Re: [Sursound] Bi-Amping the B-format
On 02/20/2013 09:20 PM, Eric Carmichel wrote: Greetings to All: The system is comprised of 16 channels. The advantage of 16 (or less) channels is that I don’t have to use a dedicated Word Clock to sync my MOTU FireWire interfaces. I’m fond of these interfaces, and two of them can sync-up via the FireWire link without complication. i don't know which MOTUs you're using, but don't most of them include ADAT outs? so you could easily get 8 extra outs per MOTU without any sync hassles, and just the minor investment into an 8ch DA converter. I intend to use two hexagonal arrays of small-sized loudspeakers (a bit larger than the 3-inch coned Genelecs, but not much more so). One array will be near floor level, whilst the second array is proximal to the ceiling. a few years ago, i was more or less forced to implement such a setup because of a frontal video screen. it works, but it's not optimal for horizontal sources. if you don't have any other constraints, i guess it's advisable to start with a horizontal hexagon and add two rings of three speakers each at +/-60° elevation. your usecase seems to be a single listener in a fixed position, so my advice may not apply, but at least for moving listeners, i can say that rings near ceiling and floor are not as stable and easygoing as a main horizontal ring plus smaller numbers of high and low speakers. According to the literature (Malham, Rumsey, and others come to mind, but I’m shooting from the hip), diametrically opposed pairs may be preferred when the listener is centered in the array. Further, it is purported that six speakers provide immunity against drawing signals toward a single speaker. BLaH2, iirc. to add some anecdotal evidence: for me, a hexagon is also far superiour to a square for first order. plus it offers the possibility of going for second order - not an option for your recorded soundscapes, but maybe interesting for synthetic cues, which can be rendered more reliably than in POA. Eight speakers is probably overkill and doesn’t leave me the four channels needed for a square array of subs. for first order and a single listener, certainly. for artificially panned cues, maybe not. Any thoughts as to whether the two hexagonal arrays providing horizontal and height information should be offset or vertically aligned? no. i've only ever used aligned rings. Regarding the need for subs: With ‘normal’ music content, twelve speakers working in concert would provide more than adequate low-frequency energy. But I’m going to be using live recordings where a particular low-frequency sound could be coming from an extreme R, L, front or back direction. In this scenario, I’d rather have the subs handle the load but I still need to preserve ‘direction’ as stated above. Because four speakers can provide adequate surround sound, my intent is to frequency-divide the B-formatted signal and send the highs and lows to their respective feeds via ‘conventional’ Ambisonic decoding. To be clearer, I will digitally filter the B-format signal so that each of its four components (W, X, Y and Z) are divided into a high and low-frequency signal component. The low-frequency components will be decoded and sent to the square (and likely horizontal) array of four subs. The highs will be decoded based on the position of the 12 ‘full-range’ speakers. I use full-range loosely here because the added bass channels aren’t for enhancement, but to alleviate the 12 speakers from their low-end duty. that approach works very well, i've used it several times to good effect. I haven’t determined the best crossover frequency, and this may be determined in part by a combination of the speakers used and the stimuli to be presented. I wish to use the lowest possible frequency, but not to the point of driving the small speakers to distortion. I’m guessing a digital (crossover) filter that is both maximally flat and phase coherent is best, though slight dips caused by frequency response anomalies are easy to EQ out. I use EQ judiciously because it is generally just a marginal cure for a loudspeaker's deficiencies. Upping the response at some frequency extreme merely adds to distortion that is ‘measured’ (in SPL) as a boost at the deficient frequency (or third-octave band or whatever). Only a spectrum analyzer or critical listening reveals where the real boost is occurring. and then there's the room... for normal p.a. use with small stacks, i like shallow filters. but if you can't get the physical alignment correct everywhere (as is the case when the listening area is large and the subs are at a non-negligible distance from the tops), you might want to reduce overlap as far as possible, because it will be wrong almost everywhere, regardless of your time alignment. hence, i'd argue for 24db/oct linkwitz-riley. in theory, 8th order (48dB/oct) should be even better, but there may be other problems in using those, and i haven't had the chance to do an a/b
Re: [Sursound] Bi-Amping the B-format
>> >> The system is comprised of 16 channels. The advantage of 16 (or less) >> channels is that I don’t have to use a dedicated Word Clock to sync >> my MOTU FireWire interfaces. I’m fond of these interfaces, and two of >> them can sync-up via the FireWire link without complication. > > i don't know which MOTUs you're using, but don't most of them include > ADAT outs? so you could easily get 8 extra outs per MOTU without any > sync hassles, and just the minor investment into an 8ch DA converter. > Did wonder if he had some new card ... I use 8-analogue + 8-ADAT from one MOTU (think you can squeeze a few* more out as well). Secondly, mine has tow firewire ports and you can daisy-chain ... with sync. But ... then again ... perhaps he has some new model ... (?) . Michael 22 comes to mind 16+ ? stereo 'headphones' ? stereo AES/EBU ? ? maybe 20? ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Bi-Amping the B-format
Eric Carmichel wrote: ... > Eight speakers is > probably overkill and doesn?t leave me the four channels needed for a square > array of subs. If you have four subs then you might want to consider arranging them in a tetrahedron. Note that subs are heavy, so there is a practical problem with mounting them up in the air. (Arranging four full-range speakers in a tetrahedron is not recommended.) Whatever arrangement you use, because the frequencies are much less than 700 Hz, you only need to drive them from a single-band "vector" decoder. Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/ ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Bi-Amping the B-format
Eric Carmichel wrote: ... > I intend to use two > hexagonal arrays of small-sized loudspeakers (a bit larger than the 3-inch > coned Genelecs, but not much more so). ... > Because four speakers can provide > adequate surround sound, my intent is to frequency-divide the B-formatted > signal and send the highs and lows to their respective feeds via > ?conventional? Ambisonic decoding. To be clearer, I will digitally filter > the B-format signal so that each of its four components (W, X, Y and Z) are > divided into a high and low-frequency signal component. The low-frequency > components will be decoded and sent to the square (and likely horizontal) > array of four subs. The highs will be decoded based on the position of > the 12 ?full-range? speakers. I use full-range loosely here because the > added bass channels aren?t for enhancement, but to alleviate the 12 speakers > from their low-end duty. Driving your 12 speakers from only a single-band "energy" decoder may be a problem. For a conventional Ambisonic decoder, the "crossover" frequency is 700 Hz. This corresponds to a wavelength which is twice the distance between your ears, and is where you ear/brain switches from phase/time cues for localisation to amplitude cues. However, in practical domestic decoders, the transition frequency is lowered to 400 Hz to better accommodate off-centre listeners. Also, in large area decoders, the transition frequency is often ignored, and a single-band "energy" decoder used. This transition frequency is well above the typical crossover frequency used with subs. Whether this is a problem for you depends on the size of the venue as well as the frequency range of your 12 "full-range" speakers. It is possible you may have to drive the 12 speakers from a dual-band decoder. Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/ ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Bi-Amping the B-format
On Thu, Feb 21, 2013 at 12:13:54PM -0700, Martin Leese wrote: > If you have four subs then you might want to > consider arranging them in a tetrahedron. > Note that subs are heavy, so there is a > practical problem with mounting them up in the > air. (Arranging four full-range speakers in a > tetrahedron is not recommended.) > > Whatever arrangement you use, because the > frequencies are much less than 700 Hz, you > only need to drive them from a single-band > "vector" decoder. Indeed, the tetrahron works with subs because you can drive them using a systematic aka max-rV decoding. Regarding the question if the two rings should use the same azimuths, or one of them should be offset by 30 degreees: it doesn't matter. That is so because the two horizontal rings don't have any preferred directions or 'speaker detent' - the resulting rV or rE is the same for all azimuths. So 'interpolating' between the two rings will always produce the same result, regardless of the relative speaker positions. This is why an arrangement like 1 + 6 + 8 + 6 + 1 (at -90, -45, 0, +45 and +90 degrees elevation respectively) works well for full 3rd order 3D: any vertical circle has 8 equally spaced intersection points with the horizontal rings and zenith or nadir speakers, so it will support 3rd order elevation perfectly. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] A Thanks... And Another Post
nit'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.mu
