Re: [Sursound] 1. Re: Ambisonics setup overview
Hi Darren vbap and max is commonly used but my first attempts used ploguebidule and the vst plugins so I guess audiomulch would allow the same thing - to input your ableton audio add ambisonic panning and output to a bformat file for playback or add a bformat decoder on the end for live playback with speaker psoitions www.radio.uqam.ca/ambisonic/b2x.html Daniels plugins work for me with decent speaker positioning mick On 17 Mar 2011, at 17:25, Darren - Bradley wrote: Thanks for all your info: I looked into the basic /// max eternal ambipanning~ /// by www,icst.net It says " ambisonic equivalent panning (without intermediate B-format) I opened the help file and looks like it might work quite well thanks for all your support Darren ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] horizontal-only decoder design for line sources
fons, aaron, thanks for your replies. On 03/18/2011 01:51 AM, Fons Adriaensen wrote: To behave as a line source rather than a point, they have to be larger than the wavelength, this imposes a lower frequency limit. On the other side, once the wavelenght is comparable to the distance between the drivers they will start to appear as a collection of discrete sources, and this limits the range at the high end. Combining these two limits results in a rather narrow frequency range for any practical array. well, with a 3.5m array (which is quite a standard length), you're down to 100 hz, which is Pretty Darn Good. as for the upper limit, with some intelligent waveguiding and other magic fairy dust, you can get the individual elements to emit something close to a plane wavefront already. which means they will combine in a more effective way. hence, hf level loss due to dispersion over distance is less of an issue in practice than the theory (assuming omnidirectional tweeters) would imply. in fact, for many systems the opposite is true: the hf beam has an annoying tendency to constrict, resulting in even more focused coverage on-axis than desired. And anyway the extent of the near field is proportional to the array size - once you are far enough they will again appear as a point source. true, but there is still great potential. line arrays can be made to work ok'ish over more than a hundred meters. of course, there is a more or less severe impact on timbre, but: concertgoers will perceive the gradual degradation as natural. it's not a near-field listening situation, we don't want to create an illusion of extreme proximity to the sound. a rocknroll concert heard at a distance of 100m will sound like a rocknroll concert heard at a distance of 100m, with perfect fidelity :-D so for practical purposes, line arrays do change the rules for large ambisonic rigs: * you don't necessarily have to resort to a "controlled opposites", strict cardioid decode, in order to avoid the collapsing of the image into the opposite speaker(s) for listeners close to those speakers, because the "good speakers" retain more level close to the "bad speakers". this will improve localisation in the inner listening area considerably. * with more uniform coverage over distance, we can create larger listening areas where speakers still contribute evenly to the sound field in terms of energy. * we can reach listening area sizes where all perceptual assumptions that hold for usual ambisonic reproduction break down completely, because while the speakers contribute evenly in terms of energy, they no longer do so with respect to time. It's not exactly true that rV/rE 'focus on the sweet spot', they are metrics that can be applied anywhere, and that apparently map well to perception (respectively for low and mid/high frequencies). what i meant is: we tacitly assume that the contributions of all the speakers will blend within a given perceptual window. phase sensitivity corresponds to the LF band, and more or less to the effect called summing localisation, with a time difference window of < 1ms (depending on frequency). rE sensitivity covers the rest, but only if the contributions of the individual speaker stacks arrive within the "haas window" < 30ms, i.e. before you begin to discern them as separate auditory events. large arrays will have areas where these assumptions no longer hold. so there are 2 area constraints: * the area of perfect reproduction (aka sweet spot), which is quite small, a function of the frequency and order, and which allows us to optimize for phase and energy gradient separately. * a larger area of usable reconstruction, where rV is no longer meaningful due to the phase errors introduced by run-time differences, but where plausible rE recombination still happens. still a function of the ambisonic order. this is where current experience with ambisonics ends (unless yours doesn't, in which case i want to hear from you!). large systems introduce a third area, where runtime differences are so large that you cannot expect meaningful recombination of the individual speaker stacks to one coherent image, at least not according to the same perceptual mechanisms we assume for small systems. this is what i'm currently thinking about: where is the boundary at which ambisonicitiness breaks down for good, regardless of order? In case you have a 'pre-echo' not corresponding to the source direction, you could still apply the rE metric to each of the 'pre-echo' and the 'main' part separately - I guess this would be the first step in analysing such a situation. very good point. so i could probably start by analysing the pre-echo and the "correct" cue separately, treat those as two loudspeaker sources, and apply some known theory about directional masking (i.e. how much louder does a second sound event need to be in order to mask the directional cue from a correlated earl
[Sursound] Masters /PHDs related to surround sound / Sound art in Europe
anyone know of any Masters /PHDs related to surround sound / Sound art in Europe (but not the UK or Ireland - I am already aware of those) cheers, Gus On 19 March 2011 13:00, wrote: > Send Sursound mailing list submissions to >sursound@music.vt.edu > > To subscribe or unsubscribe via the World Wide Web, visit >https://mail.music.vt.edu/mailman/listinfo/sursound > or, via email, send a message with subject or body 'help' to >sursound-requ...@music.vt.edu > > You can reach the person managing the list at >sursound-ow...@music.vt.edu > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Sursound digest..." > > > When replying, please remember to edit your Subject line to that of the > original message you are replying to, so it is more specific than "Re: > Contents of Sirsound-list digest..." > > > Today's Topics: > > 1. Re: 1. Re: Ambisonics setup overview (mick ritchie) > > > -- > > Message: 1 > Date: Sat, 19 Mar 2011 10:36:27 + > From: mick ritchie > Subject: Re: [Sursound] 1. Re: Ambisonics setup overview > To: Surround Sound discussion group > Message-ID: > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > Hi Darren > > vbap and max is commonly used but my first attempts used ploguebidule > and the vst plugins > so I guess audiomulch would allow the same thing - to input your > ableton audio add ambisonic panning and output to a bformat file for > playback or add a bformat decoder on the end for live playback with > speaker psoitions > > www.radio.uqam.ca/ambisonic/b2x.html > > Daniels plugins work for me with decent speaker positioning > > > mick > On 17 Mar 2011, at 17:25, Darren - Bradley wrote: > > > Thanks for all your info: > > > > I looked into the basic /// max eternal ambipanning~ /// by > > www,icst.net > > > > It says " ambisonic equivalent panning (without intermediate B-format) > > > > I opened the help file and looks like it might work quite well > > > > thanks for all your support > > > > Darren > > > > ___ > > Sursound mailing list > > Sursound@music.vt.edu > > https://mail.music.vt.edu/mailman/listinfo/sursound > > > > -- > > ___ > Sursound mailing list > Sursound@music.vt.edu > https://mail.music.vt.edu/mailman/listinfo/sursound > > > End of Sursound Digest, Vol 32, Issue 17 > > -- 07812675974 -- next part -- An HTML attachment was scrubbed... URL: <https://mail.music.vt.edu/mailman/private/sursound/attachments/20110319/fea44f57/attachment.html> ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] horizontal-only decoder design for line sources
On Sat, Mar 19, 2011 at 05:50:15PM +0100, Jörn Nettingsmeier wrote: > well, with a 3.5m array (which is quite a standard length), you're down > to 100 hz, which is Pretty Darn Good. You won't have much directivity for 1 wavelenght - first zero at 90 degrees. > as for the upper limit, with some intelligent waveguiding and other > magic fairy dust, you can get the individual elements to emit something > close to a plane wavefront already. The question is _how_ close compared to the wavelenght. Simple fact is this: a source that produces vertical cylindrical waves does not have any vertical directivity and has a frequency response going down by 3 dB/oct compared to the elementary sources is it composed of. PA arrays do not behave like that at all. > true, but there is still great potential. line arrays can be made to > work ok'ish over more than a hundred meters. Yes, but they are just point sources at that sort of distance. > * you don't necessarily have to resort to a "controlled opposites", > strict cardioid decode, in order to avoid the collapsing of the image > into the opposite speaker(s) for listeners close to those speakers, > because the "good speakers" retain more level close to the "bad > speakers". this will improve localisation in the inner listening area > considerably. This is *always* the case, even for point-source speakers. In-phase decoding optimizes in one direction - the one opposite to the source. Max-rE provides a more global optimisation. For example it will do a better job for a listener close to a left or right speaker for a front source. Also consider that even at just 3rd order, the 'back lobe' for max-rE is down something like 28 dB, and this only improves as order goes up. > * with more uniform coverage over distance, we can create larger > listening areas where speakers still contribute evenly to the sound > field in terms of energy. > > * we can reach listening area sizes where all perceptual assumptions > that hold for usual ambisonic reproduction break down completely, > because while the speakers contribute evenly in terms of energy, they no > longer do so with respect to time. Again, the extended cover is the result of carefully shaping the vertical polar pattern of the array over the range where it intersects with the the part of the horizontal plane where the listeners are. It is possible because 'close' listeners are in a differenct direction (as seen from the array) compared to 'distant' ones. It is not the result of of any 'near field' effect, or cylindrical waves. Ciao, -- FA ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
