Greetings to All, First, many thanks to Jörn, Bo-Erik, and Michael for your responses to my recent post. Your responses gave me food for thought, and I’d like to add a few comments regarding audio interfaces, psychoacoustics, and pseudo second-order miking (I’m confused in this latter area).
Bo-Erik, I fully appreciate your input. The selection of a good sub is something I need to give careful consideration to. I currently use a single sub that uses a servo-controlled 12-inch driver in a sealed enclosure. The sealed enclosure is preferred because airflow noise through ports is not an issue. It’s also a forward facing configuration, so the *directional* characteristics of the sub (if direction actually exists) are predictable; or, at very least, I can point the driver in a known direction. Unfortunately, this particular sub is no longer manufactured, but I believe there are a number of decent 10-inch subs on the market (and a plethora of junk speakers). Regarding localization, home theatre sound and the 80-Hz xover point: I’ll confess ignorance when it comes to knowledge of a separate or unique physiological mechanism used to localize (or omni-ize) ultra-low frequencies. Within the context of room reflections, music listening, home theatre, and the like, I’m fully aware that frequencies below 80 Hz are near impossible to localize. Add to the overall auditory scene the constituent frequencies that provide unambiguous sound-source information, the need for a surround of subs really goes out the window. Some spout a slightly higher cut-off frequency, but I gather that 80 Hz is the accepted standard for home theatre. I have seen literature on 5.1 and 7.1 referring to subs and low-frequency enhancement for the sole purpose of effects--the surround speakers are still operated full-range. [Although I haven't tried this, I imagine we can accurately lateralize a sub-80 Hz tone under earphones. If so, then a lot of our (in)ability to localize low frequencies in the sound field is mostly a consequence of physical variables such as long wavelengths, head diffraction, room reflections, etc., and not a unique mechanism or deficiency of the brain, mid-brain, or peripheral sensory organ.] Regardless of accepted protocol, I do have reason for using multiple subwoofers, and this reason purposely ignores psychoacoustics. Although this seems obvious, I’ll argue that sounds in nature do not morph to accommodate human perception; i.e., psychoacoustics is a product of a head (with ears and brain) in a physical space. For this reason, I do not want to present stimuli with “built-in” psychoacoustic enhancements or manipulation. Ideally, I want an “untainted” physical space for observing (yes, observing, not necessarily measuring) the perception of special-needs populations. Populations to be considered include the elderly, children who are neurotypical (NT) or autistic and, of course, those lucky normal listeners. There's reason to believe that certain populations may have compromised auditory processing ability (in addition other sensory input) that ultimately results in aberrant or atypical behavior. Because we do not know how such persons perceive and react to sound, the best I can do is provide physical realism. I wish to present accurate sound-source direction regardless of frequency. Psychoacousticians may or may not be in agreement as to how the *average* listener perceives sound, but I'm not interested in average listeners. Admittedly, some assumptions and subjective impressions always come into play when choosing and using audio equipment. For example, I’ve yet to hear any two brands (or models within a brand) of loudspeakers that sound anything alike, so we have to accept that none of this is going to be perfect. When it comes time to construct a sound system for accessing sound quality (or simply for musical enjoyment), I will most certainly use a single sub as you suggested. But for my proposed system, the subwoofer's crossover frequency and filter order does become something of a choice based on speaker performance. Because I will be filtering/processing the four B-formatted wave files before decoding (none of the processing will be done in real-time), I have a lot of choices for filter types--and perhaps the addition of group delay. I have numerous MATLAB Toolboxes for processing wav files in addition to the Advanced Signal Processing and Digital Filter Design Toolkits in LabVIEW. Thankfully, I’m no longer limited to the *bouncy* 8th-order elliptic filters I used to construct. The problem nowadays is that there are way too many choices that are relatively easy to implement. Responses to my last post provide clearer direction--thanks. Jörn brought into the discussion not only filter type and slope, but the choice of audio interfaces. I have first-generation MOTU 896HD units. Although there are two FireWire ports on each unit's backside, MOTU states that the only two units can be synchronized using a FireWire (daisy chain connection), and the combination of more than two interfaces requires a master clock (the units have word clock in and out). There is an optical interface in addition to the AES Pro (XLR). My gear is currently out of reach, but I’m guessing that the optical connect is intended for ADAT/Lightpipe, not optical S/PDIF (they're two distinctly different protocols and not interchangeable). A separate 8-channel A-D could certainly be used with ADAT. I suppose there’s no reason to worry about inter-channel timing issues when using dissimilar components (meaning a MOTU optically linked to an D-A device). Similar to the MOTU interface, my M-Audio ProFire 2626 provides a lot of input and output options (two ADAT ports), but a D-A converter would still be needed for > 8 analog outs. I like the robustness (and XLR) connects of the MOTU 896; I’ll admit that much of this is a personal choice but it's not meant to promote or discount any single piece of gear. When it comes to configuring hardware and software, I don’t know whether all DAWs provide the option of assigning tracks to all of the physically available ports (for example, one of my USB interfaces permits a choice of digital OR analog, but not both simultaneously). Furthermore, I want the presentation of stimuli to be glitch-free. I imagine most modern high-end DAWs and interfaces provide crash-free performance, but mixing 48 mono tracks to stereo isn’t the same as providing 48 discrete analog out channels when it comes to stable performance. Perhaps my fear of computer crashes (both Mac and PC) comes from past experiences. What I'm using now seems glitch- and crash-free, hence my desire to stick with it (and the 16-channel count). Regarding speaker arrays: Thanks, Jörn, for suggesting a large (ear-level) ring with smaller rings above and below the larger ring. My original idea (two large rings) stemmed from the notion that the 12 speakers would lie on the surface of a large (virtual) sphere whose poles would extend beyond the room dimensions, thus giving the impression of a *bigger* listening space. Of course, a sense of distance and spaciousness is intrinsic to the recording, not how far the actual speakers are from the listener (well, we could get into a wave-front curvature discussion, but I’m not ready for that). The other reason had to do with placement of video monitors. If the video doesn’t interfere with the speaker array, I’ll make drawings for the speaker layout you suggested. Thanks. I may have access to a 10 x 20 foot room. I don't know the ceiling height. To my knowledge, all surfaces are treated with 6- or 8-inch foam. From what I've been told, the room is practically anechoic down to 200 Hz. Maybe the leftover space could be used for bass traps, aborbers, or diffusers to further tame the low frequencies. I might have access to a B&K intensity probe for calibration. It would be interesting to look at the velocity and pressure components resulting from the surround system as measured from the listening position. This could give some measure of wave field accuracy that goes far beyond perceptual judgments. RE mics: The idea of going *second-order* occurred to me, but I’m too ignorant on this topic to say much. I’ve overheard discussions (mostly at trade shows) where first-order mics could be *stretched* to give second-order performance (and further stretched to give HOA performance or approximations). As I understand, Ambisonics can be enhanced by the addition of U and V components, and these are extensions of the B-format components derived from a first-order mic (based on the raw, or A-format, data). From what I've read, U and V contribute to horizontal, not vertical, image stability. With regard to live recordings, only a HOA-specific mic (e.g., VisiSonics or Eigenmic) can provide true HOA performance. Adding more playback channels to recordings obtained with a first-order mic may give better room coverage/distribution, but nothing is gained in terms of accurate *wave-field* reconstruction. Am I correct here? Although I could create higher-order stimuli via modeling, the plan is to use complex and dynamic stimuli (that includes moving objects such as automobile traffic) obtained via live recordings. The emphasis is on *real-world* reconstruction of representative environments, not listening assessments based on movie-goers expectations (movie sound designers provide us with ample conditioning; heck, you can even hear sounds in the vacuum of deep space!). As always, many thanks for your time and input. Best, Eric C. (by the way, I started putting C here so that readers don’t confuse me with the Erics who are 10X smarter than yours truly). -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://mail.music.vt.edu/mailman/private/sursound/attachments/20130221/c45ed548/attachment.html> _______________________________________________ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
