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Chris A

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About Chris A

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    Music Enthusiast

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    Arlington, Texas, USA
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    Small room acoustics, acoustic measurement, acoustic horn design (including multiple entry horns), sound reproduction system design, and source music remastering (restoring/rebalancing music tracks).
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  1. It's the maximum frequency before the woofers encounter the apex reflection frequencies, a band of frequencies that are cancelled due to the off-axis port acoustic waves bouncing off the throat of the horn and returning through the mouth of the horn. This is a very strong low pass phenomenon that is characteristic of multiple entry horns. The throat-located compression driver can play through these frequencies, albeit with a loss of off-axis SPL due to the loss of off-axis port area reflectivities. Usually, MEHs are set up to overlap the crossover region by the higher frequencies and lower frequency drivers in order to cover the missing off-axis horn reflectivities in the off-axis port regions. I used Hornresp to verify that the T/S parameters of the woofers that I chose were well suited to both the high frequency and low frequency performance of the woofer pass band, i.e., that their SPL output relative to the throat-mounted compression driver was the same output levels without pronounced dips in SPL on their high frequency end of their response, and that their lower frequency response was at a relative maximum and smooth with frequency vs. other woofer driver choices that could have been made. Hornresp is actually rather limited in terms of what it can do outside of that type of performance. It would take a fairly detailed Boundary Element Model (BEM) to delve deeper into the acoustic performance of the MEH geometries and driver electro-mechanical parameters to understand much more. The difficulty of using BEMs is at least 100x higher than using a lumped model like Hornresp. I've found it to be easier to use rules of thumb and "cut-and-try" approaches rather than to presently go down the path of BEMs. That might change in the future if more MEH designs are attempted, but that's another discussion entirely. The size of the off-axis ports and the relative compression ratio of the woofers (woofer diaphragm area divided by the off-axis port area) were factors that were selected based on rules of thumb and testing of the resulting off-axis port area/placement and woofer diaphragm area. The nominal efficiency of the woofers was calculated using Olson's formula for driver/horn throat area efficiency. Chris
  2. If you're primarily EQing bass response below the Schroeder frequency of your room, you'll need to back off from each loudspeaker to about one wavelength (i.e., 3.3 m at 100 Hz, etc.). This is especially an issue with Khorns, which have a wide bass mouths spread angle (72 degrees) relative to a Jubilee bass bin (about 29 degrees). This bass bin mouth spread issue alone will create issues with Khorn microphone measurements between 120 Hz--the approximate frequency at which the bass bin loses its horizontal polar control--and 400 Hz--the center crossover frequency to the midrange horn/driver. If you're primarily EQing above ~200-300 Hz, then I'd recommend measuring at 1 metre on axis, with the microphone pointed upward at 45 degrees, and vertically located at the collar between the bass bin and the top hat. For my room and the Jubs, I usually place the microphone at 1 metre on-axis and EQ flat, then I check at the listening positions below 100 Hz, making any obvious changes to the bass bin EQ that seem to be consistently presenting in the measured frequency and phase response plots. Chris
  3. DSP options for your Pro Klipsch Audio

    I use balanced connections on the Pre/Pro and on the amplifiers--except for the First Watt F3 driving the TADs on the Jubs. It turns out that the relative gain using a balanced Crown D-75A is the same as the First Watt F3 using XLR-RCA cable (by chance alone)...verified using REW and a calibrated microphone to check relative levels. I found this out when I sent my F3 back to Nelson Pass for repair: there is one FET on each channel that gets really warm and the solder connections eventually make intermittent contact. I now flip the F3 over, apply a big soldering iron to the connectors for a couple of seconds, then put everything back together again. Voila! Works again. I've got a hand full of FETs that NP included as spares. P.S. the F3 is on probably 15 hours/day, and over 4 years, accumulates a lot of hours. I'm still not giving up my F3, however... Chris
  4. Well, I guess that an MEH crossing over at 500 Hz will be too big, unless you can figure out how to miniaturize sound waves... Seriously...the SEOS 30 is actually about as small as you can go without losing polar control at the horn mouth at or before you get there (with a little margin to spare...which you'll need before you're done). What I'd recommend looking at is using a big-mouth horn that's fairly short--like a straight-sided horn...also called "conical". Then worry about how to package that size horn mouth in your height channel placement areas. Then I'd recommend looking hard at more unorthodox room placement positions. You'll find that you can do it with some amount of compromise, but you'll be giving up some or all of the low frequency performance unless using a largish open baffle or box around the horn mouth. Chris
  5. The placement of the off-axis ports relative to the throat (apex) compression driver determines the low-pass behavior of the off-axis ports. I placed the off-axis ports where they would have a low-pass of 475 Hz--and hit it right on the nose with the prototype. If you want to cross over at a higher frequency, those ports must be closer to the throat. The issue with that is that you're making a trade against the off-axis coverage angles (vertical and horizontal) due to the ports, or you're giving up horn efficiency (a LOT of it) if you make the off-axis ports much smaller in area. So there's a trade that must be made. I could've placed the off-axis ports at 1 kHz crossover frequency point in the horn (which turns out to be the 1/4 wavelength from the horn throat at the center frequency of interest), but then I'd be giving up a lot in terms of off-axis performance at and just below that frequency, and a lot of woofer-horn efficiency. The trade-off point (475 Hz) that chose was just right for my needs, and results in an extremely clean-sounding MEH that fills the room with coherent, clear, and tonally balanced acoustic energy. Chris
  6. Playing with active crossovers.

    The following two threads: and If you're using a MacIntosh, then this thread will help a little: Chris
  7. Yes, they can... The application software is free. So are the tutorials... The results speak for themselves. If you would like to hear demastered tracks, you need only mention your desire to hear a few tracks. Chris
  8. @boxerjake from the LaScala bass? thread: I've found that it's much more effective to demaster the music itself to restore the bass below 100 Hz since the original bass frequencies that have been attenuated are still there and at high signal/noise levels. When an inverse mastering EQ curve is employed to restore a1/f cumulative spectrum curve (-16 dB/decade) to the tracks, the bass is restored, i.e., no sub-harmonic "synthesis" is required, and the resulting sound is much more realistic than using bass synthesis. I've also found that many tracks possess bass noise that exists at specular frequencies corresponding to droning line noise (50/60 Hz) and HVAC noise (typically found at 7, 9, 13, 17, 19, 23, 27, 32, 37, and 41 Hz). Additionally, there are certain recordings where the second harmonic of the line noise (100/120 Hz) and sometime third harmonic (150/180 Hz) is present and visible in the spectrograms. Performing notch filtering at these specular frequencies on a case-by-case basis significantly cleans up the recording and results in much more transparent and clear midrange (via significant reduction in higher frequency modulation distortion with these problem bass noise frequencies). Note that this demastering for bass restoration requires the use of spectrograms and cumulative SPL density spectra plots to visually identify the incoming issues with the tracks, and confirm the effect of the iterated demastering inverse filters--along with careful listening using a calibrated stereo setup in-room to verify the resulting demastered tracks. The results speak for themselves. Also note: once the track is demastered, no further editing or work on the resulting tracks are required...you're done having to deal with the issues--unlike the use of plugins at playback time. Chris
  9. @DizRotus This is from the "Right this Minute" thread... I've found that the dozen or so Chesky albums that I own are interesting...from a demastering point of view. If you believe that these albums are "reference" and that they are your "go to" discs for evaluating , you may want to stop reading here. What I've found is that, while the Chesky albums are not really mastered for loudness--just a reasonable amount of compression/limiting is used to control the average levels to something like -14 dBFS to perhaps -19 dBFS, they were mastered for a particular "sound" that clearly differs from that which was recorded in the recording studio. What was done in mastering was that significant levels of mastering EQ was used to change the overall timbre of the music. In the particular case of the Rebecca Pidgeon discs (particularly The Raven) the mastering EQ seems to have been used to make her voice sound "younger" or higher pitched. [This is not unprecedented: I have found the same techniques used on Frank Sinatra albums from the 1940s-1960s.] In general, the EQ mastering made to Chesky discs changed the timbre or overall sound of the recordings in pleasing ways. When you hear the demastered Rebecca Pidgeon albums, you will be likely be torn between the two versions, mastered and demastered, and might want to retain both versions for comparison purposes. Personally, I feel that the original sound (i.e., before creative mastering EQ was applied) is my preferred target sound. Others might like the mastered versions. Chris
  10. Depends on the size of the room, for sure. If your listening room is really small (e.g., 12-15 feet laterally each direction, ceiling at 8 feet), you'll probably not hear a lot of difference between a DR and horn-loaded sub. Small rooms pressurize much more easily, so the sub is able to provide reasonably linear output with controlled compression distortion. As the room size gets larger, you'll definitely hear the difference, even if the bandwidth of the sub(s) is limited to an octave or two. This is not subtle, but can be disguised to some degree by moving the crossover frequency down as low as possible (probably 50-55 Hz) assuming that the "Uccellos" are room corner loaded. This works up to the point that the DR sub must be used above perhaps ~85-90 dBC at one metre, then you'll hear the compression distortion take hold on the DR subs, while the horn-loaded sub(s) basically keep on going without apparent limitation. You may think that this isn't a big deal because you might think that you don't really listen that loud overall. But for low frequencies, this really isn't the case. The following equal loudness contours will show you why subs basically play loud all the time: the curves get very close together below 100 Hz and therefore require the subs to play at a relatively high intensity all the time--only small changes in output acoustic pressure lead to big changes in perceived loudness, and it takes a lot . In a small room, DR subs can usually pressurize the room well without moving mass effects and compression distortion (as well as ohmic heating of the voice coils) beginning to take over. All of these effects seem to gang up on the DR subs at the same time, so it tends to be very threshold-sensitive, and I've found that it's strongly related to the size of your listening room. Chris
  11. Bumping for those new to using Xilica DSP crossovers...
  12. Bumping for those new to using REW to equalize their systems using DSP crossovers.
  13. Response curves run by PWK

    I believe that you're looking for Jim Marshall.
  14. Response curves run by PWK

    FYI, For others that might not be conversant, it's modulation distortion that is the issue...not harmonic distortion that is seen in the type of tests that are described above. In order to see modulation distortion, two tones must be played at the same time: Chris