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mitchco

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About mitchco

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  1. Here is one objective test comparing "pro" AD DA converters using a repeatable procedure, that folks can run themselves if they wish: https://www.gearslutz.com/board/showpost.php?p=14093948&postcount=1776 While the Lynx Hilo is on top (it is amazingly transparent!), any one of the converters listed in this thread will work just fine. The simple reality is that LP's typically have 10 to 12 bits of resolution and even the most mint, first time use from top mastering labs like Analogue Productions may make it to 14 bits of resolution (max). Wrt bandwidth. While there may be some output in the 20 to 24 kHz range on an LP and in some rarefied cases higher, the real limitation is the source. Meaning microphones, mixing consoles, effects boxes, tape machines, tape heads, tape electronics, tape itself, compressors/limiters, mastering lab amps, etc, and on it goes. Not to mention most speakers , when measured, don't have significant output beyond 20 kHz anyway. Most studios then and now, still consider 20 Hz to 20 kHz as the standard bandwidth. I would consider 24/192 kHz to already be overkill. 24/96 kHz is more than enough and even 24/48 kHz is likely to be just fine. Source: ex recording/mixing engineer that spent +10,000 hours in the chair. Personally, I would grab one of the higher ranking AD DA converters in the list above plus a copy of Vinyl Studio and start converting and then enjoying the music!
  2. Hi @aps no, I would always recommend a high pass or better yet, a linear phase digital XO between sub(s) and mains. A linear phase XO sums properly both in the frequency and time domain. So yes, with a multi-channel soundcard or DAC, AL or Acourate digital XO would be used to manage low/high pass to subs and Kii's. Similar to what is done in this article: https://audiophilestyle.com/ca/ca-academy/ integrating-subwoofers-with-stereo-mains-using-audiolense-r712/ The trick is to find the best XO point and this is where the room comes into play. In my case linked above, my mains have good output to 40 Hz or a bit below (i.e. the -3 dB point in the room). Using tools like https://amcoustics.com/tools/amroc I have a large room mode, right at 40 Hz. So to offset that, I chose 45 Hz as being inbetween room modes for my XO point. In the case of the Kii's and @TheStupidOne setup, is using additional woofers to supplement the Kii's. Turned out good! There are no hard and fast rules. However, if using a traditional "sub(s)" with the Kii's, I would take Kii's advice (and mine) of finding the -3 dB point in-room of the mains (whatever they are) and cross there.
  3. Good reccos from @SJK. Here are a couple of additional approaches here on AS: The Lynx Hilo I used for those articles can be found used for a good price. It has an outstandingly transparent ADC. I still use it 🙂 Good luck!
  4. @TheStupidOne Excellent! I bet it does sound great! 😉 Ah, I missed this, I get what you are doing now. Are you using TTD in Audiolense? If you are still up for an experiment, pick either the 80 Hz or 200 Hz XO and like in the graphs above, at the same SPL, move the mic from the LP a foot closer (or 2 or 3 ft whatever) towards the side you are measuring and take another measurement. You may want to take 2 to 4 measurements towards the speakers before you are like 3 feet in front of the Kii and sub. If the shape of the distortion curve stays the same, but simply goes up in level, one can conclude you are measuring the speakers (and microphones) distortion. If the distortion changes with each mic position, then it is the room (sometimes combo standing waves and noise floor). Have you plotted the rooms noise floor? Also, have a quick look at what JohnM wrote and check out the link to the help file: https://www.avnirvana.com/threads/noise-floor-seems-too-high.1732/#post-15970 Congrats again. good job!
  5. Hi Paul, ah, thanks. Yes, Solderdude - good guy for sure. @AudezeLLC thanks for that. I used a blocked in ear approach which is not HATS and using a set of binaural mic's in my own ears that are quite flat, hence the reason why I use this approach and seems to better represent what the actual response of the headphones would be up to about 10 kHz. After 10 kHz, all bets are off due to different shaped pinna's. It is too bad that the headphone industry still does not have a defacto way of measuring headphones that can be related to or deviation from a flat response. In other words, using the approach at solderdude's site, what is flat and how does one compare? And then what is the deviation from flat? Some sort of normalisation needs to be applied to the raw data or the approach taken if it is not the blocked in ear approach to help relate. Maybe solderdude has already done that, but I have no way of knowing what the deviation from flat is looking at his measurements in this thread. It is figured out in the loudspeaker industry with the free download of ANSI/CEA-2034-A Standard Method of Measurement for In-Home Loudspeakers. It can even predict the in-room response in a typical living room with a high degree of accuracy. Something like that needs to be developed for the headphone industry. Anyway, just a thought and the offer still stands.
  6. @audiobomber thanks for that. Hmmm, something does not seem right. There is a good 13 to 14 dB drop between 1 kHz to 3 kHz and covers across 3 kHz to 7 kHz. Relatively speaking, a 10 dB increase or drop means that frequency range is perceived as twice as loud or twice as quiet compared to the frequency range next to it. That's a lot. On Audiophile Style, I measured the NAD Viso HP50: Much flatter response. My results were consistent with Tyll's and the Harman target curve they were modelled after. Link to my full review: I am wondering if there was an issue with the measurement rig over at DIY Audio Heaven...? I get excellent results with my setup using SoundProfessional's top of the line in-ear binaural mics. @The Computer Audiophile and @AudezeLLC without stepping on anyone's toes, and if you feel it is worthwhile, I would be happy to measure a pair. Kind regards, Mitch
  7. How are you measuring distortion? At the listening position? If so, then unfortunately, the room dominates below Schroeder and likely not getting real distortion numbers... Distortion of speakers needs to be measured at 1 meter distance and ideally outdoors using a ground plane measurement. Also, does the mic come with a distortion spec? There are only a handful of mics that spec distortion. Earthworks, DPA, B&K and http://www.isemcon.com/datasheets/EMX7150-US-r04.pdf for example. Most measurement mics are good at measuring frequency response, but not so good measuring distortion (including the mic I am using right now). The Kii's have fantastic directivity down to 80 Hz and designed to avoid SBIR and it is worth the time spending with the measurement mic to take multiple sweeps while adjusting boundary eq to get the smoothest response from the THREE's first in your room (maybe you have already done that). Then bring in the subs. Yah, it could be that the -3 dB point in your room is 30 Hz, but subs are supposed to, you know... sub 🙂 If you still want to offload, then I would start at 80 Hz and work your way down from there...
  8. If that is the case, and you are happy with the SQ, I would simply leave well enough alone. If you wanted to experiment, I would lower lower the XO to the subs to 80 Hz or below. The suggestion from Kii is a good one. The subs should augment the Kii's and not replace their cardioid response I would find the -3 dB point at the LP with just the Kii's and then use that point as the XO point to the subs. That way you are maximizing the response of both Kii's and subs. Fyi, that' what I did with my rig (not Kii's but finding the -3 dB point on my mains) and XO to my subs are at 45 Hz.
  9. Hello, @TheStupidOne perhaps a little diagram or words to describe your end to end signal path with type of connections. If using computer with Audiolense, Audiolense can also be used for digital XO duties as well (highly recommended). What convolution engine are you using? The issue with OpenDRC and the like, is the limited number of FIR filter taps. Typically around 6000 taps per channel, whereas on the PC, 65,536 taps (or even double) are available and required if you want deep bass room correction. Have you asked Bernt if Audiolense will take the 90ms of delay into consideration if using Audiolense XO with subs? Also, curious about the 200 Hz XO point. The Kii THREE's cardioid capabilities go right down to 80 Hz, which would seem more of natural XO point for the subs, so you can still take advantage of the cardioid response of the THREE's...
  10. @mbabst Hi Martin, welcome to the forum! I had a quick look, but nothing seemed to jump out at me. I have not tried this specific approach myself, but there are a lot of steps. Unfortunately, I do not have the time to trace through each one. When I was using another DSP package to time align some drivers, it took me a good three or four tries before I could repeat my results. Unless @SwissBear can point to something specific, I would tend to try the steps over again and see if you end up in the same place or not. There is also a thread on diyAudio that may have some insight: https://www.diyaudio.com/forums/multi-way/221434-rephase-loudspeaker-phase-linearization-eq-fir-filtering-tool-195.html Sorry I could not be more help. Mitch
  11. A partial list here: https://audiophilestyle.com/ca/reviews/nad-viso-hp50-with-roomfeel-headphone-review-r720/#subjective
  12. Great question! I don't have an answer as I liked both speakers a great deal and could easily live with both. One part of me believes the magic Bruno has achieved as the most neutral speaker out there. i.e. no colour at all - truly neutral. The other part of me is trying to reconcile if I like that sound or not. I think I do. It reminds me of Rythmik subs and their Direct Servo technology. Very understated and dry sounding as their distortion measurements confirm on https://www.data-bass.com versus other subs. but that is what I have in my system today. What can I say 🙂
  13. Dryer sound - less distortion is my best guess, given the electronics quote. That's what is fundamentally different compared to the 8c. "According to Putzeys, distortion of loudspeaker drivers can be optimized by varying the amplifier’s output resistance for different frequency ranges. This requires an amplifier that is specifically tuned to the parameters of the loudspeaker driver. An amplifier like this can never be a commercial good-for-all piece of equipment. They have to be tailor made, and this is what was done for the Kii Three." From Audioxpress's excellent review
  14. Nothing in the REW files as I am talking about the direct sound. I can only attribute the difference to the overall electronic design and perhaps, "unique to the implementation used in the Kii THREE is a combined voltage/current control loop that goes beyond merely a better amp – it actively improves the distortion performance of the drive units which contributes significantly to the extreme resolution of the speaker." Don't know exactly what that means 🙂, but one would need an anechoic chamber and associated mic, preamp, ADC that has low enough distortion to be able to measure properly to confirm.
  15. Here is the frequency response of both the Kii THREE and the D&D 8c overlaid as measured in my room. Both were tuned to produce a downward tilting response from 20 Hz to -10 dB at 20 kHz as per Harman's research. This was achieved using the onboard controls only. The 8c is a bit smoother in the bottom end due to the additional PEQ's that the THREE does not have, but it in a double blind comparison, I doubt I could pick the two apart from a tonal perspective. As others have mentioned, the THREE sounds "dryer" than the 8c is about the only way I can describe any difference. The 8c's treble required next to no adjustment to match Harman's research. However, the THREE out of the box required a -4 dB shelf at 3 kHz to match. So, out of the box, without any adjustments, the Kii's may sound overly bright. Of course, both speakers fr can be made ruler flat using room eq like Audiolense or Acourate. Personally, I would just use partial correction from 500 Hz and below, as both have a wonderfully smooth response above 500 Hz. The cardioid design of both speakers really make a difference in the low end response where you don't get the crazy 25 dB of total up and down swing due to room modes, like in my room. So to get that smooth low end response is a real testament on how well both speakers deal with room modes with their cardioid design and on-board boundary controls. Given it's cardioid design and pretty smooth low end response, it will be interesting to see if any kind of FIR based room eq makes it's way into these products. The limiting factor is the on-board computers with the DSP chips that limit the number of FIR filter taps it takes to smooth out the low frequencies over time. Most hardware based DSP is limited to around 6000 taps per channel, whereas on a PC with Audiolense or Acourate one can use 65,536 or even 131,072 taps per channel for total low end control over time.
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