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almikel

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About almikel

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    Mike

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  1. they look great!...a big waveguide on the the tweeter, 3 way active sealed... ...I can't see any details of integrating a sub or subs, or a digital through - I may have missed it. Assuming you add a sub or subs, depending on mains vs sub positioning, and the crossover network between, for best mains/sub integration you may need to delay your mains more than your sub/s, even if your subs are cosited with your mains. Ideally you want a setup that allows delay individually across the active speakers and sub/s. "Digital through" will provide zero delay (well almost) to the subs, and the subs should be able to manage any further delay required. The DSP within the speakers need the capability to provide delay so they can be "behind" the digital through to achieve maximum flexibility. Ultimately it will depend on the speaker/sub/listening position/crossovers and the room to determine delays to achieve time alignment across speakers, but it's not necessarily intuitive - my sub is physically closer to the listening position, than my mains, but my main speakers require more delay to achieve time alignment with the sub. You can't do this alignment by ear - but once done your ears agree it sounds better. Mike
  2. On a day like yesterday or today, any additional heat in the room makes it that much more unbearable... ...I ran dual 10W transistor class A monoblocks on my tweeters for for a while - crazy additional heat I didn't need except in the depths of a Brisbane winter : )- no way I would have listened to the stereo over the last month with them in the setup... Any class A amp heats the room too much IMO - I swapped out my 10W amps as they clipped - lower powered amps would clearly not suffice in my setup. I'm assuming you wouldn't say those beasts don't add heat to the room?
  3. I bet you have aircon in your room...especially if listening over the last few weeks...the heat/humidity in Brisbane has been foul...no way I could put up with additional heat sources in the room
  4. I accept that all pro audio kit is designed with balanced inputs and outputs (assuming a pre-amp of any description - power amps don't have balanced outputs, they just have an output) - for good reasons - in the pro world they may run very long interconnects, with gear powered off different power circuits with different earth references - by running "balanced" they avoid earth loops and hum. Many top of the range domestic designs aren't fully balanced - as they don't need to be - although I completely accept that in a domestic system, it's a complete PITA if you have an earth loop and need to track it down and eradicate it... Possibly for the input stage of the power amp - but never for the output stage of the power amp A fully balanced input stage will have additional components on the input of the amp - but this is only a slightly more complex summing junction that a non balanced amplifier input would have...in SS this could be op-amps - so yes separate symmetrical amplifiers - and almost guaranteed to be op-amps in pro gear...but a balanced/unbalanced input doesn't define good vs bad... I don't know enough about valve power amp design, but I do know every unbalanced SS amp design will use a differential input pair of transistors/mosfets/devices...and by definition all amplifiers work on "differences" between the +/- inputs...so I'm not sure what you mean by "differential in design" - they all work that way. Mike
  5. I agree, except for a better pre-amp...as most active speakers will take a digital input, removing the need for a pre-amp entirely unless running analog inputs (eg vinyl), and even then add a decent Phono and just plug it into the analog input on the active speaker - no pre required. If I was starting from scratch I'd definitely look at active speakers - these days they have all the features that 10 years ago only high end setups like DEQX had, and MiniDSP was in its infancy. The JBL series (eg https://jblpro.com/en-US/products/708p ) includes the ability to delay the mains more than the sub if the sub has a digital input - this has been a challenge that only active preamp setups (eg DEQX, MiniDSP and others) have been able to meet before now. Completely agree - if you can get away with it, room treatment makes a massive difference to the "in room" sound - especially in the bass end - but unfortunately treatment gets too big when managing lower frequencies - and EQ becomes a better option below 150Hz or so. IMHO get the "in room" bass under control and you're 80% done in achieving great "in room" sound. Great "in room" bass is a combination of best positioning of speaker/subs/listening position, treatment, and EQ. Modern active speakers provide the ability of EQ, but still require careful positioning of speakers/subs, the addition of treatment, and the judicious/careful application of EQ to make the "in room" bass truly great. EQ has a deservedly poor reputation - poorly applied it can yield dreadful results. Well applied in combination with positioning and treatment, EQ is essential IMHO. A modern set of active speakers, with a sub or subs - setup well, with room treatment, would be hard to beat IMHO. cheers, Mike
  6. I understand that - I've run an active setup since the mid 90s and a DEQX setup since 2010 - all with external amps. Mike
  7. think about it this way - with single ended you only have a single output device (ignoring paralleled output devices for a sec) to manage the entire voltage swing - so always class A cheers Mike
  8. terminology can confuse people - I'm sure you mean "push/pull" rather than "balanced" ...lots of valve power amps are push/pull, but there are also lots of single ended valve amps (eg SET). It's interesting that back in the 50/60s before transistors took over all high end valve amps were push/pull (Quad, Leak, McIntosh). Single ended designs were for mantle radios and other low fi requirements. It's far more recent that SET style valve amps have gained in popularity for "HiFi" - they were never deemed good enough when only valve amps were available. cheers Mike
  9. I thought there were valid reasons valve amps hum more than SS, but on a quick scan of my usual technical sources I can't find any references...it certainly could be just design: single ended designs (eg SET valve amps, but also SS) won't have as much Power Supply Rejection Ratio (PSRR) as push/pull designs, so noise on the power rails will be more audible class A designs (which include all single ended, and some push/pull) have very high current demands on their power supplies, so if the power supply is not appropriately designed, more ripple may exist, increasing hum interaction between the power supply transformer and the output transformers on valve amps could increase hum in valve amps. SS amps won't suffer from this. it is common for SS amps to have higher levels of negative feedback than valves, and the greater negative feedback would reduce hum I know of one highly respected valve amp manufacturer that quotes their hum figures (full respect to that manufacturer) - that would lead me to think there is some truth that valve amps are more prone to hum than SS...but again it may be through design (eg lower negative feedback) Class A is a lot more common in valve amps than SS, and I've never come across a valve amp that wasn't hot. I live in Brisbane, and it's just way too hot to run class A amps of any sort (valve or SS). I did run some 10W class A transistor amps for while on my tweeters. The heat they generated was one reason I moved them on. I've gone back to all SS class AB, and the heatsinks stay at ambient. A key reason why good valve amps cost a lot is the quality of the output transformers - they can make or break the sound of a valve amp...and good output transformers are expensive. cheers Mike
  10. the response of passive filters depends on the impedance of the driver. Take a simple 1st order high pass/low pass filter The cutoff frequency Fc = 1/(2 x pi x R x C) R could be the tweeter, or the tweeter could be connected in parallel with R Similarly for a 1st low pass using an inductor The cutoff frequency Fc = R/(2 x pi x L) Similarly R could be the woofer, or the woofer could be connected in parallel with R From inspection of the formulas it's clear that the cutoff frequency of the filter depends on the impedance of the driver connected. Well designed passive crossovers will include an impedance compensation network to "smooth" the impedance of the driver as seen by the crossover network, so that the crossover frequency doesn't change. Some good info is available in one of Rod Elliot's articles here: https://sound-au.com/lr-passive.htm Let's say the noise floor in your room is around 30dB and you listen around 80dB, so for convenience we'll say by the time the crossover is 48dB down out of it's pass band, you'll not hear it - ie no interaction between filter stages from a volume perspective. Let's say you picked 375Hz and 3kHz as your crossover frequencies (to make the maths easy - 300Hz and 3kHz is more typical) in a passive setup with 2nd order (12dB per octave) filters. When you calculate the rolloffs for 2nd order passive, the mid highpass (@375Hz) is not far enough down not to interfere with the tweeter high pass (3kHz) to produce a smooth result, and the mid low pass is not far enough down not interact with the woofer low pass - so from this perspective the filter stages interact/interfere with each other. The other aspect of interference that I didn't mention above is where the inductors of passive crossovers couple to interfere which their individual responses unless physically separated enough, or mounted 90 degrees opposed. With 2nd order/3 way passive, including coils for driver impedance matching, this gets complicated. Going 3rd order passive (18dB/octave) reduces the interaction between crossover stages, but increases the inductor count, making physical layout harder again. 24dB passive just gets ridiculous - but is the usual starting point for active crossovers... ...an LR4 (Linkwitz Riley 4th order) crossover has a flat frequency response and the drivers are in phase (electronically), and has a minimally compromised transient response. Of course achieving an acoustic LR4 crossover is a bigger challenge! - an electronic LR4 usually doesn't equate to an acoustic LR4 unless the drivers in their enclosures are flat way beyond the crossover frequencies (ie >1.5 octaves away) of course - all crossovers have overlaps - and it's a given that gaps are unacceptable. And crossovers shouldn't be so steep to compromise the transient response too much - always compromises. do you mean more DSP filter taps? I accept that with DSP, the processing power (and taps) required increases as you go lower and steeper - and the processing/taps required for linear phase (FIR) are higher than with IIR (eg Linkwitz Riley). But with my DEQX setup I've never run out of processing power for crossover frequencies >300Hz as steep as I want, say 96dB/octave, but I choose to use lower slopes. It's the crossover to the sub below 100Hz where my DEQX forces me to use IIR filters rather than linear phase, as the delay gets too large...and I'd prefer to use IIR LR4 24dB/octave filters in this region anyway. Be careful with terminology - I ran analog active for a long time before going DSP active - don't confuse the term "analog" with passive crossovers. I would disagree - IMHO the compromises in passive crossover systems are too great to bother attempting to build a "well built" passive crossover setup. Sure active setups have their issues, but IMHO a well setup active system will always exceed the performance of a passive setup. Another Rod Elliot article that's worth a read: https://sound-au.com/bi-amp.htm cheers Mike
  11. Hi Brian, even though it's all internal, and will never be seen, an artist would appreciate what you've done... I'm sure they will sound fantastic... Mike
  12. the only "active" speakers I've ever run was a Logitech 2.1 setup - at the time more expensive than their other setups, but proper 3 way active satellite 2 ways with sub - sounded amazing (for the price), but clearly consumers didn't go for it, so they dropped it from the range. It was my back deck system for years until a tweater died. Active setups have some key advantages over passive setups, as I outlined above. I'm biased, but in a high end system, I don't know why anyone would accept the compromises inherent in passive crossovers. Some people wouldn't like that active speakers also contain the amplifier...I accept that there are audible differences between amplifiers, but IMHO the room response based on the speakers' response in the room far out weighs any "sound" of the electronics in the chain by a very large margin - YMMV Mike
  13. In the context of this thread - ie box main speakers that are intended to have a sub below them, compared to ported/passive radiator designs - I would agree... there are obviously other options for main speakers (open baffle, electros, maggies etc etc) beyond the scope of this thread. Coming back to "typical" box speakers - in a DIY context, sealed designs are virtually impossible to get wrong - they are just not that sensitive to box volume variations - especially if you have EQ and reasonable amp power available. Ported designs are a completely different ball game - much harder to design, and finicky wrt driver actual response (compared with on paper specs) and box volume/port diameter and length...any "well designed" ported enclosure IMHO requires prototyping...as a lazy DIYer I don't bother - sealed designs just work. In the context of this thread, for main speakers crossing to a sub, I agree sealed mains are a better approach. But sealed mains enclosures don't need to be large, and many drivers are available that get low enough - you just need EQ/driver excursion/amp power available to cross to a sub. In my case my "mid bass" drivers are Acoustic Elegance TD18's in sealed 60 litre boxes - their natural high pass rolloff in their boxes is around 100Hz - this isn't a "problem", as I have ample EQ/excursion/power to push the TD18's down to 50Hz or so to cross to the sub. Going a bit off topic, I don't agree that "sealed" is always better for bass when discussing the bottom octave (ie below 40Hz) - I'm discussing subs now - not mains crossing to subs. The power/driver excursion required from sealed enclosures to hit decent SPL targets "in room" below 40Hz does get larger the lower you go - so I agree with your point that sealed enclosures may have a problem when you run out of amp power/driver excursion. I'm OK with using other designs for the bottom octave - I built a dual driver/dual amp tapped horn for my sub. From a DIY perspective tapped horns are tricky to design - being a lazy DIY'er, I just bought @Paul Spencer's T20 tapped horn design/amps/drivers, and I just made some sawdust and an awesome sub. One of Paul's blog posts compares my drivers/amps in sealed boxes compared to the same drivers in the T20 tapped horn enclosure: http://redspade-audio.blogspot.com/2012/02/how-does-sensitivity-of-tapped-horns.html From Paul's blog, "So the result is that two 12" drivers with 370 watts each can achieve the same output as eight (my edit: drivers) with twice as much power. Since drivers and amps are much more expensive than a few sheets of cheap ply, this works out as a cost effective choice." This is Hoffman's Iron Rule at play combined with the trade-off tapped horns have of being narrow band devices, but being highly efficient within their band...and requiring steep crossovers. I use a symmetrical 24dB LR crossover between my T20 sub and TD18s at 50Hz, with a bunch of EQ to push the TD18s down below 50Hz...then I apply room EQ... ...the room response is pretty flat down to around 20Hz or so, falling off below 15Hz. The difference the T20 sub makes is very subtle - it just adds "weight" in the bottom octave. All the bass slam comes from the TD18s - listening to the TD18s in a treated room with the bass "reasonably" under control is a revelation - tight/dry bass that slams you in the chest, with the sub adding weight below... ...when deciding what mid bass drivers to use to match to my PSE144's, Paul Spencer said, "if you can afford TD18's, go with them"...fantastic advice - I've never heard mid bass like my TD18s in their "small" sealed boxes (and treated room). The TD18's need a sub below - but being sealed it made the integration with the T20 sub below that much easier. cheers Mike
  14. agreed - positioning of mains and subs, and choosing the best locations based on measurements, with some treatment and EQ thrown in, will make a far bigger difference than whether the mains are sealed or ported. air doesn't "move around" in a sealed box, the pressure just goes up and down, unless your box is really big/long. You can generate "modes" within a speaker box, but only at higher frequencies/shorter wavelengths where the dimensions of the box get closer to the wavelength of the sound being generated. speaker stuffing actually increases the apparent volume of the box (by approx 5% or so), and at higher frequencies approaching/within the "modal" range of the speaker, will reduce/damp any resonances caused by modes within the speaker box. Modes within a speaker box behave identically to room modes, and it's pretty easy to calculate if your speaker box is large enough to generate modes. The lowest (in frequency) mode that can exist inside a speaker box is wavelength = longest internal dimension of the speaker box/2. Let's say you had big floor standers with the woofer enclosure 1m internally. So Wavelength = 1m/2, the wavelength = 0.5m - this is the longest wavelength that will generate a mode in that speaker box. V (velocity of sound 343m/s) = Frequency x Wavelength F = 686 Hz - this is the frequency of the lowest mode that will be supported inside the speaker box. If your woofer crosses to the mid or tweater say less than an octave below that (ie above 343Hz), I'd treat the inside of the woofer box with stuffing. If your woofer crosses to the mid/tweater below 340Hz, then modes internal to the speaker won't be an issue. Adding stuffing to sealed boxes has the effect of slightly increasing the box volume, slightly lowering the Qtc, which will slightly change the frequency response - but not limit the voice coil movement if you apply EQ back to flat. Consider a small sealed box (say Qtc around 1.0, ie with a small peak prior to rolloff, then dropping steeply), compared to a much larger sealed box (say a Qtc around 0.5, which is drooping prior to rolloff, but then not dropping as steeply as the small box)... If you applied EQ to each example to achieve the same frequency response, they would have the same excursion of the voice coil (ie driver excursion) - just different power requirements - with the larger box requiring less EQ and less power below rolloff (Hoffman's Iron Rule at play). Mike
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