Jump to content

Langston

Regulars
  • Posts

    148
  • Joined

  • Last visited

Everything posted by Langston

  1. Exactly - and it makes sense to use the 12V DC output of the Oppo as well to avoid messing with 120V, but either will work assuming the 120V controlling device is a switched outlet (as your amp has per your first post). The relay's sense input works exactly like the 12V trigger system on audio components: it keeps stuff on as long as voltage is present on the sense input and turns off the outlets when that voltage drops below 3V DC or 12V AC. BTW, another (less reliable, but it might work) way to to do this is to use a a power strip that has a "master" receptacle and several "slave" receptacles. If the device on the master receptacle pulls enough current during turn-on and then stops pulling current at turn-off, it'll work. The problem is that many audio components either don't pull enough current at turn-on, or continue to use some current when turned off (standby mode), thus it's hit or miss with these things. God bless you and your precious family - Langston Edit: you'll need something like this to tap off your Oppo's 12V trigger.
  2. My amps and DAC's use a 12V DC trigger circuit that turns them all on or off when operating a single device. Typical and quite convenient. My processor is a professional unit like yours with a rear On/Off switch and I have another gizmo as well that has to be on for the system to work. I found a great solution that ties them into the trigger circuit for $30. This grounded 120V relay has a trigger input that accepts 3V-48V DC and 12V-120V AC, thus you can use your switched outlet to turn On/Off your processor. The relay has (2) outlets that turn stuff on, (1) outlet that turns stuff off (such as lights when you turn on your home theater) and a 4th always on outlet. God bless you and your precious family - Langston
  3. Nice and simple. Haven't seen that before. : ) Another thought is to make sure the big tubes that consume the largest heater currents are closest to the transformer (source) in the heater supply circuit, followed by the lesser current users toward the end of the circuit. This minimizes a potential noise source across the small tubes. The "humdinger" adjustment pot you referred to helped a lot with the Fender Twin, especially after increasing the twist rate of the downstream heater circuit wires. It's been awhile since I've been inside tube amps. Most of my experience was with guitar amps I rented out as backline in concert work. Crazy simple and amazingly reliable stuff. God bless you and your precious family - Langston
  4. Dimensions: 40 miles width 78,000 feet (14.8 miles) height 400 modules identical to the above Casing: Solid Krell Metal 26 inches thick Warranty: Self maintaining Power Supply: 9,200 Thermonuclear reactors Remote control: THE SUBCONSCIOUS ID!! Monitors: Analog meters as God intended! Be warned or die!! As you were - Langston
  5. 1969 August in Bethel (House of God in Hebrew) NY. 400,000 beautiful whippersnappers.. \\ \\ God bless you and your precious family - Langston --- Pictures Woodstock aerial 1 Woodstock aerial 2 Macintosh MI350 Mono amplifier used for the event Macintosh MI350 internal Macintosh MI350 racks Macintosh MI350 racks with engineer Arlo Guthrie (right) Woodstock couple Woodstock attendees with lighting towers Woodstock attendees with haze Woodstock attendees in color Woodstock aerial 3 Woodstock back stage Group: "Back to the Garden" Bill Hanley mixing the event (the guy I wanted to be) Monitor world: console, mixer, limiter Grace Slick Gurus Janis pouring a drink Jefferson Airplane Jimi Joe Cocker Lighting tower Max and Miriam Yasgur - the land owners Santana
  6. This post, if understood, will save you many thousands of dollars and untold numbers of hours chasing audio nirvana with loudspeaker formats that are physically unable to achieve it. Spoiler: that includes every loudspeaker format other than MEH. Simply put, it is impossible to fix physical driver displacements with DSP. DSP can help immensely, but there is a domain that cannot be entered apart from a single acoustic wavefront containing the entire frequency spectrum of interest. Just for fun: All the best designers know this, but Tom Danley threw his whole audio career at the mercy of it, and for a long while it looked like the market was going to reject it. My audio background until recently was concert production and you still can't get a promoter at a major event to use the superior sounding Danley Jericho series in place of line arrays. But Tom finally found a great friend that threw in with him (Mike Hedden) and they still keep expanding the Jericho line (concert level MEH loudspeakers). They are still waiting for the market to catch up, but in the meantime installations (churches, performing arts facilities, etc.) and home audio have caught up and Danley Labs is doing just fine, thank you. Here's a picture from an event I did with my line array system where Danley offered me their Jericho line FOR FREE if the promoter would accept it. Danley was going to deliver them to the site, help with setup, and take 'em back after the week long festival. I offered the promoter a significant discount. No, no, no. I pleaded with the guy and he knew that I knew more than any other company involved with the event. For fun, here's a couple of pictures:
  7. Claude! I thought maybe I'd lost you forever! OT, but you are probably the most encouraging person on this forum. As I catch up on things around here I've noticed that you make a habit of complementing everyone throughout your years of involvement and I feel bad about this semi-Class D smackdown. OTOH, it's hilarious that there's another much larger thread about "Class D amps are cool" right next to this one! What a wonderful world! : ) You will have to put up with me in your home when I visit my son in Chicago again. I want to hear that system. And of course make fun of the amps (after I steal the K-402's). : ) The paper that @Edgar referenced is amazing and relevant. I was too busy with concert production to pay much attention to SACD when it appeared, but kept hearing that there was more to the story than the marketers were publishing. Lipshitz and Vanderkooy have always been entertaining - my favorites were their arguments vs. Heyser about group delay and arrival time of drivers in multiway systems. The exclamation marks are the best. What's also fascinating at the end of the paper is the tight correlation between their SACD criticisms and those I'm learning about Class D. A final point that Chris made about production quality vs. format shouldn't be missed. Formats intended for the audiophile community such as SACD and higher than CD multi-bit recordings almost always bring more effort to the recording and mastering of the final product. I noticed this the first time on my laptop just clicking on the song clips available on HDTracks while switching resolutions. The higher resolutions on the same album often sounded much better on my dumb laptop speakers - that cannot have anything to do with bit rate and digital resolution - they're either dumbing down the 44.1k previews or the high res. stuff is mastered differently. God bless you and your precious family - Langston
  8. Online reviews tend to all be "oh, this is great" about everything, so I went with a Blu-Ray drive from a trustworthy company (Samsung, model SE-506) and it worked very well - also useful for movies once in a while. This was several years ago and it's still working perfectly, but I'm sure this model has been superseded. Edit: just found this - OWC (the seller) is a long-time favorite of mine and this is what I'd buy for CD ripping if I were looking at a pile of 'em.
  9. I converted hundred's of CD's with dBpoweramp. I can't imagine anything else that could be as painless and accurate. You might want to purchase an external USB optical drive if the one in your computer is pretty old. God bless you and your precious family - Langston
  10. One more thing I thought would be interesting is to use the scope to measure the ultrasonic output of the NC400 at the end of a 25ft Canare 4S11 cable (14AWG quad) while connected to a biamped Khorn. There are no passive components, just the Crites cast-frame woofer and the DE750 compression driver. I made (3) measurements, none of which had signal applied to the amp. The first was with a 4Ω resistor across the output without the loudspeaker cable, the second had the scope probes on the woofer input terminal block, the third had the probes on the input terminals of the DE750. The take-away is that there's little difference between these measurements, thus the ultrasonic noise from the amp is very low impedance and the loudspeaker cable capacitance and driver loads did nothing to attenuate the noise. Nevertheless, both passbands are in the hundreds of ohms at these ultrasonic frequencies, thus almost no ultrasonic current is passing through the voice coils. (Approx. 0.4V at 400kHz fundamental switching frequency; current = volts/Ω). 4Ω Load Khorn Woofer DE750 God bless you and your precious family - Langston
  11. Hi Dave - I have the same issue in Safari. This doesn't address that issue, but it's something you should try if you haven't yet - scroll all the way to the bottom and click on "Theme" and try 'em out. : )
  12. The new column format where the summary statistics are on the top right of more developed threads makes the posts too narrow and results in the need to open many more of the graphic uploads to see their detail. I hope this can be changed, automatically as you scroll down past the summary stats, or by user selection. Thanks.
  13. After several days playing with DeltaWave, my opinion has changed. It's an excellent effort to do what may be impossible (quantifying perception), but just like other forms of measurement, it can help in this regard if you figure it out. I can't imagine how many hundreds of hours the author put into it only to release it for free. Audio has some amazing people in it. : ) The following plots are called Delta Spectrograms, which are 3 dimension difference plots of the frequency domain (amplitude, frequency, time) of the reference recording (direct output of the DAC3 converter) vs. the comparison recording (output of the AHB2 or NC400). If I were to use the same file for the reference and comparison, the plots would be solid green (no difference). This is just for fun, but it continues to imply that there is more difference between the NC400 and the original than the AHB2 and the original. AHB2 vs. Original NC400 vs. Original God bless you and your precious family - Langston
  14. It's a Q-SYS processor. : ) As you know - anything you can imagine pretty much. : ) I got the Linux based 48k/24bit 250i Core (the top 2 space unit in the pictures) in 2012 (!) as a digital playground for my ideas in concert production. I had been using a Peavey Media Matrix processor prior to that for that purpose, but QSC bought almost all the Media Matrix engineers. If you've ever worked with the Media Matrix line that preceded Q-SYS, the programming and software component interfaces are obviously authored by the same people. QSC pretty much unleashed those engineers for a while and they achieved what is still sine qua non of the realm. Now Q-SYS development is focused on video and remote meeting integration which reminds me a bit of my beloved Mac computers morphing into cheesy web surfing iPhone extensions. I hope QSC can resist the Borg.. Oddly, my old 250i is still current - the I/O cards haven't changed and the software/firmware updates are still fully supported. QSC's tech support and service on these things is also without equal. I still have free 24/7 support if needed, which it isn't. That's for installers that learn just enough about a product to sell it, then find out the customer really needs it to work correctly for an event the next morning. On the ABX, which ceased production in 2004, I left a voicemail with QSC last night asking for a schematic (don't need it, just want it), and a tech guy I've dealt with for over a decade there replied this morning offering his help to solve any problems I had, but said he couldn't give me the schematic unless I was an authorized QSC service center. About a minute ago I got another reply from him with details on the audio path I was interested in. /end QSC fanboy brag/ In my home setup, I use the Q-SYS in the digital domain only, it's analog A/D/A is very good, but I chose to output digital from my Mac mini Roon music server via the miniDSP UDIO-8 into the Q-SYS, then from the Q-SYS to Benchmark DAC3 converters. I separate processing logic into pages to keep things simple and pretty. Doesn't cost any processing cycles. Clicking on the Klipschorn DE750 page reveals its processing modules. I opened (2) of them to show you something you may not be aware of. Not only can you upload inverse FIR filters to convolve with the audio stream, but you can skip the zero latency minimum phase (Proportional Q) 32 filter parametric module and use the custom IIR module that takes up to 32 biquad sets. Biquads allow you to do anything you want with an IIR filter, vs. the constraint imposed by the standard parametric filters (not that it's much of a constraint, they are actually forcing you not to do stupid stuff by limiting things). Of course you can daisy chain components to increase filter counts, but that would mean you don't know what you're doing. Fewer is better. Do it Chris's way, make everything as perfect as possible physically first, acoustically second, then use this stuff as icing on the cake. If you want the best result, that's the way to pursue it. Here the transfer function of the FIR (for the full loudspeaker system) and IIR filters (for the HF system) are shown: The same thing, just impulse response view. Notice I only needed about 6ms delay with the FIR to correct the system-wide time domain: God bless you and your precious family - Langston
  15. If you've done a double blind thing before you know why I didn't - it's just a tad less painful than waterboarding. Maybe. I much prefer saturating myself with a few tracks over the course of a half hour or so, then switching amps. The change is immediately apparent usually. I looked into the apparently unavailable VanAlstine ABX unit and it doesn't appear nearly as well executed as the QSC. Whatever the case, I cleaned and calibrated my unit today and it measures like it's supposed to - like a wire. Switching with the IR remote is silent and the audio path is DC coupled and completely passive. It accepts balanced and unbalanced line level devices and ground referenced or bridged amps. It has an internal 1kHz 1V generator that is used to make both channels of both amps the same volume. The 3 digit numerical display shows the voltage reading after it passes through the amp channel. I used a balanced switched resistor passive attenuator to bring down the NC400's within 0.29dB of the AHB2. You're supposed to go through 25 ABX trials before giving up, but I only made it to 11 tonight. Sometimes I finished a trial within seconds, sometimes it took several grueling minutes. I got 8 right* and I'm sure I can get in the 90's tomorrow when I'm not tired. For me, a solo female soprano is usually the best amp separator. God bless you and your precious family - Langston * Edit on 27 Oct 2021: I just figured out the probability math on this and you divide the possible "combinations" of getting 8 right (r) out of 11 trials (n) "without repetition" by the compound probability of two possible outcomes (right or wrong choice) over 11 trials. This latter number is the denominator and in this case is simply 2^11 = 2,048. The numerator requires a tiny bit of statistics math that can be done here and works out to be 165. Then the probability or level of confidence that I in fact heard a difference between the two amps when selecting 8 right out of 11 ABX trials = 1 - 165/2048 = 92%. Pretty good, but the rule of thumb the stats folks use is 95% or better to be sure enough to start bragging. : ) ABX testing requires a bunch of wiring Classic excellence
  16. Hey Claude! I thought a little salt for the wound would be medicinal. : ) I got another 6dB of S/N out of the NC400 to equal the AHB2 and Purifi amps by removing (Rg) from the instrumentation amp section. This means the NC400 now shares best in class noise specs with the AHB2 while having superior CMRR compared to the Purifi amp. The NC400's now have 12.37dB gain and clip at 8V (9.3V at 8Ω)* instead of 2V. Got the padding I wanted and cut the noise in half as a bonus. * Line level consumer audio devices typically clip from 2V to 5V, line level pro audio gear typically clips at a minimum of 12V. There's a reason this amp came with 26dB of gain. The schematic as a review The location of (Rg) labeled R141 on the amp module NC400 modified S/N NC400 modified noise God bless you and your precious family - Langston
  17. DeltaWave does what DiffMaker does and a whole lot more. It showed the AHB2 modifying the audio much less than the NC400, but it showed most of that difference below 500Hz and above 20kHz (we're dealing with 96k/24bit files). Where I heard differences was in the midrange where this software said both amps were nearly the same. DeltaWave does all kinds of cool things like matching time, frequency and level, then plotting the results of over a dozen comparisons between the files and then allowing you to listen to the difference as well as the reference and file under test. It even has an ABX facility, but I much prefer hardware solutions for that kind of thing to eliminate the computer from the loop. Obviously this kind of software is yet another attempt to quantify perception, yet requires subjective interpretation to do that. It's not useless, and it is best of breed, but another miss just like DiffMaker. IMO. : ) I did listen to the files I made available to download as if I were one of you and the difference is there to hear. Not quite as clear as the original playback, but good enough if you have a very good playback system. God bless you and your precious family - Langston
  18. Subjective measurement?! You surely don't think that interests me?? I have one. Works great. We should do a meeting somewhere and embarrass ourselves with it! : ) God bless you and your precious family - Langston
  19. And that is what matters. Measurement is my single greatest instructor that began in earnest to prevent and predict failures in my concert work and I never had a show go down or an installation fail largely because of it. I also found hundreds of low-lying cherries to pick to make things better, mainly with loudspeakers, due to measurement. The most important thing I continue to learn are measurement's shortcomings. It's a much longer list than its strengths. Still, measurement is great fun and and I personally would not enjoy tinkering with my audio system without it. It weeds out most of the obvious and not-so-obvious stuff, but still leaves a large gap that can only be closed by listening. Many of the folks in the "measurement is god" camp will never experience a really good playback system because their religion prevents it. Same for many on the opposite end of the spectrum, but at least they may stumble onto it if they have any money left after buying cables. Perhaps more than any other discipline, audio engineering involves not only purely objective characterization but also subjective interpretations. It is the listening experience, that personal and most private sensation, which is the intended result of our labors in audio engineering. No technical measurement, however glorified with mathematics, can escape that fact. - Richard C. Heyser The "measurement is god" website I learn much from brings this to mind: God bless you and your precious family - Langston
  20. I am very familiar with THAT Corp, you obviously are too, but the following may be of interest to others. : ) Here's a reformatted summary of the why's and how's of the design. The designer, Bill Whitlock, is tenacious, brilliant and one of the most giving people I've ever met in pro audio. He's retired now but still teaches on grounding and other topics at a level that no one else can or will compete with. He too wears "No BS" buttons, though with a bit more illustration. : ) He's a transformer man (ex-Jensen president) and designed and patented the circuit you're referring to that achieves about a fifth of the common mode impedance (10MΩ) of a top of the line Jensen transformer (50MΩ). That allowed it to achieve better than 90dB CMRR for source imbalances up to 1kΩ. That means an unbalanced RCA output feeding this thing will have 90dB+ full bandwidth common mode rejection! That's why the best pro audio mixing consoles offered transformer I/O back during analog's hey-day. The one Achilles heel that all active input circuits have is the limitation (think clipping) of not being able to deal with common or differential mode voltages exceeding their power supply rail (24V max). Much above that you have to pre-clip the incoming voltage with diodes to protect the chips. Jensen's best input transformer, the JT-16-A, can handle 250V for a minute and peaks many times that. Some of the best sounding audio gear ever made uses transformers - their problem is cost, size and weight relative to chips. This was the source of my interest in getting to the bottom of the autotransformers a couple of months ago after I saw one in my Klipschorns. It also accounts for my pro-autotransformer bias, which they proved to deserve. I'm very biased about many things. : ) I've read everything Bruno Putzeys has written that I can find and that guy knows his stuff and obviously has pro audio background. He's the first consumer amp guy I've seen that is serious about the "Pin-1 Problem" and the fact that he added an instrumentation amp frontend to the Hypex modules means he's fully knowledgable about common mode issues. He also mentioned that the Whitlock patent made the bootstrapped instrumentation amp too expensive for him, thus the NC400's common mode input impedance is "only" 1MΩ or so. Still world-class compared to every other amp input out there other than the AHB2, which also uses the non-Whitlock instrumentation input. - - - So back to the cheap and cheerful Little Bear passive attenuator. Its use of ganged potentiometers makes impedance unbalancing unavoidable, but nothing along the lines of the 1kΩ imbalance that Whitlock's design or transformers or the AHB2 or the NC400 would care about. But it will trip up the CMRR of standard balanced inputs ubiquitously used elsewhere, such as in the new Purifi modules. This STILL isn't a problem unless you have common mode voltages on pins 2-3 due to EMI near a much longer run of cables than a home setup will normally have. Even then you can simply move the cable a foot or two from the EMI source and all is well. A matched switched resistor attenuator will maintain tight balance, but again that's unnecessary if you have 1MΩ+ common mode input circuitry. God bless you and your precious family - Langston
  21. I looked at the Purifi specs and noticed Bruno eliminated the instrumentation amp with it's 4.7x gain (as found in the NC400) in front of the differential amp's 4.5x gain. Thus the Purifi amps just have the differential amp at 4.4x or 12.8dB gain, which is a helpful move toward further noise reduction (gain optimization). You won't need to pad the input of these new amps, but you will need a preamp that'll output 9.6V (19.6dBV) cleanly to get full power out of the amp. The advantage of the instrumentation amp frontend is that you get transformer-like common mode rejection due to the MegΩ range input impedance. The loss of this feature is no big deal for home audio IMO. The new 4.4kΩ input impedance will push the current capability limits of some preamps, but it's likely a finished product will add a buffer between the I/O connectors and amp module. Back to the issue of padding the inputs of amps that have too much gain - the $37 passive attenuator that Claude provided a link to really is amazing for the money. I got one just to measure it and it's pretty good. You can't get matching impedances between pins 2-3 with these things, so CMRR is compromised, but again I don't see the harm in that with the short cable runs in the home. If you want to go there, switched (matched) resistor passive attenuators can compete or better high-end active preamps in my experience. Not cheap. I had Shallco build me one in the 80's for around $500 as I remember. They don't make old school attenuators anymore, but there are plenty of others, both kits and prefab. If you just need a set-it-and-leave-it pad for an amp, I'd use a matched 4 resistor double L-Pad inside the male XLR connector feeding the amp. $37 Little Bear passive attenuator (balanced I/O) gain reduction vs. magnitude Little Bear gain reduction vs. crosstalk Switched, matched resistor gain reduction vs. magnitude to half volume Switched, matched resistor gain reduction vs. crosstalk to half volume God bless you and your precious family - Langston
  22. Here's what I did: 1. I chose a single, well-recorded 96k/24bit track freely available from HDTracks in their 2020 Sampler for the test. 2. I played it back with no EQ, etc., from a PC using Roon playback software through a DAC3 (USB connection) and into the amps while they were connected to a pair of unmodified Heresy II's. I matched the output of each amp within 0.01dB using a -1dBFS swept sine at a voltage equivalent to 1 watt into 4Ω. This resulted in the test song RMS being closer to 0.1 watt during playback, which sounded about right for a critical listening volume. 3. I simultaneously recorded the output of each amp loaded by the Heresy's with the Audio Precision analyzer set for 96k/24bit at a fixed input gain. Given the effort they made to perfect the audio interface, it makes a phenomenal recording platform. 4. Finally I recorded the original test song at the same level, but directly out of the DAC3 to allow you to hear it without the amplifiers in the circuit. Recording Setup Block Diagram Recording Setup Front Recording Setup Rear Music File Downloads 1. Original Song (DAC3 Output without Amp) 2. AHB2 Song Output 3. NC400 Song Output God bless you and your precious family - Langston
×
×
  • Create New...