Re: [Sursound] A higher standard of standardness
At 06:31 3/7/2013, Robert Greene wrote: Variations from reality ought surely to be based on knowing how to reproduce the reality first and then introducing the variations. One does not bend pitches for artistic effect until one is able to play in tune, so to speak. Yes, indeed; but such question begging exposes the problem per analogiam. What does one define as in tune? What you are asking for is the ability to reproduce a complete soundfield with 100% accuracy, and then to introduce variations. We have not yet progressed to this level. If people want to treat recording as a pure art form where one simply judges the results on aesthetic grounds. it would be hard to say that was wrong. But it surely takes recording out of the realm of science. I am not sure that many of its practitioners (even Blumlein) regarded recording as a science: it is rather an exercise in engineering combined with aesthetics and as such intrinsically hard to theorize about. To my mind, offensive or no, it remains startling to me that there is no recorded demo of how various stereo mike techniques reproduce the sound of a pink noise source at various spots around the recording stage, for example. I cannot imagine that anyone would want to listen to a CD of pink nose or that anyone can believe that objective determinations can be made by doing so for longer than a few minutes. The ear adjusts to what it is hearing, as the eye does to colours under different lighting conditions and there is no equivalent to grey cards for white balance. Even doing A/B comparisons with the flick of a switch is fraught with self-deception, unless the levels are controlled and enough time is allowed to accustom oneself to A before assessing B. Surely people might want to know whether the mike technique was changing the perceived frequency response of sources depending on where the sources were? How can people NOT want to know this? There is a book by Jürgen Meyer (Acoustics and the Performance of Music). The blurb on Amazon says: This classic reference on musical acoustics and performance practice begins with a brief introduction to the fundamentals of acoustics and the generation of musical sounds. It then discusses the particulars of the sounds made by all the standard instruments in a modern orchestra as well as the human voice, the way in which the sounds made by these instruments are dispersed and how the room into which they are projected affects the sounds. I have had this book for over 30 years. It contains polar diagrams of most orchestral instruments plotted for different frequencies. Nobody that I know has ever found much use for the data in making a recording, beyond those generalizations that are obvious to the ear. I agree with EC that a complete analysis of the relationship between recording and musical sound would be a tremendous task, perhaps one that is not even well defined. I think that is a conceit: there are far too many independent variables and the exercise would probably become what Glen Gould would describe as centipedal. This is how science works. One works out simple cases first. The fact that no one knows if there are infinitely many primes pairs with difference 2(eg 17 and 19) does not make it irrelevant to know that there are infinitely many primes. One answers simple questions first. Again: recording is not a science. If anything it is a craft with elements of engineering. I have been teaching it for over 30 years at university level and the number of textbooks that are of any use whatsoever, and those with caveats, can be counted on one hand. Take, for instance, the excellent book on Stereo by Streicher: most of the information is either theoretical (e.g. the combination of unrealizable polar diagrams) or else cannot be used without extensive empirical experimentation. Personally, I would just like to know which mike technique does what to the tonal character of sources at different locations around the recording stage. If you don't care, you don't care. But I wish I had a disc where I could listen and find out. I find it hard to believe that other people are not interested in this. As I am sure you know, active listening is a very tiring process that most people are not trained to participate in. If one cannot identify differences within seconds it is best to take a long rest and try again much later. Few have the patience for this and professionals cannot afford the time when musicians are waiting to perform. Years ago I decided to learn the piano(I am a violinist!) just to see how it would go, by learning the Rachmaninoff 3rd piano concerto --a measure at a time. As you can imagine I did not get very far! (the first statement of the theme went ok but soon, no soap). Of course this was a joke! I knew from experience of learning to play the violin that one learns the basics step by step and builds up to the complex
Re: [Sursound] A higher standard of standardness
On 3 July 2013 07:37, David Pickett d...@fugato.com wrote: At 06:31 3/7/2013, Robert Greene wrote: Variations from reality ought surely to be based on knowing how to reproduce the reality first and then introducing the variations. One does not bend pitches for artistic effect until one is able to play in tune, so to speak. Yes, indeed; but such question begging exposes the problem per analogiam. What does one define as in tune? What you are asking for is the ability to reproduce a complete soundfield with 100% accuracy, and then to introduce variations. We have not yet progressed to this level. And I doubt if we will ever even begin to be able to do this. Ambisonics theoretically can get it exactly right at just one point (and does get close-ish) but it is still relatively far off when you factor in the effects of the microphones timbrally (because of the individual characters of the capsules), in terms of the disturbance of the sound field in the original space by the presence of the microphone and because of the non-coicidence of the capsules. Then, of course, theres the limitations of the loudspeakers and the playback space acoustics. If you want to assess, in a controlled manner, what is the best recording technique, this would be a huge experiment since it would surely be different for every type of music - and every different ethnic group, multiplied by their different life experiences - stay-at-home, traveller, immigrant (multiplied by their generation), refugee, etc., differentiated by their experience of music, both short (did they go to a nice concert last night?) and long term (do they go to concerts/gigs/discos regularly?). This doesn't even begin to include things like the fact that the current generation of students find it difficult to hear (some of the) obvious defects in recorded sounds that us antiques find glaringly obvious because they have grown up listening to mp3's and have had to learn how to tune out the rubbish generated by the encoding/decoding algorithms. But, assuming someone manages to get a sufficiently, stupendously, ginormously huge grant and actually want to take this on, here are some suggestions; Under no circumstances use artificial test signals like pink noise for the main work - use these only for later, detailed work if absolutely necessary. Record real sounds anechoically and replay them from the best possible loudspeakers, one per sound source** on stage in an appropriate venue (concert hall for DWMM, outside in a square for Gamelan, in a pokey little club for jazz and so on) which, importantly, will also have to have an excellent replay room attached to it. In turn, set up the differing record/replay set ups, recording the same piece of music each time. Have your experimental subjects listen to the live replay in the venue and the recording of it in the replay room. Now repeat the experiment with the same venues, music and equipment but different replay rooms n order to remove the effects of the replay room acoustics and other perceptual effects like comfort and lighting. Repeat until you can't stand it any more then analyse listener preferences. After five years and several million euros you might have some answers. Dave. PS I am available for consultancy if one of you gets the grant, at appropriate rates, of course :-) ** I am aware this isn't perfect as ideally it should be the case that the speakers also reproduce the directional effects of the instruments but it is, at least (a) reproducible and (b) not a totally artificial signal. -- As of 1st October 2012, I have retired from the University. These are my own views and may or may not be shared by the University Dave Malham Honorary Fellow, Department of Music The University of York York YO10 5DD UK 'Ambisonics - Component Imaging for Audio' -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/9f46c0ae/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] A higher standard of standardness
On 03/07/2013 05:31, Robert Greene wrote: If people want to treat recording as a pure art form where one simply judges the results on aesthetic grounds. it would be hard to say that was wrong. But it surely takes recording out of the realm of science. I am not sure that recording is a science per se. That's not to say that there isn't, or cannot be, or shouldn't be, such a thing as a science of recording, but it's not what most of us actually do. What we actually do is fundamentally artistic, though it uses an array of more or less technical tools and relies on a good deal of engineering to produce those tools. This of course is true of virtually any art: all rely on some kind of technology, whether it's what makes a hammer hit a tuned string or the materials that are combined to make a paint of a certain colour. But technology is not simply applied science, and in these areas we are not, generally, interested in how it works (the science) though those who make the tools no doubt are: we are interested in how it can be worked - how you use the tools to get what you are looking for, and then, most importantly the art of using them to get something that communicates emotionally and effectively at the end. If we are communicating emotion, there is a path along which that emotion travels. Perhaps it is from the performance of musicians in a certain acoustic environment, captured in a certain way and designed to be listened to in a certain way, as determined at least partially by the musicians and the team in the control room. If they decide, arbitrarily or otherwise, that what they are hearing (and thus, ideally, what you will hear at home - the destination of that emotional communication) communicates the emotion they wish to communicate, then that's it - its closeness to what you might hear acoustically in the vicinity of the musicians is irrelevant as far as the emotion is concerned (although it might be relevant to the techniques used to create and capture the performance). --R ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] Ambisonics - Decoding 16 channels in DAW
Dear Members of Sursound, i am using the VVMicVst Plugin in Reaper for mixing and decoding my B-Format recordings. The plugin is limited to an output of 8 channels. For a new sound installation, I would like to decode to 16 channels (two circles of 8 speakers stacked). I know that I could use ICST for Max, but if possible in any way, I would to keep on working in a DAW. Are there any other plugins or tools available for this purpose (OSX) ? Any help would be greatly appreciated! Best, Moritz -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/2415a0a0/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
Dear Members of Sursound, i am using the VVMicVst Plugin in Reaper for mixing and decoding my B-Format recordings. The plugin is limited to an output of 8 channels. For a new sound installation, I would like to decode to 16 channels (two circles of 8 speakers stacked). I know that I could use ICST for Max, but if possible in any way, I would to keep on working in a DAW. Are there any other plugins or tools available for this purpose (OSX) ? OSX : Ambdec ... ? Michael You can input / output through a DAW if you use Jack. ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] A higher standard of standardness
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Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
hi everyone, thank you very much for your replies -- what i would like to achieve is playing a mix of a b-format recording combined with several mono- and stereofiles (have been doing this a lot, but only with a maximum of 8 channels). my mixing platform is reaper on osx. i am going to record a space with a soundfield mic and i would like to then make a simulation of it by setting up an array of 16 speakers. one speaker circle is on ear level, the other one above. i would like to use the second circle above to add height information to the ambisonic soundfield. as i can see now, adding a second instance of vvmic or harpex might not be suitable as it would generate two separate soundfields. (not sure if i am right here...) the b2x plugins seem to have a maximum of 12 outputs. ...i will look at ambdec but it does seem to need a lot of routing using jack. would the decopro vst plugin (http://www.gerzonic.net/) be a good choice for this purpose? thank you ! moritz Am 03.07.2013 um 15:38 schrieb Matthias Kronlachner: hi! you may just add an additional 8 channel track for a second instance of vvmicvst in reaper. send the 4 channel ambisonics signal to this newly created instance hosting vvmicvst, and route the outputs as you like. but if this approach gives you good decoding is another issue.. matthias On 7/3/13 1:37 PM, Moritz Fehr wrote: Dear Members of Sursound, i am using the VVMicVst Plugin in Reaper for mixing and decoding my B-Format recordings. The plugin is limited to an output of 8 channels. For a new sound installation, I would like to decode to 16 channels (two circles of 8 speakers stacked). I know that I could use ICST for Max, but if possible in any way, I would to keep on working in a DAW. Are there any other plugins or tools available for this purpose (OSX) ? Any help would be greatly appreciated! Best, Moritz -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/2415a0a0/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound * Moritz Fehr mobil: 01749231733 moritzf...@web.de www.moritzfehr.de -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/66f54ac4/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
...i will look at ambdec but it does seem to need a lot of routing using jack. Sixteen speakers need a lot of routing whatever you use ... said not to be unfriendly, just to emphasise I'm not sure I understand ;-) If you are worried about repeatedly having to connect everything, then IIRC the GUI's to Jack allow for a 'save this configuration'. Even without that Ambdec configuration allows for named connections (sorry it's along time since I set one up). Michael ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
I have done this--smbisonic decoded to 2 hexagons one above the other. I have also add a stereo mix into the decode. It is known as B+ ThomasChen In a message dated 7/3/2013 7:34:00 A.M. Pacific Daylight Time, m...@moritzfehr.de writes: hi everyone, thank you very much for your replies -- what i would like to achieve is playing a mix of a b-format recording combined with several mono- and stereofiles (have been doing this a lot, but only with a maximum of 8 channels). my mixing platform is reaper on osx. i am going to record a space with a soundfield mic and i would like to then make a simulation of it by setting up an array of 16 speakers. one speaker circle is on ear level, the other one above. i would like to use the second circle above to add height information to the ambisonic soundfield. as i can see now, adding a second instance of vvmic or harpex might not be suitable as it would generate two separate soundfields. (not sure if i am right here...) the b2x plugins seem to have a maximum of 12 outputs. ...i will look at ambdec but it does seem to need a lot of routing using jack. would the decopro vst plugin (http://www.gerzonic.net/) be a good choice for this purpose? thank you ! moritz Am 03.07.2013 um 15:38 schrieb Matthias Kronlachner: hi! you may just add an additional 8 channel track for a second instance of vvmicvst in reaper. send the 4 channel ambisonics signal to this newly created instance hosting vvmicvst, and route the outputs as you like. but if this approach gives you good decoding is another issue.. matthias On 7/3/13 1:37 PM, Moritz Fehr wrote: Dear Members of Sursound, i am using the VVMicVst Plugin in Reaper for mixing and decoding my B-Format recordings. The plugin is limited to an output of 8 channels. For a new sound installation, I would like to decode to 16 channels (two circles of 8 speakers stacked). I know that I could use ICST for Max, but if possible in any way, I would to keep on working in a DAW. Are there any other plugins or tools available for this purpose (OSX) ? Any help would be greatly appreciated! Best, Moritz -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/2415a0a0/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound * Moritz Fehr mobil: 01749231733 moritzf...@web.de www.moritzfehr.de -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/66f54ac4/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/3854bdfa/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
Moritz Fehr wrote: ... i am going to record a space with a soundfield mic and i would like to then make a simulation of it by setting up an array of 16 speakers. one speaker circle is on ear level, the other one above. i would like to use the second circle above to add height information to the ambisonic soundfield. You will have more success if one speaker ring is *below* ear level and the other above. Alternatively, if you need a ring at ear level, try three rings of, say, 4, 6, and 4 speakers. 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] Ambisonics - Decoding 16 channels in DAW
as i can see now, adding a second instance of vvmic or harpex might not be suitable as it would generate two separate soundfields. (not sure if i am right here...) I can't speak for Harpex (and I can imagine reasons why it might not work), but it should be no problem to use two copies of VVMicVST. Each output depends only on the inputs and the parameters for that output. A given output will not be affected by the azi/elevation/etc. of the other mics. Of course, what is correct is a question of the whole set, but that set can span several instances of the plugin. David P.S. the standalone Windows program VVMic supports 32 outputs if this helps ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
Hi Moritz, I've been building Ambisonic decoders in Faust, which can then be compiled into a variety of plugins, including VST, PureData, SuperCollider, and so forth. What you need sounds easy to do. Contact me directly ( hel...@ai.sri.com) and we can work out the details. Info about Faust here: http://faust.grame.fr Aaron Heller (hel...@ai.sri.com) Menlo Park, CA US On Wed, Jul 3, 2013 at 7:33 AM, Moritz Fehr m...@moritzfehr.de wrote: hi everyone, thank you very much for your replies -- what i would like to achieve is playing a mix of a b-format recording combined with several mono- and stereofiles (have been doing this a lot, but only with a maximum of 8 channels). my mixing platform is reaper on osx. i am going to record a space with a soundfield mic and i would like to then make a simulation of it by setting up an array of 16 speakers. one speaker circle is on ear level, the other one above. i would like to use the second circle above to add height information to the ambisonic soundfield. as i can see now, adding a second instance of vvmic or harpex might not be suitable as it would generate two separate soundfields. (not sure if i am right here...) the b2x plugins seem to have a maximum of 12 outputs. ...i will look at ambdec but it does seem to need a lot of routing using jack. would the decopro vst plugin (http://www.gerzonic.net/) be a good choice for this purpose? thank you ! moritz Am 03.07.2013 um 15:38 schrieb Matthias Kronlachner: hi! you may just add an additional 8 channel track for a second instance of vvmicvst in reaper. send the 4 channel ambisonics signal to this newly created instance hosting vvmicvst, and route the outputs as you like. but if this approach gives you good decoding is another issue.. matthias On 7/3/13 1:37 PM, Moritz Fehr wrote: Dear Members of Sursound, i am using the VVMicVst Plugin in Reaper for mixing and decoding my B-Format recordings. The plugin is limited to an output of 8 channels. For a new sound installation, I would like to decode to 16 channels (two circles of 8 speakers stacked). I know that I could use ICST for Max, but if possible in any way, I would to keep on working in a DAW. Are there any other plugins or tools available for this purpose (OSX) ? Any help would be greatly appreciated! Best, Moritz -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/2415a0a0/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound * Moritz Fehr mobil: 01749231733 moritzf...@web.de www.moritzfehr.de -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/66f54ac4/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/c547d59e/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] Round Arrays in Square Spaces: A Case of Iteracy
I was one of those kids who put round pegs in square holes. Out-of-the box thinking didn't apply. Now I'm one of those adults... Regarding recent posts: I don't think anybody wants to listen to pink noise unless you're performing the exercises in Dave Moulton's Golden Ear training. But recordings of Gaussian, weighted, and band-limited noise are highly purposeful--we all know this. Digital recordings of pink noise are even better than the old days of analog noise generators because we have a replicable reference that we can overlay or compare measurements to. On average, pink noise gives a predictable spectrum, but without a recording and known time reference, we can't repeat the EXACT same signal over and over--again, no news here. But here's something I wish to try (I've touched on this in past posts, but now my design is more concrete). Briefly, I propose a recording of a recording in order to validate *accuracy* of spatial reproduction. A human element need not be present (this ain't social science). By rotating my TetraMic on a fixture that permits rotation on its central axis (see figure in link below)**, I can use a single loudspeaker to create the equivalent of a circular array of n loudspeakers playing bursts of narrowband noise (or music, if you prefer). I use narrowband (octave or third-octave) noise in lieu of pink noise to improve the SNR. This recording will provide the initial B-formatted files of noise bursts. I'll arbitrarily rotate the mic in 60-degree increments for a total of 6 positions. Because a single speaker is being used, I only have to calibrate one speaker at one location. Regardless, I now have an equivalent recording of a 6-speaker, horizontal-only array. For playback, I will use a cubical array that consists of eight loudspeakers: four below the horizontal plane (plane as it passes through the mic) and four above this plane. Four of the speakers are inverted so that the speakers above mirror the speakers below. I am building a frame that permits easy mounting of the speakers. Each speaker has its own *shelf* that angles the speaker toward the center of the cube. The frame can be transported out-of-doors and away from reflecting surfaces (other than ground reflections). I work on a ranch (part-time) and is why I have ready access to an open space. Next I play the initial recording that consisted of noise bursts emanating from six virtual speakers, but the processed recording is played thru the cubic arrangement. At the center of the cube is the TetraMic. This time there this is no speaker (or speakers) on the horizontal plane passing thru the mic, but the initial recording was made from a virtual array of speakers lying on this plane. If the playback provides a true physical replication of the original recording, the resulting B-formatted files of the recording-of-a-recording should closely match the B-formatted files from the first recording in both level and spectral make-up. To a listener, the virtual surround (first recording) should appear as speakers in a circular array, each equally spaced 60 degrees apart and at ear level, when played through the cubic array. Of course, I'm assuming the listener is positioned such that his/her ears lie on the horizontal plane that passes thru the center of cubic array. But when we replace the listener with the mic, the physical wave fronts will provide objective evidence of *accuracy* in terms of spatial orientation at the listening position. If the radius of the virtual (circular) array is greater than the distance to the faces of the cube, we might also get a sense of sound-to-source distance that goes beyond the (imaginary) sides formed by the cubical array. But because distance-to-source judgments depend on familiarity of a sound or SNR, I'd rather rely on objective results obtained via this proposed iterative recording process. Maybe my idea is not original (though it is independently conceived), or even the bestest of ideas. But then, it isn't beneath me to put round pegs in square holes and do my own experimentation. Note that this experiment is void of music, doesn't require human subjects, but it is all about Ambisonics. Best to All, Eric 'Blockhead' C. **URL to photo is www.cochlearconcepts.com/for_sursound/tetra_mount.jpg -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130703/3cff4821/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] A higher standard of standardness
On 07/03/2013 06:31 AM, Robert Greene wrote: I apologize if people took offense. fwiw, i did not take offense at your clear preference for realistic recordings (which i share and aspire to as well). i do object to hand-wavey cultural pessimism that postulates the end of scientific thinking. stereophonic techniques have been scrutinized and researched in very great depth and detail, and test recordings of the sort you were alluding to are routinely done by sound engineering students and seasoned recordist alike. the papers and data are out there. stating otherwise doesn't change that fact. let's not make sursound into a boring solipsistic debate club that negates everything which hasn't been discussed here before. snip Except in audio, where no simple question ever seems to get definitively answered and every almost discussion turns into mush by means of enlarging the complexity of the situation to the point that there are so many variables that no analysis is possible without wild difficulties, if at all. Personally, I would just like to know which mike technique does what to the tonal character of sources at different locations around the recording stage. If you don't care, you don't care. But I wish I had a disc where I could listen and find out. I find it hard to believe that other people are not interested in this. that's because they demonstrably _are_ interested in this. it's just not as easy as you make it sound. let's begin with the simple definition of tonal character. you won't be able to separate tonal character from spatial rendition. coloration and comb filtering are a fact of life, and a perfectly uncolored monophonic source will often sound less pleasing than a comb-filtered stereo reproduction (unless your listening room helps a bit). moreover, the brain is able to extrapolate from severely comb-filtered sensory input and gives us the impression of hearing an uncolored auditory event. good luck simplifying that :) i'm looking forward to hearing about your test design. Science works like that:one step at a time. Assuming that people are interested in science. yeah, that's why we have complete understanding of the human brain. because it's sooo easy to understand, if only people would read more sursound and not add needless complications. come on! Years ago I decided to learn the piano(I am a violinist!) just to see how it would go, by learning the Rachmaninoff 3rd piano concerto --a measure at a time. As you can imagine I did not get very far! q.e.d. your approach to scientific evaluation of recording techniques seems similar. Audio seems to be missing a lot of the basics. yes, because psychoacoustics is _hard_. PS There is a good bit of this sort of thing about LOCALIZATION. But not so much about timbre. check out for example theile's spectral objection to summing localization, but do get a case of wine and cigars before you dig in, because it's going to be a loong and very interesting night if you follow through some more papers. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Ambisonics - Decoding 16 channels in DAW
On Wed, Jul 03, 2013 at 01:15:58PM -0400, Daniel Courville wrote: Two instances of Harpex-B, each one on a separate output bus, in shotgun mode, each decoding to eight shotguns, using the Cube preset as a starting point. M Readers of this list will know me as one of those who, whenever Ambisonic file formats etc. are discussed, will spoil the fun by stating that there's no life below third order or so. And of course I'm still of that opinion - if you want a system able to emulate whatever speaker layout and working over an extended listening area things start to work at third order. But that doesn't mean that first order doesn't work. It can work incredibly well in good conditions. Ten days ago I spent an extended weekend at the music conservatory of Pesaro (Italy), where David Monacchi (who is a teacher at the conservatory) has built an electronic music studio featuring a 3rd order periphonic Ambisonic system using 21 speakers. The room has had extensive acoustical treament, the only remaining problems are some low frequency room modes (bass traps are being installed to deal with those). I got involved for specifying the ambisonic speaker layout and decoder. David has also made field recordings (Ambisonic, stereo and bin- aural) in primary forests in various places around he globe. These are absolutely fascinating - I you were at the first AMB convention in Graz you will remember his presentation. I had already made a third order Ambdec preset for this room. But since we had little real third order material to test with, we spent a lot of time listening to David's field recordings and to some others he made recently using an ST450. David was using several instances of Harpex to render those. This worked, but neither of us were really satisfied with the results. So I created a first order Ambdec preset using a subset (12) of the available speakers. The results were astonishing. Suddenly there was depth, perspective, involvement, and an uncanny sense of realism. It took both of us half a minute or so to adapt to it - something that has been reported before. But after that short time it was really a completely different experience. In short: the Ambisonic magic only works when things are done right. Using virtual shotgun mics pointing at some arbitrary collection of speakers (or even a near optimal one as in this case) may produce some effect, but it doesn't even come close to what can be achieved. And in fact it has little or nothing to do with real Ambisonic reproduction. 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
