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Chris A

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Chris A last won the day on November 23 2020

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  1. Believe it or not, the effects that are being described in that excerpt you provided (thanks--that is what I was looking for) pretty much match what I've settled on in my listening room, with some slight exceptions: 1) the QSF minimum dead time is quoted as 2 ms (i.e., 27 inches of travel at room temperature), while I've found in my listening room that that dead time needs to be increased to about 2.5-->3 ms (i.e., about 44 inches, minimum offset distance). This includes the path length of sound being emitted inside the loudspeaker and being directed via the horns to the listener (about an extra 17 inches in the case of the K-402 horn and the compression driver path length). 2) In the case of recording musicians (the original QSF concept), the reflected sound is first emitted near the microphone by the vocalist or instrumentalist, then it travels to the reflective/diffractive surfaces of the tube traps, then it returns back to the microphone--a two-way trip. In the case of a listener's position in stereo reproduction, it's the direct arrivals from the stereo loudspeakers (in your case) and a 5.1 array in mine --> to a reflection or even diffusion from the tube traps--> then on to the listener. This means that for stereo reproduction, the minimum distance of the tube traps from the listener's position needs to be increased over that distance found in a sound recording studio with a musician singing or playing into a microphone--because of the geometry of the sound propagation/reflection paths. I put that minimum distance for home-sized listening rooms to the reflective objects around both the loudspeakers and the listener's position closer to ~40 inches. 3) The tube traps of QSF design are themselves curved-surface diffusers to a degree (by the figure you posted, above, and reproduced below): The setup in my listening room includes diffusion from the disc racks on the right (starting about 1 kHz in terms of depth of the diffusion), and equipment racks on the left. The surrounding AMT-1/Belle bass bins provide additional diffusion surfaces at 90 degrees to the on-axis sight line to the center of the stereo loudspeakers/center loudspeaker. I keep a clear path from the loudspeakers (including center MEH and two corner-located Jubilees) to the listening position, and just enough covering on the floor to control the floor bounce a bit (but not enough absorption to get good REW measurements including phase/group delay, etc.). 4) I've found that the minimum and maximum T30 (reverberation times) vs. frequency is dependent on the room volume, with smaller rooms requiring shorter T30 times. I find that 0.4 second T30 times (measured near the loudspeakers) are about optimal for my size of listening room--which is ~5500 ft3 (156 m3) in volume. 5) The front wall diffusors in your room would be comparable to the brick masonry and elevated center loudspeaker in my array, except that I found empirically that there needs to be more absorption covering the brick masonry on my front wall in order to create a strong phantom center stereo image, along with a row of 2' wide absorption tiles just at the K-402/Jubilee bass bin mouth exits along the side walls. All of these characteristic reflection and diffusor distances are visible in the REW spectrogram plots, including the specular intensity vs. time/distance. Chris
  2. I've heard the differences below 300 Hz using bass traps (which I do use and understand their effects), so I've got a very good idea of those changes below 300 Hz. So the remaining piece is the "filling up the Haas integral with in-room reflections"--at higher frequencies than ~500-600 Hz. These can be very early reflections before 5 ms (and after 0.7 ms), or they can be later reflections. This is the part that I'm curious about, and what the difference in sound in-room is. And identifying whether the reflections added to the direct arrivals are earlier arrivals or later ones. Chris
  3. FYI only: it's been my experience that the frequency band 600-2500 Hz is where I see a rise in T30 values when measured at 1 m in front of the loudspeaker in a predominantly untreated room with loudspeakers having fairly good full-range directivity control, like the following: If you plug in 600-2500 Hz frequencies into the "QRDude" application, it might be interesting to see what results. Chris
  4. Can you describe this in a bit more detail? What did you hear, vis-à-vis with-and-without the bass traps, etc. surrounding the listening position? Chris
  5. So, Mike, you're trying to create a "QSF sound field" in your listening room around the listening position? Chris
  6. I'm aware of "membrane traps", of which I personally classify drywall as a pretty common example for extreme low frequency absorption in-room. In fact, that's where all the real low frequency absorption comes from in real home-sized listening rooms--at least below ~70 Hz. I've personally seen the effects of stiffening the side walls next to the TH subwoofers in the room's from corners around floor level. Anecdotally, I've actually seen somewhere between 3 and 9 dB improvement in some in-room measured SPL response at certain low frequencies--no kidding. Tube traps are another animal--sort of like Helmholtz resonators without the resistive portion of the entrance, but a bit more specific in the shape of the contained internal acoustic volume (i.e., cylindrical). I'm wondering why the interest in that shape, for one thing. And why the concern about "how well" they perform? Are we talking the same kind of measured response differences in these devices in-room? If so, I'd like to know some of those sources/links to some information. Chris
  7. Mike, you've piqued my interest in "tube traps" (which I don't currently own). Can you quantify via data somewhere what you're saying? Something that you've measured yourself is even better. Anecdotal stories are fine but data is much better, in my experience. Chris
  8. In the preferences menu, when you connect an external digital device that can handle audio, it shows up in the "Output Device" drop-down menu: Chris
  9. I had nothing but problems using analog output from a PC using REW more than a decade ago (version 4.11), i.e., analog output from a sound card. It actually delayed my use of REW 5-6 years until I got an AVP that had HDMI input (my first AVP only had DVI video inputs and S/PDIF audio inputs). Once I made the shift to HDMI input from a laptop via HDMI output, all those problems ceased--problems like you're having now. I have to say that some sort of digital output to a DAC, player, or AVP/AVR is highly recommended, even if you have to buy a DAC. DACs with USB inputs can be had for not a lot of money (there are good ones available at less than $150 USD), and will likely improve your sound quality a little bit if you buy a better quality one. I'd recommend taking a look at Topping in their lower price ranges if the price is within your budget. The figure below from Audio Science Review: any DAC not in the "red" group is probably good enough. I've found that it's just not worth the headaches to mess with a PC soundcard. Chris
  10. Do you have an HDMI out from your computer? That is the preferred method if using an Oppo player.
  11. Oh boy....analog out...🤢 Can you connect via a USB to a DAC? Chris
  12. Since I haven't seen how you are connecting to your setup via your computer (output and input), I have to ask... Are you using an analog output from your computer (like a soundcard) or are you using USB or HDMI? If digital, what output driver are you using from your computer (ASIO, Java, etc.)? Are you using a UMIK-1?
  13. So have you changed the gain of your preamp/integrated amplifier? I have a digital readout on my AVP, which I typically set to a movie--loud position (-12 dB in my setup presently), and which is part of my checklist of things every time I pull out the UMIK-1 and REW to measure again. Also, not moving the microphone between measurements is a pretty big deal, unless you're trying to average measurements around the on-axis measurement, in which case, you'd move the microphone around by perhaps 6-12 inches or so, then average the measurements. This is much more time consuming, however, than using just the on-axis measurement. Chris
  14. Yes, I think you missed my points. Perhaps a PM/email would be a better place to discuss. (If you still want to discuss it here, I will of course.) Chris
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