Re: [Sursound] Patents, Serendipity, and Questions
On Wed, Dec 21, 2011 at 12:51:45PM -0800, Eric Carmichel wrote: > 1. Is there any preferred method of calibrating speakers used > in an Ambisonic setup? If you have a decent set of identical speakers there is in general little reason to do this - or at least it's not a priority compared to getting other aspects (see below) right. Now your application context is not the usual one of listening for entertainment and that may change the picture. Whatever method you use to measure/EQ the speakers, beware of any form of 'extreme' or very detailed (in the F domain) EQ. For Ambisonics to work well you want the speakers to remain matched, including phase response up to a few kHz. As a measuring method I'd use a log sweep with deconvolution to an IR. Then compute the (complex) inverse response with normalisation - don't try to correct anything that would require lots of EQ. > 2. Has anyone compared or noted differences between the Virtual > Visual Microphone (VVM) software and offline processing using > MATLAB? There is no reason to use off-line processing. AMB decoding done in real time takes 1% or so of CPU load on a modern PC. If you think of speakers feeds as directional microphone patterns then these patterns should be frequency dependent. This is particular true for small diameter rigs, and related to some of the things discussed below. VVM doesn't provide this AFAIK, and I would not recommend it as an AMB decoder. > 3. I have seen discussion and articles regarding Ambisonics and > shelving filters. Any recommendations as to "best" filter settings > based on speaker-to-listener radius? The issues of number of speakers, shelf filters and near-field effect compensation, while not directly related to each other, can be understood only by looking at a bit of theory. AMB, if reproduced using the so-called 'systematic' decoding matrix, reconstructs the sound field in a limited area. The size of this area is measured in wavelengths, so it can be very large at LF and will be very small for HF. For first order, the radius of this area is around half the wavelength, and it increases for higher order. Now if both the listener's ears are within that 'area of reconstruction' all directional cues will be the same as they were in the original soundfield. But this possible only up to a few hundred Hz, and even less if you want a larger listening area. Outside the 'area of reconstruction' the sound field produced by a systematic decode (complex interference patterns, since the systematic decode will typically use nearly all speakers even for a single source) is not one that works well. So even for a single listener system above 700 Hz or so something else is needed. The optimal solution here is a decoding that optimizes the magnitude and direction of the energy vector, (the vector sum of the energies from each speaker) known as the 'max rE' decoding. Note that the crossover between the two regimes matches the frequency range where inter-aural phase differences become ambiguous and where our directional hearing switches to using amplitude differences instead. So a good decoder needs to be frequency-dependent. There are in practice two ways to implement this. The first is to use the same matrix for both regimes, and use phase- matched shelf filters on the input (B-format) signals to modify the effective matrix coefficients. This works well for regular layouts. The second one is to use phase matched crossover filters and two fully independent matrices. This is required for non-regular layouts (e.g. 5.1 and most 3-D installations which are almost never regular). Let's now look at the number of speakers. It has already been mentioned that eight speakers for horizontal-only first order is too much. This is absolutely true, you shouldn't use more than six in that case, and even that is a compromise (but a good one). To understand this you need to look at the radial dimension. As you mave away from the 'sweet spot', the soundfield reconstruction depends more and more on higher degree spherical harmonics. This is the reason why the 'area of reconstruction' is limited in the first place. The mathematical expression of this is the Fourier-Bessel sum. With a first order input obviously only the zero and first degree components can be reconstructed correctly. So what about the higher ones ? In the sound field of a 'real' source they are present. In an AMB rig they will be present by spatial aliasing. Think of your ring of speakers as samples on a periodic waveform to understand this in an intuitive way. A ring of eight speaker driven by an AMB decoder will reconstruct the (horizontal) components up to 3rd order plus one of the two 4th order ones (in exactly the same way as eight samples on a periodic waveform allows up to the 3rd harmonic and 'half' the 4th). If the input is just first order, the 2nd and 3rd order components (and their higher order aliases) are forced to zero. And in practice that is worse than allowi
Re: [Sursound] Patents, Serendipity, and Questions
> On Wed, Dec 21, 2011 at 12:51:45PM -0800, Eric Carmichel wrote: > > >> 3. I have seen discussion and articles regarding Ambisonics and >> shelving filters. Any recommendations as to "best" filter settings >> based on speaker-to-listener radius? > > Very few AMB decoders provide both dual-band operation > (or shelf filters) and near-field compensation. I'm > the author of one that does (Ambdec), but since I > suspect you are on Windows you can't use it. I'd give a big vote (and thanks) for Ambdec. Having also (this Spring/Summer) got bogged down in cross-platform (dual operating system) 'problems', I would also emphasise that many are highly soluble. You can take a selection of B-format recordings/files and 'cook them' to speaker feeds using Ambdec on either MacOS or Linux. (For Linux you can even install dual boot on a MS machine, rather than borrowing cycles of a friend's Mac / Linux box.) Save the speaker feeds as a six (or four (but preferably not eight;-)>) channel file ... and 'bring it back home' for direct playback on a MS box. Just an idea, Michael ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Patents, Serendipity, and Questions
Happy Holidays from Raymond Ward, Burlington, NC Christmas song's now up at http://www.reverbnation.com/#!/raymondward -Original Message- From: Michael Chapman To: Surround Sound discussion group Sent: Thu, Dec 22, 2011 8:32 am Subject: Re: [Sursound] Patents, Serendipity, and Questions > On Wed, Dec 21, 2011 at 12:51:45PM -0800, Eric Carmichel wrote: > > 3. I have seen discussion and articles regarding Ambisonics and > shelving filters. Any recommendations as to "best" filter settings > based on speaker-to-listener radius? > Very few AMB decoders provide both dual-band operation (or shelf filters) and near-field compensation. I'm the author of one that does (Ambdec), but since I suspect you are on Windows you can't use it. I'd give a big vote (and thanks) for Ambdec. Having also (this Spring/Summer) got bogged down in ross-platform (dual operating system) 'problems', I ould also emphasise that many are highly soluble. You can take a selection of B-format recordings/files nd 'cook them' to speaker feeds using Ambdec on ither MacOS or Linux. For Linux you can even install dual boot on a MS achine, rather than borrowing cycles of a friend's ac / Linux box.) ave the speaker feeds as a six (or four (but preferably ot eight;-)>) channel file ... and 'bring it back home' or direct playback on a MS box. Just an idea, Michael __ ursound mailing list urso...@music.vt.edu ttps://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: <https://mail.music.vt.edu/mailman/private/sursound/attachments/20111222/94045e06/attachment.html> ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] Thanks for links, insights, etc.
Greetings: Hello Michael C. and Fons A., Thank you for your detailed and informative responses to my questions. Fortunately, the speakers I have chosen are well-matched and have good response characteristics. I matched them some time ago; however, each speaker underwent testing at an identical location, not at their respective positions in my listening room. Because I am interested in three-dimensional Ambisonics, four of the eight speakers in the (current) octagonal array will have to be close to floor level: This is the only way to get moderately wide vertical separation without putting the listener in a high chair. I recently observed that speaker response (independent of room characteristics) changes because the floor imparts an affect (I believe more than just the proximity effect). Fortunately, large amounts of EQ aren’t needed, and I’m mostly interested in smoothing the response in the 100 Hz to 10 kHz range. I’m a minimalist when it comes to audio. I was never one to use graphic EQs (or modern-day VSTs to achieve the same). I began building amplifiers while in grade school, and a 10 watt, class-A amp designed by J. Linsley Hood and described in Wireless World (1969-ish?) was a favorite of mine for many years. Later I built a class-A, push-pull VT amp with 300Bs and an interstage transformer. This was for my Lowthers. I never got into the single-ended stuff because it seemed easy to mitigate transformer core saturation issues with class-A push-pull designs that operated along the transfer characteristic as SE biasing. My point is this: I don’t like too many things in the circuit path, and I only use EQ when absolutely necessary. However, measurements serve to “validate” my research findings, particularly when they’re slated for publication or under scrutiny. If I use EQ, I try to use filter types that yield the best transient characteristics and minimal phase anomalies. I downloaded, as per your suggestions, the PowerPoint / PDF by J. Nettingsmeier. Looks like really good information. I will give it a thorough reading after Christmas. Thanks for recommending. RE MATLAB: Some of the cochlear implant (CI) simulations I do are simple phase vocoder scripts written in MATLAB. While in graduate school, my doc committee consisted of respected researchers (does W. Yost, M. Dorman, or S. Bacon ring a bell with anybody?) who were huge proponents of MATLAB. The general attitude was “if you can’t do it in MATLAB, it isn’t worth looking at; furthermore, if it requires hardware, we don’t even want to look at it.” Kind-of strange attitudes in my book, but I’ve always been more of a hardware person, whether it’s digital or analog. I continue to do off-line wav processing in MATLAB because I can show the underlying math as well as the statistical outcome. More recently, I’ve been using Visual FORTRAN for projects. RE Linux: I’m mostly a PC (Windows) user, but I’m not one to argue about the superiority of one OS over another. I have a BIG investment in software, and I don’t want to buy two versions of everything. It’s bad enough keeping up with the latest Adobe media suite or incarnation of Windows. I’ve mostly stayed with PCs so that I get best support for my National Instruments DAQ hardware or other (legacy) devices. Because I have several computers, setting one up with Linux is no problem at all. I used to run Red Hat Linux on one machine, and I really did believe in the superiority of Macs when Windows 98 repeatedly crashed. Nowadays I’ll use what works best or is accessible. So that I can experiment with Ambdec, I’ll load Linux on a dedicated hard drive. My audio hardware consists mostly of MOTU FireWire interfaces, but I also have an Avid PC extension chassis that has four identical PCI SoundBlaster cards on it. I’m sure I can find ASIO drivers for Linux that will work with my MOTU gear. The SoundBlaster cards are generic enough to work with about any OS (maybe even OS2 Warp). I’ve been duly warned of the consequences of using more than six loudspeakers in a horizontal-only, first-order Ambisonic configuration. Thanks, Fons, for the very clear explanation. I do, however, want a flexible system because I’d like to move towards a 3-D setup (or higher-order Ambisonics via recordings made with an mh acoustics Eigenmic). Additionally, I have plans for an experiment that compares energetic versus informational masking of vocoded speech in the sound field, and I’ll be using two quasi-independent 4-channel systems for this. When it comes to music enjoyment, I’ll stick with your recommendation of six loudspeakers. Again, many thanks to all for the help! Sincerely, Eric -- next part -- An HTML attachment was scrubbed... URL: <https://mail.music.vt.edu/mailman/private/sursound/attachments/20111222/dec78d4f/attachment.html> ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.
