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Chris A

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  1. I once read a story about how "analog was better than digital" from some guy that was talking about mixing tracks. I was puzzled by this claim, until I read how many tracks he was mixing: over 100 tracks at once. 😬 I wonder if the concept of "high fidelity" actually means anything after that operation. I don't have a lot of use for multi-tracked recording techniques, just in case you might have wondered. I understand limited use of these techniques, but nothing like what we see today from popular music recordings. Chris
  2. This brings up some questions: 1) Does anyone remember the SNR of a typical RTR tape or vinyl LP? 2) What does quantization noise sound like? 3) Has anyone here ever heard quantization noise off a CD? Chris
  3. It means the 144 dB figure of merit dynamic range isn't a good way to look at the added 8 bits to a digital word for 24 bit PCM recordings. It means that all you can hear in those added 8 bits is (perhaps) different music decays (as I alluded to above)--if the quality of the recording can reveal those differences. 99% of the recordings that I've come across can't do that. Additionally, this throws into some question the notion of "HDCD" recordings--the added bits of effective bit depth are bogus--unless... Chris
  4. By the way--I forgot to add this: 24 bit music has a maximum of 144 dB of dynamic range. There aren't any microphones that have that kind of dynamic range...
  5. Unless you've got a room that's got 96 dB noise-floor-to-loudest-output, adding another 8 bits to 16 bits does absolutely nothing if looked at in terms of reproduced dynamic range, defined from the digitization noise floor. However, if you look at the difference in adding bits to the music decays using much less than 96 dB of actual dynamic range playback levels in-room (in effect--inserting more bits per dB quanta of reproduced loudness), then you might recognize the effect of adding bit depth in the recordings, something that I've detected myself, but the recordings have got to be absolutely outstanding. Classical music recording halls are limited to much higher background noise levels than 96 dB, or even 60 dB of dynamic range, as measured from the ambient noise floor to a nominal loudness level (i.e., "83 dBC") is difficult to find nowadays, and the probability is pretty much zero for pop/rock music recording studios. That's in fact what I hear--a more "solid" decay of transients on the best recordings--going from 16 bits to 24 bits--but it's difficult to say when you've got it and when you don't (i.e., being able to detect added bit depth in A-B fashion). But you're not going to hear that on any pop/rock recording, unfortunately. It will necessarily have to be on a recording having much greater dynamic range than anything produced for online streaming or typical popular music CD albums. Most rooms have something like -40 dBA noise floor. I'm listening to Ravel's Bolero as I type this, and have just turned it down once due to that dynamic range of the recording being too great for casual listening with someone else in the house trying to concentrate on what they're doing. Chris
  6. If the world keeps ignoring the production quality of albums (especially the kind of production methods used on pop/rock tracks since 1991), and continues to pay so little attention to directivity and phase response of their loudspeakers (in addition to SPL response), the results that were posted above won't change. I have a particular bone to pick here: it seems that there is an assumption that all music is in stereo format nowadays. This isn't a very good assumption, and the sound quality effect of moving to 5 surrounding channels is fairly dramatic. Lots of people own 5.1 or greater sound systems in their home theaters (whether dedicated HTs or not), but apparently do not choose to upgrade the sound quality of these systems to match or exceed the quality of typical "audiophile" stereo systems. If audiophiles continue to pay so much attention to the format of stereo music tracks, then nothing much will change. I think Waldrep pretty much was okay with posting the results of the study because he is doing music production in a way that's a cut above, i.e., he doesn't do "mastering" of his AIX recordings, etc., and is marketing basically the "mixdown" tracks to 5.1. What he produces reflects that fact. I've got ~10 of his albums and I can say that the production quality is outstanding--and all of it is in 5.1 format as the first format encountered. It's interesting that the difference between DVD-V (16/48) and DVD-A/Blu-Ray (24/96) is audible on his discs (...try his Lowen & Navarro recordings, for instance...) but only in a subjective way and only if everything is dialed in carefully on the Jubs/K-402-MEH/AMT-1-Belles. It's impossible to hear these differences on any other system that I've listened to. What you're listening to in terms of the loudspeakers, their setup, and the room acoustics is the major limiting factor, I've found. Old ears can be a limitation above 15-16 kHz, but there's not a whole lot of music above that point (and very few notes on the keyboard)... Chris
  7. I really don't know. Here is a Google Books hit that I just found: https://books.google.com/books?id=SIFjKr9IAYMC&pg=PA197&lpg=PA197&dq=Cox+D'Antonio+Avis+software&source=bl&ots=A9JpAryg2N&sig=ACfU3U28qBnoB04jX59_QpxnWf2z6RjK-g&hl=en&sa=X&ved=2ahUKEwianu7Uk7LqAhVEQ6wKHZZKASAQ6AEwAHoECB0QAQ#v=onepage&q=Cox D'Antonio Avis software&f=false Chris
  8. Sorry, I had confused the Peavey FH-1 and FH-2 with the EAW BH 882 (the reason for the nose extension credited to Larry Levan--actually designed by Richard Long & Associates ) Chris
  9. In order to use the DSP crossover, you'll need preouts from the preamplifier section of that unit: As far as the dimensions of the room, they look good to me. If you look at the following figure, you can use the height of the room (10'--the shortest dimension of the room) as the baseline for the other two dimensions to determine "sweet spots" in relative room dimensions. In your case, your relative room dimensions are 1.7 and 2.3 and are shown in the figure in red: darker areas are better on the plot. (The figure is taken from here: https://community.klipsch.com/applications/core/interface/file/attachment.php?id=79899): Make sure to use carpet on the floor at least next to the bass bins and outwards radially for at least 3-4 feet (i.e., 1m). Chris
  10. Almost. In the BMS 4592ND, there is a one wavelength time lead that the HF diaphragm has over the midrange diaphragm at the natural crossover frequency (145 microseconds). I would assume that the new B&C driver has the same time delay of the midrange diaphragm relative to the HF diaphragm. No matter--it's easily correctable using one of these plus another stereo amplifier: https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4-hd Once you do correct the BMS 4592ND on a K-402 horn, the sound is indistinguishable from TAD TD-4002s, I've found. It looks like it will be the same for this new B&C driver, too. Chris
  11. See http://arqen.com/acoustics-101/speaker-placement-boundary-interference/ The characteristic boundary reflection distances for dash mounting drivers (i.e., those without "waveguides" or horns to control their polar responses) is about 4 inches or less. Chris
  12. The issue that you're looking for, I believe, are the 1/4 wavelength boundary reflections. These reflections will probably occur in the 700-2000 Hz band, and will be cancellations (nulls). Frequencies below that band will be supported like you have a horn there, so you're going to get a lot of gain (as much as 20 dB vs. the null frequencies). Additionally, this means that the polar lobing issues will be pronounced in this frequency band, so as you move the microphone around (up/down, side to side) from the seated position head heights, you're going to see pretty wide swings in output. I'd recommend drawing a side view of the loudspeaker/windshield and a top (plan) view of the same to understand where these 1/4 wavelength cancellations will occur. These are fairly straightforward to draw if the curvature of the windshield is known, but more difficult if the windshield shape or the side wall reflectors aren't straight. But I think you'll see how to estimate the problem areas once you lay it out and calculate the path lengths from the driver to the reflector to the listener's ears. Chris
  13. The offer is open, but presently subject to the recent constraint of pandemic quarantine. The offer will again be open to all that wish to hear what I've described. _______________________________________________________________________________________________________________ To others reading this that are not often through the D/FW area--note that there are a few other Jubilee owners spread out geographically in the US and internationally that I've also helped to dial-in their setups using REW and DSP crossovers, too. I'll let those people self-identify. Those individuals here wanting to hear the difference won't be disappointed in what they experience. If you alternatively get a chance to hear Synergy™ MEHs, these loudspeakers are also inherently time-aligned and have full-range directivity. The only variable left in those cases is room geometry/placement and acoustic treatments to suppress early reflections (within 3-4 feet of room boundaries, acoustically reflective furniture, or electronics equipment), but also retaining boundary gain (1/4 space or 1/8th space) for low bass distortion. Chris
  14. Anyone wanting to hear the difference due to time alignment has to first get them time-aligned and then listen to them--in order to hear the differences that you describe. [I believe that is what I referred to in post #6, above.] If you don't know if they're time aligned, you can almost bet that they're not--unless they are MEHs or perhaps higher priced studio monitors (which have the issue of nearfield reflections due to their lack of full-range directivity, as I also stated above). To my knowledge, I know of no stock Klipsch loudspeakers other than dialed-in cinema (professional) loudspeakers like Jubilees or MCMs--all the others use passive crossovers and are not designed to have time-aligned drivers using those passives, and all others beside the Jubilees and MCMs have direct radiating woofers. It takes the combination of full-range directivity and time alignment with suitable room treatments to suppress in-room early reflections to maximize the audibility of time alignment. Khorns, Belles, and La Scalas all have significant time misalignments as sold by Klipsch, i.e., you'd have to tri-amp them and dial them in using a DSP crossover to hear the difference. Chris
  15. I've also found that dialed-in Jubs and K-402-MEH exhibit a subconscious listening effect once time-aligned, besides the measured differences in performance. [Note again that these have full-range directivity control down to the Schroeder frequency of the room.] They're dialed-in such that polar lobing is minimized via DSP crossover tweaking and moving the K-402s down toward the bass bins in the Jubs--something that I'm pretty sure you didn't hear in Hope during Roy's class. The MEH doesn't require anything but perhaps first-order crossover filters to achieve time alignment of all drivers and effectively drive the effects of polar lobing to inaudible levels. Those listening effects of time-aligning that I've found can be summarized as: 1. significantly increased perception of bass (so much so that I had to re-demaster my stereo recordings). 2. overall apparent depth and seamless soundstage improvements, and an indescribable naturalness of sound that's easy hear with live acoustic instrumentation recordings 3. elimination of harshness, particularly of acoustic instrumentation like acoustic guitars, violins/violas/cellos/double basses, and wind instruments (brass & woodwinds). Chris
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