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ecwl

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About ecwl

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  1. Yes. I was anticipating a headphone amp using Hugo TT2 technology and a streamer (though not 2Go) for the desktop now that they have refined their Poly firmware & iOS app. That said, I do like their new amplifiers. They are not the only ones to use "feed-forward feedback" as Hegel, Benchmark seemed to use similar schemes and the Bryston new Cubed amplifiers sound fantastic with their own feedback scheme. I do appreciate how Chord Etude sounds wonderful. I guess for the people who can afford the "cheaper" Ultima amplifiers, good for them?
  2. I think the audibility of jitter is a very challenging discussion if people want a scientific blinded study in support of audibility. At least from what I've read by Rob Watts, the Chord DAC designer, he said when he used to design DAC chips for companies, he usually gets told what specifications are desired, e.g. noise floor and jitter and he would create the design and if the company wants a different target, he would design a different chip. He says there is an audible difference with different levels of jitter but in my mind, it's hard to know what sonic differences for these chips are due to the noise floor and what is due to jitter. Moreover, now that he stops designing DAC chips, he claims that depending on the jitter immunity of his design, he can hear a difference. Now I'm sure he can change his prototype for different levels of jitter but that's not something that we can do. Ultimately, if there is a very low threshold for jitter that is audible, only DAC designers who can tightly control jitter levels can truly A/B or ABX various jitter amounts and see if they are audible. For most of us, if we are to compare various levels of jitter from different recordings, we would be limited by the jitter level of the DAC we are listening through. If we are comparing different DACs with different jitter amounts, our listening experience would be altered by other aspects of the DAC design. I personally believe that very low levels of jitter are audible based on comparison of different DACs but like I just said, I actually don't know whether I'm just hearing differences in noise shaper, noise floor, digital filters, etc. and then attributing the sonic improvements to lower amounts of jitter.
  3. The first time I read that USB cables should be 1.5m was from the Berkeley Audio Design Alpha USB manual: “1.5 meters is a good default length for USB, SPDIF and AES cables.” And I believe some audio reviews also reiterated this and went more into the explanations I’ve always thought the USB part didn’t make much sense but I don’t know enough to really know for sure. Good to hear from a few experts it’s not true. Too bad I already bought a 1.5m cable. I could have used a shorter one.
  4. I suspect ESS chips when fed DSD512 just plays it like a DSD DAC as if it's a 64-element shift register DSD design. So I don't think there's any additional noise shaping involved. But when you're playing a PCM file on an ESS chip, I'm not sure if we can easily know whether it upsamples and noise shapes to 5-6 bits at 64fs, 128fs or even at higher fs. We also don't know whether it upsamples to say 16fs 24-bit first and then to say 128fs 6 bits or whether it'll be a one-step process. But I think whatever the process is, it's almost a little irrelevant. First of all, I am quite certain ESS chips do not upsample from PCM and noise shape directly to DSD512. Moreover, the computational power involved in the ESS chips would be dwarfed by what HQPlayer can do when it upsamples and noise shapes to DSD512. Similarly, if ESS chips upsamples to 16fs first like M-Scaler, the computational power for the upsampling filter with ESS would be dwarfed by what the M-Scaler can do. And the subsequent upsampling or noise shaping to 5-6 bits at say 128fs to be output to the 64 elements SDM would be computationally significantly less intensive than say the Chord Qutest upsampling from 16fs to 256fs and then to 104MHz then noise shaping for playback on the 10 discrete elements of the pulse array DAC. I think the bottomline, whether you're going with a final DAC output of DSD512/DSD1024 or a multi-element discrete SDM design is that the more computational power (and superior algorithm) you can throw at it for upsampling and noise shaping, the better the sonic result. (Unless you believe that all DACs measure the same and all noise/distortions measured are already beyond the threshold of hearing in which case you should stick with your headphone amp/DAC/jack that comes with your cellphone...)
  5. https://chordelectronics.co.uk/product/qutest/ Or more specifically https://chordelectronics.co.uk/wp-content/uploads/2017/01/Windows-10-768KHz-driver.zip
  6. I did try to convert DSD to 705.6 using JRiver or Roon in the past. With DAVE in PCM+ mode, I concur with you that the DSD to 705.6 conversion, particularly with Roon sounds better than DAVE’s own decimation from DSD to 705.6 which presumably uses the same filter as Mojo. But DAVE in DSD+ mode sounds better with the DSD file than with 705.6 from Roon/JRiver. With Blu2 playing DSD directly (so same DSD to 705.6 conversion as Qutest), the Blu2 conversion clearly sounds better than Roon doing the DSD to 705.6 conversion. Obviously that doesn’t preclude the possibility that there are better DSD to 705.6 conversion algorithm with other software. But I don’t listen to recorded DSD tracks much so I rarely bother to compare or seek different software for this purpose
  7. The Qutest 49000 taps only refer to the filter that takes you 44kHz to 705kHz. So if you use HQPlayer to upsample to 705kHz, you’re getting the HQPlayer filter rather than the Qutest filter. That said, Qutest would still take the 705kHz and upsample to 11.3MHz using either the WTA or FIR filter depending on which filter color you use (white/green WTA is optimal whereas red/orange FIR is less optimal) and then it’ll upsample again to 104MHz to the pulse array DAC for playback. And as @barrows said, you should hear a difference between the native Qutest 49000 taps WTA filter vs HQPlayer’s filter. You’ll have to decide for yourself which one you prefer.
  8. My understanding is that all modern Chord DACs upsample to 104MHz 5-bit before sending the signal to the pulse array DAC to be output at 104MHz (at an obviously lower bit-rate through additional noise shaping) Hugo2/TT2/Qutest/Blu2/M-Scaler all take DSD signals and convert it to 705.6kHz, 24-bit??? before being upsampled to 11.3MHz using another WTA filter and the being upsampled to 104MHz 5-bit. Chord DAVE DSD+ mode takes the DSD signal 2.8/5.6/11.3MHz and converts it to 2.8/5.6/11.3MHz, 24-bit???, does some sort of filtering on the signal and to 11.3MHz and then to 104MHz 5-bit Rob Watts the Chord DAC designer discontinued the DSD+ mode for Hugo2/TT2/Qutest/M-Scaler/Blu2 because he finds that with an improved decimation filter to convert DSD signals to 705.6kHz, 24-bit???, he finds the DSD files to sound better (which i can attest to based on my testing comparing DAVE DSD+ mode vs Blu2 DSD playback). So the official Rob Watts recommendation is that if you want to listen to DSD files on Chord Qutest, you should just send the file directly to Qutest to get the best sound. Obviously, people can manipulate the file in whatever way they want for playback.
  9. Having listened to Hugo 2 vs Mojo and having a friend own 2Qute, I would say if you can afford it, go for Qutest. There are so many technical improvements from Qutest that the sound quality is far superior. Here are the main ones that I think matter the most. Qutest has 49000 taps for 16fs upsampling compared to 2Qute 28000 8fs followed by simple 2x FIR upsampling to get to 16fs. Qutest then has an additional WTA filter to get to 256fs whereas 2Qute uses a very simple FIR filter and this contributes a lot to sound quality. Qutest has 10 elements and a 11th? Order noise shaper whereas 2Qute has 4 elements with a 5th order noise shaper. Qutest can output 1, 2, 3V so it won’t clip your preamp whereas 2Qute only puts out 3V.
  10. So nobody in my local head-fi meet upsamples from PCM to DSD512/DSD256. We were all listening straight off the DAC. But more to the issue of noise floor modulation, it is definitely not something that many DAC designers talk about. And there seems to be a lot of people saying it doesn't happen or that when they upsample to DSD512 it solves the problem. Truth is, noise floor modulation can happen at multiple stages. First, it's in the recording stage so it's possible for the original recording to have noise floor modulation, making it sound harsher. Second, it's in the digital playback stage so it's theoretically true that any DAC that takes say 44/16 and upsamples to DSD512 or 64-element 5MHz (Sabre DAC chip I think) can introduce noise floor modulation during the upsampling but I think that's actually exceedingly rare for any competent upsampling algorithm. The last stage of noise floor modulation that can be introduced is at the actual Digital-to-Analog conversion. According to the designer Rob Watts, this happens with virtually all DAC designs for a variety of reasons, ranging from RF noise, jitter from the clock but most importantly, the DAC design itself, be it DSD/R2R/multi-bit PWM/DSD, including most modern DAC chips. The issue is that every time your DAC switches from 0's to 1's, or in the case or R2R DACs 256 to 16 or 18 to 238, there is noise that's generated in the rest of the circuit. Moreover, simple switching from 0 to 1 generates a different noise than switching from 1 to 0. The example Rob Watts give is that let's say you have silence and in DSD256, you can encode it with 101010101010101010101010101010101010 or 11001100110011001100110011001100. Technically they are equivalent int he digital domain but the first sequence would involve 32 switches and the other would involve 16 switches so when you're dealing with an actual DAC circuit, the amount of noise generated by these two sequences would be different. Moreover, the frequency of switching would also be different, generating a different noise pattern, despite being "silent". This is the reason why with loud music, the noise floor tends to be higher and with quieter music, the noise floor tends to be lower, leading to noise floor modulation. In addition, specific to DAC chips, because all the DAC chip elements are so close to each other on silicon, they are much more prone to noise. This is the reason why a lot of DAC designers have moved to discrete elements, ranging from DSD DACs to multibit DACs, like dCS. The solution Chord uses in the Pulse Array DAC is that using the 10-element in Qutest to represent say signal levels of 1-9 (through the noise shaper), the DAC is constantly switching at 104 MHz by flipping either two elements at the same time to maintain the same signal level (switching 0 to 1 in 1 element or 1 to 0 in another element) or only 1 element switches (either 0 to 1 or 1 to 0) to change the signal level (from say 4 to 5, or 7 to 6). As a result, the noise level fluctuation is constant regardless of the loudness of music you're playing. So in answer to your question at the beginning, no I did not get to listen for noise floor modulation off the ADI-2 with DSD256 or DSD512. But the bigger issue is that by design, ADI-2, as with other DAC chips, would have noise floor modulation regardless of what signals you feed it.
  11. I listened to my Chord Mojo, my Chord DAVE (no Blu2) and the RMI ADI-2 in the same session off headphones. I have never listened to Qutest. Closest was Hugo 2
  12. I have listened to the ADI-2 at my local head-if meet. I think it’s an excellent DAC chip based DAC. I personally prefer Mojo to the ADI-2 but I think people are so used to listening to DAC chip based DACs that I can see they might prefer ADI-2. For me, the “advantage” of ADI-2 is the lower noise floor which creates this extra sense of transparency. This is a result of Mojo’s reference power, noise shaper and the 4-element pulse array DAC. The ADI-2 problem compared to Mojo is the noise floor modulation which makes music sound digital/bright/harsh so smooth, warm vocals or instrumental sounds often sound extra exciting with ADI-2 whereas they sound more natural with Mojo. There is also the transient timing issue so when you hear finger snapping or drum strikes, Mojo is more realistic but ADI-2 still sounds dynamic because even though the strikes are less precise, the noise floor modulation makes it sound more exciting. And by natural and realistic, I mean compared to live unamplified performances. I have not heard Qutest in any system yet but I’ve listened to Hugo 2 via headphones at a stereo store and it far surpasses Mojo and Hugo and 2Qute in everywhere. Hugo 2 is much closer to DAVE than Mojo. So I suspect Qutest would have the same transparency as ADI-2 and all the advantages of Chord DACs. Ultimately though, when we are spending that much money on ADI-2 or Qutest, you have to like the sound and how it integrates into your system. I’ve seen people swear they prefer their R2R or DSD DACs. And DAC chip based DACs are so ubiquitous (Sonos, iphones, just basically everywhere), I think people are so used to hearing their favorite music with imprecise transients and noise floor modulation from DAC chip based DACs, they are frequently preferred over other types of DAC designs. I think the challenge is that we rarely listen to live unamplified music so we forget what natural realistic music sounds like and the DAC chip based DAC sound is now our reference instead of live music. I definitely think anyone thinking about purchasing a DAC in Qutest price range should audition the Qutest though. If it’s not your cup of tea, don’t buy it.
  13. Haha... I just flipped through this thread and realized despite all the technical discussions, only @romaz and myself actually own M-scalers in the form of Blu2 and DAVE (and we have both tried HQPlayer) and we barely talked about our subjective sound quality impressions of M-Scaler. And @romaz and I basically owned both Chord products around the time when they came out. The best way to describe M-Scaler over DAVE is to describe DAVE over other DACs you might have had before. I don't know if you've noticed DAVE, with some instruments or vocals, the timbre of the instruments and the voice just sound more realistic than what you're used to with other DACs. Or when you're listening to percussion instruments, going from guitar plucks to finger snapping or hand clapping all the way to strikes of drums, DAVE sounds more realistic and dynamic compared to most DACs you know. Well, whatever DAVE to the other DACs is M-Scaler to DAVE. I almost never listen to DAVE alone but ironically in the past two days I did. My dealer had a new setup so I dropped by to listen to some unfamiliar pieces through his DAVE (direct to PS Audio BHK 300 mono amps) and things sound great except the instruments just didn't sound as realistic as I'm used to so when I got home, I listened to the same songs through Blu2 and the transients and timbre of instruments were at another level that I really enjoyed. Because I disconnected my video system from my Blu2, I decided to watch TV using DAVE alone (instead of the 0.6M taps of M-Scaler) and within 2 minutes, I just didn't appreciate how the sounds of explosions or people fighting didn't seem quite right for me, in addition to people's voices not sounding as realistic as I'm used to. Plugging Blu2 back into the video system had a nice improvement. I would say that if you own DAVE and can afford it, M-Scaler is a no-brainer. The only caveat is that there is a possibility that your downstream devices after the DAVE, e.g. preamplifier (if you have one) or amplifier might be a limiting factor in hearing all the additional improvements. I think that's why the Head-Fi forums tend to rave about Chord products more because there are fewer intermediate factors that can reduce the performance of the DACs or cause synergy issues. Anyway, that's my personal subjective (and obviously biased) opinion. Hope it's helpful to you.
  14. @The Computer Audiophile, I thought the project is already complete and we are just eagerly waiting for you to write up the next part? Or is nothing set in stone yet? I was always under the impression that the most important part of setting up the room is getting the most even bass response which you can use your UMIK1 to do by moving it around to find the optimal seating position. I noticed you may not have a lot of room to move the seat a few feet forwards or backwards. And then if the best seating position is useable, then speaker placement and toeing in comes next followed by acoustic treatment. Mainly because acoustic treatments are unable to address low-bass issues in a room. I used to use room ratio calculators but I’ve found there are always nuances of a house/room that those calculators don’t capture so it’s just easier to play pink noise with speakers and move the microphone around. Or maybe you’re going to talk about all that? I just got a little confused by Mitchco’s comments.
  15. Cool. @The Computer Audiophile, care to share the UMIK-1 REW measurement of the new room pre-acoustic treatments? Or is that reserved for part 2?
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