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About almikel

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  1. No well designed amps are slow - their slew rates are plenty high enough for music. And within the pass band of the amp (say 10Hz - 100kHz or so) all audible frequencies will pass through the amp with no impact on timing. All filters cause delay - and the steeper/lower the filter, the more delay - crossover filters and the natural low pass filter rolloff of a woofer in a a box cause delay - not amplifiers. A well designed HiFi system caters for the delays inherent in the filters necessary for multi driver speakers and separate sub/s...often requiring the mains to be delayed more than the subs. Filters muck with timing - amps don't... And the audibility of the timing is questionable - IMHO target a smooth frequency response as the first priority - you'll hear lumpiness in the FR before any timing differences. Mike
  2. because "that's your thing" - which is fine I haven't changed a component in my setup since 2015 or so, but I muck with EQ occasionally, and every change is discernible, just not necessarily better or worse, only "different"... ...I take a pragmatic view, and firmly believe in the maxim "best is the enemy of good" - and I'm happy with good - YMMV. Mike
  3. I have two unused turntables and a stack of old vinyl but no way to play vinyl in my current digital setup. I'm running a DEQX HDP3 pre-amp which doesn't have a great ADC, but it does have a spare digital input. Are there good but reasonably priced phono's out there with a SPDIF or AES digital output? cheers Mike
  4. ^this A phono pre-amp does 3 things: applies RIAA EQ to get a "flat" output applies appropriate impedance loading to the cartridge amplifies the signal to "line level" Some integrated amps/pre-amps have really good phono pre-amps, but most are not as good as a dedicated phono pre-amp. A "moving coil" cartridge will require higher gain from the phono amp than a "moving magnet" cartridge. Some phono pre-amps provide different impedance loading options - particularly capacitance Mike
  5. In simple terms a great room either absorbs or transmits (lets out) low bass so it doesn't remain bouncing around inside the room which results in ringing/long reverb/decay times. Two good references that provide metrics for this: Paul Spencer's Bass Integration Guide: https://www.hifizine.com/2011/06/bass-integration-guide-part-1/ Spectral decay: Decay rate of 20 dB in the first 150 ms from 40 – 300 Hz. Acoustic Frontiers "Acoustical Measurement Standards for Stereo Listening Rooms": http://www.acousticfrontiers.com/wp-content/uploads/2011/10/acoustic_measurement_standards.pdf For room low frequency decay: Resonances from 35Hz­ - 300Hz should not extend beyond 350ms before decaying into the noise floor or reaching a level of ­40dB. Below 35Hz this standard is relaxed to 450ms. Meeting either of the above IMHO would be a great room. cheers Mike
  6. A bit off topic for this thread, so apologies to the OP... QRDude is an amazing free tool to help design 1D and 2D Quadratic Residue Diffusers (QRDs). https://www.subwoofer-builder.com/qrdude.htm User Guide here: https://www.subwoofer-builder.com/qrdude-user-guide.htm You can set a lower design frequency (how low you want diffusion to go), and it will calculate depth , or you can set the maximum depth and it will tell you how low diffusion goes. Just in case you're not familiar with QRDs This is a 1D QRD - diffuses in 1 dimension This is 2D QRD - diffuses in 2 dimensions This is a 2D Skyline QRD - it doesn't have the walls between the wells QRDude does everything you need for designing QRD diffusers for whatever frequency you want...of course the depth gets large if you want to diffuse low - but you can muck with the parameters, and QRDude will tell you where you only get scattering and where diffusion starts to help you make decisions on compromises before making sawdust... Keep in mind that Cox and D'Antonio (the gurus of diffusion) recommend a listening position at least 3 wavelengths away (at the lowest frequency diffused) from the QRD diffuser in order for the diffuse field to be generated - closer than this and you might hear artifacts from the diffuser...and the lower you diffuse the more gap required...always compromises. A simpler type of diffuser is the Binary Amplitude Diffusers (BAD panels) - these don't provide as much diffusion as QRDs - but because they don't provide as much diffusion as QRDs, you could likely sit closer without hearing artifacts. BAD panels are a reflective mask in a random pattern over absorption. A 1D BAD panel is simply different width slats/gaps over absorption - very easy to DIY a 2D BAD panel mask that would be mounted over absorption Similar to QRDs, a 1D BAD panel diffuses in 1 dimension, and a 2D BAD diffuses in 2 dimensions. I've never found design criteria for the lower and upper limits of diffusion for BAD panels - but IMHO they're ideal for the scenario where you've added too much absorption into the room and want to reflect higher frequencies and want to add diffusion at the same time. I tried to pull together some info on BAD panels in this thread - but I didn't get very far @hochopeper wrote some code ages ago to calculate the hole placings for 2D BAD panels in whatever size required. Thread here For 1D BAD panels, you can use any random sequence to derive the slat/gap sequence - flipping a coin would work or use a random MLS sequence like in this thread https://www.gearslutz.com/board/bass-traps-acoustic-panels-foam-etc/395773-diy-binary-amplitude-diffuser-anyone.html You would use the sequence by choosing the minimum slat/gap width at (say) 20mm. Every 1 (or head on a coin toss) means slat, every 0 (or tail on a coin toss) means gap. Three 1's in a row means a slat 60mm wide, four 0's in a row means a gap 80mm wide etc. cheers Mike
  7. I've never tried it, and from their website there's not much info on how it works - but it appears to apply EQ based on multiple mike positions, which is good, but costs around AUD$400, and is targeted at studios running a Digital Audio Workstation (DAW), which is likely fine if your source is a PC. DSP (and EQ generally) can do great good when applied well, but greater evil when applied poorly For room correction I prefer to maintain a high level of control over what EQ and delay I apply - using old school IIR EQ (parametric minimum phase EQ) but leveraging the convenience of DSP for implementation - delay is particularly tricky without DSP, and trivial with DSP. IMHO DSP/delay capability is an essential tool for achieving great "in room bass" particularly when integrating 1 or more subs into the system... ...if integrating multiple subs - providing you have access to EQ/Delay (eg via a miniDSP HD), and a measurement microphone and laptop - I'd recommend free tools such as REW and MultiSub Optimizer (MSO), before costly tools such as Sonarworks. If you don't have subs, then REW and a measurement mike is still useful to measure what's going on with the bass in your room... ....and will help you make informed decisions on when and more importantly when not to apply DSP/EQ. Mike
  8. The speakers have a huge impact - way more than amps/interconnects/speaker cables, but less than the "component" you left out - the room - which has the biggest impact of all - especially below 250Hz or so. Keep the speakers you love, and spend some money on room treatment - easily the best "bang for buck" upgrade for improving "in room" sound. Very hard to make treatment look good in shared spaces - but the difference between an untreated room and a well treated/appropriately treated room that has the bass under control is truly chalk and cheese...significant improvements to the "in room" sound both subjectively and objectively. The room response swamps everything else by a large margin: A good room with reasonable speakers = great sound A poor room even with the best speakers = poor sound IMHO the room has the greatest impact on good sound, followed by the speakers. All the other components have a significantly lower impact - with speaker cables IMHO way down the list of priorities. Bi-wiring vs having the room's bass under control? I would choose having the room's bass under control every time. cheers Mike
  9. monster cable was clever marketing - clear insulation made the conductor look bigger - but that didn't/doesn't make it poor speaker cable - it's fine not much more or less than most other speaker cables - but I am in the camp that considers any "reasonable" cables and interconnects to be sufficient - I use Blue Jeans interconnects and a variety of speaker cables of reasonable gauge for a 4 way active setup. Of course everything has an impact on frequency response, but I prefer to use active EQ when I want to make a change in the frequency response. Many people on this site agree - it's such an inexpensive experiment - just do it - replace the metal plates with "reasonable" speaker wire well terminated - I won't comment further as I haven't run a passive crossover for 25 years or so. yes passive crossovers cause losses, but I went and re-scanned Rod's article on bi-amping https://sound-au.com/bi-amp.htm - and Andy's paraphrasing of Rod's article was based on voltage clipping of amps - where the treble signal rides on top of the bass signal, beyond the voltage swing capability of the amp - under this scenario of course active bi-amping will provide more output without clipping compared to a single amp driving a passive crossover. If a single 100W amp isn't voltage or current clipped, then 2 x 100W amps in a biamped setup is still 100W In lots of your posts I've read, you've demonstrated you have a good handle on electronics - I'm surprised at your "I just believe what Rod says!" comment... I also believe what Rod Elliot says - but not blindly - he writes great stuff and I've learned heaps from his site - he deserves to be taken in context (although I'm sure he's very used to be being taken out of context 😎). cheers Mike
  10. Each mix on every CD/Vinyl/Flac will have a different volume - don't you change the volume on how loud you want to listen? constant volume should be left in the domain of muzak, not hi fi. I do love a remote with EQ adjustment to dial treble/bass up and down. mike
  11. Hi cazzesman, I think you missed @Ittaku's point - redbook 44.1 doesn't upsample nicely into 96 And as @davewantsmoore says, the re-clocking may produce issues. Physical media CDs are the only "things" using 44.1kHz still (it's a legacy sampling rate) - 48kHz and 96kHz sampling are far more common from recording through to streaming...and I'm still not convinced we need 192kHz sampling - but that's a different topic. cheers Mike
  12. I've never done a spectrum analysis on my noise floor, but previous measurements on an SPL meter have shown my noise floor to be around 32-35dB "A" weighted in the evening without plane fly overs and my pool filter or other noise sources easy to control turned off (fridges were still running, but no A/C, washing machines etc). Your graph has piqued my interest - I'll do an RTA when I have some time - I live about 1500m (as the crow flies) from a major freeway - I can't hear it, but I reckon my curve would be similar to yours. Mike
  13. agreed the SVS has a bigger amp (300W for the Krix vs 500W for the SVS) - I can't find details of either regarding driver Xmax (excursion) etc, both have 2" voice coils, although the SVS does have details on their driver such as shorting rings - the Krix possibly has these also (most modern sub drivers do). If each driver has similar sensitivity (which is not a given without detailed specs), then the additional power of the SVS is worthwhile to provide some headroom when volumes get cranked. I will add to this the position of your sub - which was implied in @rayone's post - you need to place the sub in the room to get the smoothest bass at the listening position. Best done with a measurement mic, but you can try it by ear...subs can be placed anywhere in the room as long as they're crossed over low enough (say 80Hz or below), and have low distortion - meet these 2 criteria and your ears won't be able to "locate" the sub - it will just add weight and depth to your main speakers. ...The easiest way to try multiple spots quickly is to place the sub where your ears would be in the listening position, and put the mic or your ears in each likely spot the sub would go - seeking out "moderate" low bass - not boomy, and not a bass "suckout" - much easier with a laptop, free software such as REW and a measurement microphone - such as a UMik: UMIK-1 WWW.MINIDSP.COM The UMIK-1 is an omni-directional USB measurement calibrated microphone providing Plug & Play acoustic measurement. From speaker & room acoustic measurement to recording, this... cheers Mike
  14. significant impact on the FR - it's hard to see much on the spectrograms - what do the waterfalls show? good write up - cheers mike
  15. I don't use automated systems like Dirac or Acourate, and I agree that room treatment and speaker/sub/listening position placement are important, but IMHO well applied EQ/delay is just as important as treatment and speaker/sub/LP placement to achieve great "in room" bass - particularly if you run 1 or more subs. I also agree that crossovers are important - but matching the high pass response of main speakers with the low pass response of 1 or more subs that are physically separated from the mains is a challenge even with DSP EQ/delay available - IME it's not achievable without DSP EQ/delay. Having run a DSP setup for many years (in a lightly constructed room with loads of absorption), I couldn't go back to not being able to manage the bottom end room response without DSP EQ/delay. My room measures OK (not great), but the bass is amazing - lots of absorption cleaning up >150Hz or so and a few bands of EQ cut with a well integrated sub = bass bliss cheers Mike
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