless of jitter, optical isolation definitely lowers the noise floor. For those who wish to keep their server in a different room or even a different building, you can purchase very long optical cables for very little money and still get excellent sound while remaining "straight USB." If you follow the Adnaco with a tX-USBultra, you come reasonably close to what the SOtM trifecta provides.
  7. Several have asked about the specifics of my current build. It has been in continual flux until recently but finally, I have reached my final build. I don't claim it to be the best there is but it is the best that I know how to put together and the best that I have thus far heard. To be honest, this is not what I envisioned it would look like when I first started this thread back in January but for those who have followed along, as you know, there were exciting and unexpected revelations that came to light and so this digital front end has become a much more ambitious build than I initially expected it to be. At so many points, it was sounding so good that I felt like finalizing my build but it was hard to stop when each successive gain seemed so significant. Along the way, many of you made constructive suggestions, some that I had never considered, and for that, I am grateful. They didn't always work out but the lessons learned and the perspective gained have proven to be invaluable. To that extent, I consider this build to have been a community effort and so I am happy to share what I have learned. Server chassis - Streacom FC9 Alpha ($285) I have used Streacom cases on numerous occasions in the past as I have found them to be attractive, well-constructed and minimally resonant. I chose the version that can hold a CD-ROM drive but eventually decided that I wanted nothing that would vibrate within the chassis as I felt this might adversely impact the sCLK-EX board and so I opted for an external CD-ROM drive to rip my CDs instead. This also provided me much needed space inside. If I could do it over, I would have gone with the cleaner look of the FC9 Alpha without the CD slot as pictured below (although I went with black instead of silver): Streacom makes smaller cases but I went with this larger case for flexibility. As you can see in the photo, when used with a mini-ITX motherboard, I was able to install my sCLK-EX board adjacent to the motherboard which allowed me to use very short 15cm clock cables. Also installed adjacent to the motherboard is my Intel X25-E SATA II OS drive. For those who want to go for the ultimate, Stillpoints makes standoffs that supposedly incorporate the same level of isolation as their Ultra Minis that could have been used to isolate my motherboard, SSD, and sCLK-EX board from the chassis. At $45 per standoff and since 12 standoffs were necessary, that would have cost me $540 for standoffs alone and so I passed. Instead, I bought a trio of Stillpoints Ultra Minis ($375 although I found a used set of 3 for $250) to use under the chassis and I have confirmed that these absolutely make a difference with respect to detail clarity (a larger difference than the inexpensive Black Raviolis I was previously using) although the overall improvement is small. SOtM eABS-200 paper ($120 for 1 sheet) I have used this type of EMI-absorbing paper in the past with several builds, specifically the Stillpoints ERS sheets but I have found the SOtM eABS-200 to be considerably more impactful with regards to a calmer and blacker presentation. If you are finding your server to sound fatiguing and bright, you might find this paper to be very helpful. Unlike other products, this paper supposedly absorbs a broad range of EMI and converts it to heat, thereby actually removing EMI instead of just scattering it. This product is so powerful that Lee advised caution as he indicated it was was possible to use too much resulting in an overdamped sound. I applied it to most of my chassis, to my RAM, and to my SSD and I have heard only good things. I even applied it to the inside of my tX-USBultra chassis and its collective impact is very easy to hear. I have found it well worth the $120. DFI BW171 motherboard ($331 which includes an embedded Celeron CPU) Finding a suitable motherboard was extremely difficult and it became s a matter of finding the motherboard with the least compromises. DFI was willing to design a motherboard for me to my specifications, however, it would have cost $1,500 minimum to get the project off the ground. I discussed the prospect of doing this with Lee of SOtM while in Munich and while he expressed interest, he told me he believed he had already figured out how to filter out most of the noise from a noisy motherboard with his tX-USBexp card. While I suspect there would have been something to be gained by designing my own motherboard devoid of unnecessary noisy components and noisy switching regulators, the time, effort and expense didn't appeal to me and so I ended up going with Lee's tX-USBexp card instead. Having built many servers over the years, none of the servers I had built could match the wonderful SQ I got from the $9k CAD CAT that I purchased in 2015. I opened this machine up and studied it carefully and realized that this was essentially a minimalist PC with detuned components (ie both the quad-core i7 and RAM were both detuned to run at 800MHz). At that time, for an i7-based PC with 4GB of RAM and 3TB of SSD storage to draw less than 20 watts was not a common thing to see. When I first got this machine, I went online to download some drivers and noticed that this thing ran as slow as molasses but boy, did this machine sound good. Heavily shielded SATA cables were used (made by Paul Pang) along with point-to-point DC cabling, EMI paper throughout, and finally a custom-specified PCIe-based USB card with a special clock. The whole thing was powered by a quad-rail custom HDPlex LPSU. I found this setup to be superior to a TotalDac d1-Server and to an Aurender N10 that I had on hand and so I bought the CAD CAT but I also knew this server could be improved upon (especially with regards to its power supply). https://www.head-fi.org/threads/review-comparison-of-5-high-end-digital-music-servers-aurender-n10-cad-cat-server-totaldac-d1-server-auralic-aries-audiophile-vortex-box.787020/ As far as choosing an actual motherboard for this build, based on my experience with my CAD CAT and as I had no intention of oversampling, I knew I wanted a low power setup. I was further convinced when I directly compared the SQ from a very powerful HP workstation (with dual Xeon CPUs, 64GB of RAM, and PCIe-based SSDs) against a powerful Mac Pro (with a 12-core Xeon, 64GB of RAM and 1TB of PCIe-SSD storage) and found that the SQ I got from a much less powerful stock Mac Mini with the OS running off of a low power SD card to be easily superior. With the Mac Mini modified to accept 12V power from a Paul Hynes SR7, it was no contest. The SR7 was such a difference maker that I chose to only look at options that allowed me to use my SR7. Having tried various motherboards from Gigabyte, AsRock, SuperMicro and Dell, I ended up with the DFI BW171 for the following reasons: 1. SoC (System on a Chip) architecture that did away with the slower (higher latency) processor controller hub (PCH). 2. Mini-ITX form factor resulting in shorter data paths 3. Simple Celeron CPU with only 6w TDP 4. Could be powered from a 12V rail from my SR7 If there is a downside to these types of motherboards, they generally only contain 1 PCIe slot (usually a PCIe x4 or slower). If they contain an M.2 slot, they run off the slower SATA bus and not the faster (lower latency) NVMe bus. There are other motherboards that could have fit the bill but I went with this company because they had presented me the option of designing a new motherboard from scratch for me if I found this motherboard to be unacceptable. Thus far, I have been very pleased. As an industrial motherboard, it has not only proven to be very reliable running 24/7 but even prior to modification, I found it to sound at least as good as my Mac Mini with SD card even though my OS was running off a noisier SSD. With this motherboard running Windows Server 2016 + Audiophile Optimizer (something that a Mac Mini with SD card cannot run), this machine was sounding even better. Intel X25-E Sata II SSD ($80) + SOtM SATA II filter ($65) + Pachanko Reference SATA cable (about $250) for OS storage Those of you who have followed this thread from early on know that in my testing, I had found the impact of the OS drive to be greater than a separate data storage drive with respect to SQ and this is probably due to the fact that OS drives are in perpetual use, especially with Windows, while a separate data storage drive is only in use up until the digital file has been buffered into memory, at which time it becomes idle. Those of you who have followed this thread from the beginning also know that while I was initially quite happy with my Mac Mini with its MacOS running off of an SD card, I was also looking for a way to run WIndows Server 2012R2 + Audiophile Optimizer. Unfortunately, no recent flavor of Windows Server can boot from either an SD or compact flash card and so I purchased a 2nd Mac Mini but this time, it had a PCIe SSD card installed. I was successful in installing Windows Server 2012R2 + Audiophile Optimizer and while I was pleased with the improvement over MacOS, I also noticed a fatiguing HF harshness that was not present with my Mac Mini with the SD card and no matter what I did, I couldn't get rid of it. It was one of those annoyances that you may not initially notice is there but every time I went back to my other Mac Mini with the SD card, I found myself not wanting to switch back as the SSD setup was more irritating and fatiguing. While SSDs are acoustically silent, it turns out they are electrically very noisy, especially the faster and more powerful SATA III SSDs that are being sold today as they apparently emit noise in the 6GHz range that is very difficult to mitigate even when powered by something like an LPS-1. There are many who have come up to me and have suggested that they aren't hearing the harshness that I'm hearing with their SSDs but those that have had a chance to compare my 2 Mac Minis have easily picked up how much calmer and less fatiguing my Mac Mini with SD card sounds. Based on this experience, I felt challenged to find a way to install Windows Server onto a compact flash card and I eventually succeeded but ultimately had to give it up since compact flash boot drives are only possible with motherboards that incorporate a BIOS that allows for the older PATA drives and unfortunately, none of the SoC motherboards I wanted to use allow for this legacy option. I considered the option of using a spinning hard drive for my OS which I had confirmed is electrically quieter than an SSD but I didn't like the idea that I would have a spinning and vibrating drive in the same chassis as SOtM's sensitive clock board. Over on the JPlay forum but also on Paul Pang's website were reports of how SATA II SSDs sounded better than SATA III SSDs with SLC sounding better than MLC or TLC and so I went ahead and purchased a NOS (new old stock) Intel X25-E 64GB SLC SSD for about $80 from EBay. This turned out to be a very good purchase as indeed, this lower power, slower SSD drive was sounding less harsh and fatiguing then my Samsung 850 EVO SSD. The downside of older SATA II SLC SSDs is that they're not available as high capacity drives and so they can function just fine as an OS drive but they don't have enough capacity to store much of a music collection. Based on the limited capacities of SLC SATA II SSDs, I elected to go with the Intel X25-E for my OS drive and with compact flash cards for the storage of music files. Here is where things get really interesting. When combined with the Pachanko Reference SATA cable, with my DFI motherboard, I felt I had equaled the SQ I was getting from my Mac Mini with SD card as a boot drive and so I found myself quite content with this solution. While at Munich, May realized I was using a SATA II drive for my OS (and not a SATA III drive) and so she strongly suggested that I try SOtM's SATA II filter. I knew they made both a SATA II and a SATA III filter and I asked if the SATA III filter was the improved version. I was surprised to hear May tell me that their SATA II filter is better but unfortunately, this filter does not work with SATA III drives and so they were forced to design a new SATA III filter. What the SATA II filter apparently does that the SATA III filter does not is that it filters both the SSD's power and data lines (their SATA III filter only filters the power line). Here is a picture of their SATA II filter: Here is a picture of their SATA III filter: Those that know Lee knows that he is a filter specialist. In the same way that his filtering methods have transformed products like the tX-USBultra but also SOtM's dCBL-CAT7 and their new USB cable, this SATA filter does the same for SATA II drives and the impact is astonishing with respect to a lower noise floor devoid of any HF harshness but also this more open soundstage. The impact of this filter is eye opening in terms of just how much noise OS SSD drives create. This $65 filter is definitely one of the stars of the show and something that I consider a "must have." I would say its impact is considerably greater than the Pachanko SATA cable or the Intel X25-E SSD. Operating System - Windows Server 2016 Essentials ($124) + Audiophile Optimizer ($129) I believe the full version costs more but I was able to purchase a license for the Essentials version for only $124 from an online vendor. Even the GUI version of WIndows Server 2016 sounds better than the Core version of Windows Server 2012R2 and so for now, I am sticking with this version. Combined with the latest AO beta and its digital filters and sound signatures, it is the best sounding OS solution I have yet heard for music playback for Roon but also for ripping CDs via dBPowerAmp. The benefit of Windows above and beyond the Linux OS's I have heard is software compatibility. With Windows, I know I can (or will be able to) run just about anything now and in the future. More than that, I am finding Chord's ASIO driver for Windows to be superior to the stock Linux and Mac drivers for my DAVE. With my sMS-200ultra or microRendu, any time I chose to play DSD128 or DSD256 files, I frequently got skips and pauses. Not a problem at all with Chord's ASIO driver for Windows as now I am capable of smooth native DSD playback of all my DSD tracks. Also, Windows Server 2016 has been super stable. I haven't had to reboot yet due to a lock up or some system instability since I installed it. sCLK-EX board ($850 with 4 clock outputs activated) Along with Chord's new Blu Mk 2, this device probably gets my vote for most revolutionary product of the year. This clock board has opened up new vistas I never knew were possible just a year ago and upon replacing clock after clock, I am just amazed at how my system has become transformed. Even without connecting to an external master clock, the impact of replacing 8 clocks in my setup using this board has just been astonishing. It is this clock board that makes my build different from any other server on the market today (except for SOtM's own sMS-1000SQ server). When people tell me their server has this or that, knowing that their system probably still contains a bunch of noisy clocks and having now experienced just how big the impact of replacing clocks can be, I know their server can still sound so much better. With the sCLK-EX board in my server chassis, I am using 1 clock for the motherboard's 25Mhz system clock, 1 clock to concurrently replace both of the integrated LAN clocks, 1 clock to replace the stock clock in SOtM's tX-USBexp and tX-USBhubIN. WIth the sCLK-EX board housed in my tX-USBultra, 1 clock is being used for the tX-USBultra itself, 1 clock is being used for my modified ISO-Regen, and 2 clocks are being used for my Netgear integrated internet modem/router/switch. SOtM tX-USBexp ($350) Considering the importance of this card in my system, I consider it's $350 price a bargain. As previously stated, according to Lee, this is his best product. The photo below compares the tX-USBexp to the tX-USBhubIN and you will see that the tX-USBexp is nearly twice as large with many more parts. While this card has been around for awhile, I believe it has gone through improvements with time. While the newer tX-USBhubIN is still using a USB 2.0 chipset, the tX-USBexp is using what Lee considers to be a better sounding USB 3.0 chipset. This card also incorporates his best regulator circuit and better filtration. What is interesting is that SOtM is an OEM supplier for other music server builders and I noticed that both Antipodes and Baetis have chosen to use SOtM's tX-USBhubIN for their servers. When I asked Lee why they would do this, all he could say is that "they didn't know any better. I designed both of these cards and I know which one is better." With SOtM's own sMS-1000SQ, this is the card they use. What is interesting is both the tX-USBexp and the tX-USBhubIN cost the same. SOtM tX-USBhubIN ($350) Based on what Lee has told me, if I had 2 free PCIe slots, I would have gone with 2 tX-USBexp cards, especially since they cost the same. At the same time, I am extremely happy this option exists. Even though my motherboard has only 1 free PCIe slot, my Streacom chassis has 3 available slots in the back. This card occupies one of those free slots and then connects to my motherboard via its USB 2.0 header. In the same way that SOtM's SATA II filter significantly improves the SQ of my OS drive, this card does the same for my data storage drives. Compared against the motherboard's stock USB ports, there is greater immediacy but also a more open soundstage. Even with the clocks replaced on my router, using the identical track played from my compact flash drive connected to the USB port on the back of my router (you could consider this a NAS) vs the same compact flash card connected to the tX-USBhubIN, playback through the tX-USBhubIN sounds noticeably better. Here's another benefit of having both the tX-USBexp and the tX-USBhubIN. When ripping a CD, I can now connect my external USB CD-ROM drive to my tX-USBultra USB port (which is directly connected to my tX-USBexp) and the files rip directly onto my compact flash drive connected to my tX-USBhubIN. Using dBpowerAmp on Windows Server 2016 + AO, these are the now very best CD rips I have ever made. Lexar HR-1 Compact Flash Hub (approx $90 + $23 for each additional CF reader) This device can be powered with a 5V iFi PSU (no difference heard compared against an LPS-1). It has the capacity for 4 readers (CF, SD, microSD, USB, etc) and works very well for its intended use. Other devices would probably sound just as good when connected to the tX-USBhubIN but I already had this one on hand. Lexar has discontinued their 512GB Compact Flash cards although I was able to buy several of them from B&H Photo for $199 each during their closeout sale. ISO-Regen ($325) I purchased mine used from another CA member. I didn't find it as resolving as the tX-USBultra (my main complaint with it is that it tends to flatten the sound) but it added a certain smoothness and a weight to my piano tracks that I found very much to my liking. In this sense, I am using the ISO-Regen as more of a tuning aid but I definitely like what it adds when placed between my tX-USBexp and my tX-USBultra. Replacing its clock has definitely improved it. tX-USBultra with 12V option and 75ohm Master Clock connectors ($1,200) If I didn't need the extra clock taps in the tX-USBultra for my router, I could have easily skipped the tX-USBultra (and ISO Regen) because my single box server paired with the REF10 is already so good, however, adding a reclocked ISO-Regen and tX-USBultra definitely adds further improvement and for those looking to go to infinity and beyond, I believe this gets you there. Having opened up my tX-USBultra, I can see why Lee couldn't fit the tX-USBexp into the chassis, it's just too big. Most of this chassis is occupied by the sCLK-EX. Netgear C3000 cable modem / router / switch ($90) This particular Netgear device that integrates a cable modem, router and switch into one small chassis was chosen for its low power characteristics. As I don't use this device for its wi-fi capabilities, not only did I save a fair amount of money not having to buy an expensive NightHawk (for up to $400) but I found this low power device (12V/1.5A) to sound a bit better than a $400 Nighthawk and so I sent it along to SOtM for modification. One thing that is interesting but not surprising, when I connect my Mac Mini to this device, even with the assistance of an SOtM dCBL-CAT cable and iSO-CAT6 LAN isolator, the improvement heard is there but it's small. The improvement is much larger when connected to my server that has it's LAN clock replaced. Undoubtedly, it has improved Tidal streaming and considering how little I paid for it, this was a very worthwhile thing to do. I am currently powering this router with my sPS-500 and it does a very good job. Mutec REF10 + 2 Habst Cables (approx $4k) The REF10 is the icing on the cake and it is what elevates my setup to simply otherworldly status. Even with the REF10 turned off, this new setup is already the very best I have heard in my system at home sound but with the REF10 turned on, the first thing that becomes immediately apparent is this sense of buttery smoothness that I have never before experienced (especially when combined with Blu Mk 2), even when compared against the best turntables I have heard. The benefits provided by each component with just the sCLK-EX in place are enhanced to a higher level with the REF10 activated but it is this smoothness (ie complete lack of glare) that is so mesmerizing. It's quite possible that SOtM's new external master clock will be better, especially since it can be powered by something like an SR7 but if you compare one against the other, I suspect it will be a matter of splitting hairs meaning they both will sound very good. Where the real benefits become apparent are when you replace multiple clocks and especially when you replace every clock in your chain. USB cables (SOtM USB cable with filter block - $1,000, Phasure Lush USB cable - approx $275, Uptone Audio USPCB adapter - $35) Like many of you, I purchased a Lush USB cable from Phasure. Mine is 0.75m long and I paid 237 Euros. Unlike many of you, I don't care for this cable as much. It does some nice things with respect to tone and warmth but it flattens sound. The perceived loss of spatial resolution when compared against SOtM's new USB cable with filter block is quite stark and even greater than ISO Regen vs tX-USBultra. With my setup already so very smooth and tonally rich, I'm not sure the small benefits it imparts are worth the very significant downside although I haven't completely made up my mind to sell it. At the present time, I am using it to connect the ISO Regen to the tX-USBexp but I am thinking my Clarity Cables Natural is sounding better at this position. I am then using the USPCB cable came with the ISO Regen to connect it to my tX-USBultra. At this time, I am finding USPCB cable preferable to the Lush. I am using my very best USB cable, SOtM's latest USB cable with filter block to connect the tX-USBultra to the USB input on my Chord Blu Mk 2. Regarding SOtM's new USB cable with filter block, I believe this cable will be officially showcased for the first time at RMAF in a few days. It is now the best USB cable I have heard although I suspect there are some who will still prefer the tonality of the Lush. It uses the same filter block as the dCBL-CAT7, however, I find its impact to be greater and I believe I know why. With the dCBL-CAT7, I was using it before the sMS-200ultra or before the reclocking switch. With SOtM's new USB cable, I can use this cable much closer to my DAC where it is imparting a greater effect. If I swap this cable with the USPCB adapter or worse yet, the Lush (meaning I place it back further in the chain), it's impact is significantly diminished. Placed right before the Blu Mk2, the soundstage opens up beautifully. I went with UPOCC silver as a conductor and the improved detail resolution over the Lush is quite evident. Paul Hynes SR7 I would be remiss not to mention the SR7 as I consider it the foundation of my digital system behind only my DAVE and my Blu Mk 2. If I could get Paul to build all my power supplies, I would be a very happy man as this guy has a magic touch. Yes, there's a long wait for one and even now, I'm waiting patiently for him to build me another SR7 but I'll happily wait as long as I need to. Other tweaks - Synergistic Research Tranquility Base XL UEF ($3,250) I found one used for $1,300 and so I took a chance on it. It was used at an audio show and it had some obvious cosmetic wear but was in perfect operational condition. At the very least, it would serve as a means of mechanical isolation in my equipment cabinet and perched on top of a trio of Synergistic Research's MIG 2.0 footers, it did a very nice job. When you turned this device on, however, I found it to be transformational in its impact. I didn't think it was possible but detail clarity improved further and the improvement was not subtle. This impact was remarkable enough that I felt compelled to squeeze every electronic component I had onto this 20" x 23" base including the REF10, server, tX-USBultra, Chord DAVE and Blu Mk2. They just barely fit and while the whole arrangement isn't as aesthetically tasteful as what I had before, I cannot deny what this thing does and the resultant improvement in detail clarity. This thing supposedly addresses airborne EMI but also EMI in your electric components. Just when you thought you had sufficiently addressed EMI, this device tells you there's still so much more of it. I was sure this was voodoo when I first heard about it but having heard a brief demo at an audio show, I felt compelled to give it a shot at home. At $3,250, I would question its value but at the price I paid, I consider it a must have. This will likely be my last post of this depth and detail. I hope to see some of you at RMAF in a few days.
  8. You make very good points, especially about the THD of speakers. I have not heard a Lyngdorf although now, I would very much like to.
  9. This is correct. When I am at a live performance, never am I thinking to myself that I wish the sound was warmer or cooler or that there was more bass or that the midrange had better detail clarity. If your system is well-balanced, resolute and transparent, then it becomes more about the quality of the recording and the quality of the performance itself. I have found that as my server and DAC have improved, several things have happened. Glare has dropped profoundly, the soundstage has opened up and the presentation is now smoother and more tonally rich, so much so that I have no desire to color it. In the past, as a former tube guy, I loved how certain tubes resulted in a more holographic image. They also helped smooth out what I considered a fatiguing and unengaging "digital" sound. Now, I find these same tubes only to be a hindrance as they impact resolution and transparency. Instead of applying cologne to body odor, sometimes it's better to take a bath. In my opinion, the most difficult thing for an audio system to convincingly portray is an unamplified human voice followed by an unamplified musical instrument. If a system can get these things right (or as close as possible), they can generally get the other stuff right. I do listen to other things when I evaluate gear including heavily amped music, pop, rock, EDM, etc. I just find that they are simply less challenging. We each have our methods of how we evaluate equipment. There are many ways to do it but here is what has worked for me. When evaluating equipment, I believe we have to be very intentional, meaning we can't just sit there and listen. Since DACs can be among the most difficult pieces of equipment to evaluate, I will focus on DACs for now. Some have shared with me that they look for a DAC that moves them emotionally and while this is what we all want our systems to do, when evaluating the performance of a piece of equipment like a DAC, it should be more objective than subjective. We also have to remember what a DAC is supposed to do and not unfairly burden it with responsibilities that belong to another component. A DAC is supposed to extract information from a digital file and faithfully convert it to an analog waveform. The best DACs retrieve not only the subtlest details but should be capable of presenting these details in the proper timing. The best DACs should be invisible meaning they should have no sound of their own. This is why comparing DACs can be so difficult. Unlike comparing headphones, speakers, or even amplifiers, how are you supposed to assess something that shouldn't have a sound of its own? Also, you have to have a point of reference, otherwise, how do you know how a recording is supposed to sound? Here are some things I intentionally listen for when I evaluate a DAC and I suspect many of you listen for these things also: 1. Tone and timbre + timbre variation 2. Air 3. Depth 4. Clarity 5. Image focus 6. Soundstage 7. Delineation of complex details and the ability to layer fine detail and depict subtle nuances 8. Macro and microdynamics 9. Coherence and flow 10. Musicality For each area, I use a specific portion of a track that I know well. I have read some people listen for up to 5 minutes at a time. My memory isn't good enough to do this and so I listen for only 10-20 seconds when doing critical A/B comparisons. For years, I have hosted concerts in my home and I record every session. I know exactly how these tracks should sound and for the past couple of years, I have been using a certain track as my reference for assessing depth. This is a recording of an Ecuadorian Jazz trio comprised of a guitar, stand-up bass, and percussion. When I listen to this track, the DAC should convince me that the guitar, bass, and percussion are located exactly as I experienced it that night. For those that don't have this luxury, one option would be to record an acoustical instrument from 5 feet away, 10 feet away, and 15 feet away and experience for yourself how well 2 different DACs portray this depth. I believe you will be surprised. When assessing timbre, because I play the piano and I know the timbre of the piano better than any other instrument, I use simple piano tracks. As I record pieces that I've played and then play them back in my room, it's easy to compare how well one DAC does versus another. As my piano is in the same room as my speakers, I can easily A/B my own piano against a DAC and while I have yet to hear a completely accurate reproduction from any DAC of the piano, thus far, the DAVE (especially with Blu Mk2) has come the closest. As always, I am not looking for "pleasing," but rather "real" and "accurate." Beyond timbre, there is timbre variation. I have a track of 2 friends who are concert-level violinists. They both play different brands of violins and each one has a slightly different timbre. I want a DAC to be able to tell me there are 2 violins of slightly different timbres playing and the DAVE provides this delineation better than any DAC I've heard. As a further example of this, a while back, I had the privilege of being able to directly compare my DAVE against a Nagra HD DAC (about $30k) and the mighty dCS Vivaldi stack which included the Vivaldi Master Clock and the Vivaldi Upsampler ($68k). Speakers used were a pair of Wilson Sasha 2s driven by Spectral amplification via Transparent cabling. Using a certain reference track that I use frequently (Magnificat, Trondheim Solistene, 24/352.8 PCM, track 10), one observation was very clear. With both the Nagra but especially the Vivaldi, these DACs seemed to coalesce details together. With the Vivaldi especially, it appeared as if all the performers were in perfect sync as all you could hear was this single solid harmonic tone when the mass of violins were playing and while this is pleasant to hear, I found it to be inaccurate. With the DAVE, you could tell many violins were playing at once and since performers can never be in perfect synch, it was easy to hear subtle differences in timing. When I attend the symphony, I routinely hear this and am accepting of it. When I assess for air, I listen to both small and large ensemble. Instruments and voices should sound dimensional and not flat. This is perhaps one of the things I am most sensitive to. Back in 2015, I was invited to participate in Tyll Hertsens' "Big Sound 2015." Many participated including Bob Katz, the legendary mastering engineer (who has mastered several Grammy winning albums) and Tyll himself. As part of the exercise, we were asked to listen to 3 different headphone amplifiers (Bakoon HPA-21, TTJV Teton, and SimAudio 430HA) via a Sennheiser HD800 followed by a HiFiMan HE-1000 headphone. The amps were volume level matched and a 10 second track of a Brazilian vocalist singing in Portuguese was continuously looped. As these amps were not necessarily familiar to any of us, we got a couple minutes to familiarize ourselves with the sound signature of each amp on each headphone. We were then blind-folded and were asked to name which amplifier we were listening to on each headphone. In general, each candidate scored about 50% and this included both Tyll and Bob. In my case, out of 30 blind tests, I missed the first (due to not being familiar with the process) and then the last (due to fatigue). This means I got 28/30 correct (93%). When Tyll asked me what I was hearing that no one else seemed to be hearing, I told him that each amplifier had a different amount of "air" and that was it. I'm not about to suggest I have a better ear than Tyll Hertsens or Bob Katz but when it comes to air around voices and instruments, I am extremely sensitive. I believe we each have our particular sensitivities. https://www.innerfidelity.com/content/big-sound-2015-participant-report-roy-romaz I can go on about how I assess equipment but I think you get the picture. The only time I listen for minutes at a time is at the end when I'm listening for musicality. To be honest, this is the only part that is enjoyable but by the time I get to assessing for musicality, I have a pretty firm grasp of the strengths and weaknesses of a DAC and already know which ones will be musical. What I do isn't the only way to assess a DAC or any piece of equipment but it has been effective for me and it has allowed me to more easily discern even subtle differences that when added up, can be very meaningful. For some criteria, with the right track, you can tell within a few seconds how good a DAC is. For other things like coherence and flow, it can take a long time to figure out that one DAC is better than another. Because speakers are better at imaging and depth, this would be the best way to test these areas. Because headphones are often better at presenting subtle detail, I believe both should be used when possible. One also has to remember that each of these criteria are influenced by every other component in your audio chain and so what you are really trying to do when evaluating something like a DAC is gauging how well that DAC will fit into your system. Yes, as far as I know, no other DAC can drive speakers directly because their impedance is too high. I have been trying to drive headphones directly for years dating back to when I owned a Bricasti M1. Here is a photo: The output impedance of a Bricasti M1 is about 32 ohms. It can drive an HE-1000 ok but it sounds better driving a much higher impedance headphone like the HD800. The TotalDac d1-monobloc that I used to own had a slightly lower output impedance of just under 20 ohms and so this did a better job but these DACs can't drive 8-ohm speakers. This is where Chord is unique. There are a few amps that have very high DR such as your Benchmark but once again, this is not a very meaningful quality beyond a certain point. A DR of 135dB equates to about 22.5 bits of resolution. Most humans can only hear about 21-bits of dynamic resolution. Again, DR is not a measure of how loud your system can play but what the range is between the quietest sound to the very loudest sound and while no one has problems hearing the loudest sound, as we get older, most of us have trouble hearing the very quiet sounds. Also, most ADCs are limited to only 21-bits of DR. If you use Roon, it will tell you the DR of the track you're listening to and very rarely will you hit 120dB of DR. Finally, the only genre of music that has this type of DR is generally large orchestral classical music. Most vocal tracks, especially studio recordings will have a very narrow DR of no more than 60dB. I found the VAC to be better than the Constellation. It provided better textural information. With regards to texture, I felt it was even slightly better than the DAVE but this was with a very mediocre media server. Despite this subtle difference, I didn't think it was worth paying $30k. With regards to speed and dynamics, DAVE was better on its own. With a more resolving music server (like I have now) and especially when paired with Blu Mk2, I find that I am not wanting for any augmentation. Here are some THD + noise measurements of DAVE against several top flight preamps: Chord DAVE: 0.000015% at 2.5V VAC Statement Preamp ($75k): 0.007% at 1V Audio Research Reference 10 ($30k): 0.006% at 2V Pass Labs Xs preamp ($38k): 0.001% at 5V On average, the best preamps have 300x more distortion than DAVE.
  10. Sure enough. However, THD + Noise ratings provided for these amplifiers by the manufacturer are nearly 700x more than DAVE.
  11. My Habst cables are 50cm each. Compared against a much cheaper 30cm cable from Blue Jeans Cables, I found the longer Habst cable still sounded better. The Habst is extremely well shielded and has an extra ground lead. It also utilizes a very pure grade of cryo'd silver. One thing you can try if you don't want to spend a lot of money on clock cables is to apply 7.5mm ferrite cores to your cable and here is a photo of a cable that contains 5 of these ferrite cores. You can buy these ferrite cores on DigiKey for about $30 for 5: I can't take credit for this as the idea was proposed by Rob Watts but it certainly improves sound considerably (much less glare), however, the Habst still sounds better.
  12. There is no consensus on what the best specs are for a music server and there are also different approaches that will take you down opposite paths. If you plan to use DSP or oversample via software (ie HQPlayer), you will need a powerful machine. Those that have embraced this pathway have found that the gains from upsampling have been worth the electrical noise generated by their hardware. I have heard very good implementations of this, the Sound Galleries SGM 2015 being a good example. Because my DAC does my upsampling for me, I am free to build a low power server and there are definite advantages to this including lower cost. Because I am committed to running Windows and Roon, I am bound to Intel X86-type CPUs and not the Atom you used to use. Mark Jenkins, creator of the highly regarded Antipodes servers found that CPUs with simpler architectures like the Celerons sounded better than i3/i5/i7s. I haven't done my own comparisons but based on how good the Antipodes servers sound, I just took Mark's word for it and so my simple motherboard has an embedded Celeron that has a TDP of only 6.5 watts. It can be easily passively cooled (no fans) and at all times, I can place my hand on it and it never feels hot. I can run it 24/7 with no power concerns as my entire server consumes less than 10 watts. In my own testing, I also found that 2GB of RAM sounds better than 4GB which sounds better than 8GB. It was shocking to learn that RAM can draw considerable amounts of current, more than many CPUs in bursts. Fortunately, with Windows Server 2016, 2GB of RAM works just fine. Storage devices generate plenty of electrical noise with SSDs being the worst offender. The noise they generate is in the 6GHz region and very difficult to mitigate even when you power one with a quiet PSU like an LPS-1. Hard drives generate less electrical noise although as they have spinning platters, they cause vibrational disturbances that can be upsetting to sensitive clock boards. SD cards are less noisy with compact flash being quieter still.
  13. Others may have different results but here are mine. I used to own an Entreq Poseidon and a loom of Atlantis grounding cables (I still own a few and if someone wants to buy mine, PM me). They did wonders for my former TotalDac d1-monobloc. When my DAVE first came in, I grounded it with my Entreq Poseidon via DAVE's XLR output (which is always live unless you're listening to headphones) and I heard no improvement at all. I tried it again with a Synergistic Research grounding block and their Hi-Def grounding cable and I heard no improvement. I tried it again with Sound Galleries' new D2 grounding block that I purchased in Munich this past May and again, no difference. Now, with my music server, the D2 grounding block results in a very noticeable improvement.
  14. The thought has certainly crossed my mind to upgrade as many clocks as I can given that the REF10 has 8 clock outputs. With my DAVE, Blu Mk2, and incoming Hugo2, however, I have never considered modifying the clocks on these pieces at all. Maybe there's something to be gained but I have my doubts that there are and I won't be the one to try it. First, Rob's DACs are not your ordinary delta-sigma or R2R DACs. They incorporate a pulse array architecture that he invented years ago and this is what he has to say: "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." "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)." As far as trying to do away with a noisy clock, DAVE's noise and distortion measurements are already as good as it gets and so I'm not sure what there would be to gain. While at RMAF last year, I brought my DAVE with me. As Audio Precision was there offering free measurements of any electronic component people were willing to bring in using their state-of-the-art APx555, I asked to have my DAVE measured and here is a photo: Here are actual measurements of my DAVE: While it's hard to see, hopefully, you can appreciate that the red tracing is roughly the same as the blue tracing and that they are both at the bottom of the y axis. One tracing represents the DAVE's distortion measurements while the other represents the noise floor of the APx555. This means the THD + noise measurement @1 kHz of my DAVE is so low (0.000015% @ 2.5V) that it is just barely measurable by what many consider to be the most sensitive analyzer in existence today. According to the AP analyst that measured my DAVE, the measurements he got were the best he had ever seen for a DAC. As far as DAVE's jitter and noise measurements reported by others, here are John Atkinson's measurements: https://www.stereophile.com/content/chord-electronics-dave-da-processor-measurements In John's own words, DAVE's measured performance are "beyond reproach."
  15. 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.
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