On 2010.07.21. 19:26, Paul Davis wrote: If I code program to handle attenuation of sounds depending on their source (emitter) position in virtual 3D space, I guess then there's no simple way to relate the effect to real world. How this is usually handled? I guess you misunderstood me a bit. I use floats internally regardless of inputs or outputs. I simply don't (won't) divide integers by 2^(16-1) (or whatever bit depth else) when converting. I expect to divide the internal float value when file output is of floating point type. I think this should reduce problems with two conversions to/from floats, when both inputs and outputs are of integer type. -- JohnLM

On 07/21/2010 08:56 PM, JohnLM wrote: you reduce the level with distance (amount depends on whether you are in the near or far field and whether you have a point, line or planar source), and attenuate the high frequencies to simulate air damping. if you're in an enclosed space, you'd also want to take into account the dry-to-reverb ratio, and ideally simulate correct early reflections.

Continuation of talk from Re: [LAD] Floating point processing and high dynamic range audio Simply Subject line no longer described the actual content. On 2010.07.21. 20:19, Jörn Nettingsmeier wrote: Thanks! Though this is quite sophisticated. I currently only worry about sound (pressure) level. I will try to implement the rest when the current code actually works. Well I have a set of (omidirectional) point sources. The code currently has no information whenever given space is open or enclosed. (I guess open or empty space is implied then) As much as I can understand the sound pressure is inversely linearly correlated to distance. p~(1/d) or p=k*(1/d) where k is currently undefined constant coefficient. What's the thing about far and near fields? On Wed, Jul 21, 2010 at 2:01 PM,<fons(a)kokkinizita.net> wrote: Ummm... is it sound pressure or sound pressure level? Or it doesn't matter? (are they equivalent?) On 2010.07.21. 20:40, Ralf Mardorf wrote: I'm sure any digital alogrithm are far from perfect to represent the acoustic attenuation (and related) effects like in real world. But it just need to be close enough. Ummm... I somehow fail to see how the microphone rig is related to this.

On 2010.07.21. 21:20, Ralf Mardorf wrote: True. But this is of no concern for my program. It doesn't try to emulate psycho-acoustics of human perception. IMO that would be a problem of acoustic engineers. My program just tries to translate sounds in virtal space to audio track of one, two, six or whatever else number of channels as best as it can doing it automatically. The project was started when the amount of sounds in a video ("foley" sounds, background effects and such) was becoming more than I wanted to deal with manually with a DAW. And I couldn't find any FOSS alternative. If it is of interest to anyone, program (non-working draft of program) is located here. https://launchpad.net/mage

@ nonsense and bullshit, where are the examples that it works? There is no valid recording with more than 1 or 2 channels, regarding to a natural impression. Some art projects that didn't try to give a natural impression are something very, very different. Most audio engineers still fail regarding to stereo and mono issues. I wonder about the geniuses who are able to do 5.1 and all the other stuff. Please post links to the geniuses work, but call me names.

On Wed, 21 Jul 2010 17:10:35 -0400 Paul Davis <paul(a)linuxaudiosystems.com> wrote: I remember when 2001 first came out, being totally astonished by the sequence where the ape-man first uses a bone as a club. There was the totally realistic effect of a gust of wind moving from the screen to the rear of the cinema. I've no idea what system was in use, but it certainly wasn't 2 channel stereo! -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

On Wed, Jul 21, 2010 at 10:23:03PM +0100, Folderol wrote: 1968 IIRC.... Well before any Dolby-XYZ The 70mm version would have had 6 audio tracks, the 35mm one 4. Ciao, -- FA There are three of them, and Alleline.

On Wed, 2010-07-21 at 17:10 -0400, Paul Davis wrote: Indeed I never heard a recording done with this equipment. Where can I get an example done with this equipment? I can't listen to it at home, but I guess I'm able to find somebody with the valid equipment to listen to the recording. At least somebody from the list, e.g. you Paul does know some studio were I could listen to a recording here in NRW, Germany. Seems to be very common equipment, so that it makes sense to work with it.

On Wed, Jul 21, 2010 at 5:36 PM, <fons(a)kokkinizita.net> wrote: and had you been in Berlin in either july or fall of 2008, you could have listened to a multi-mic ambisonic recording of a performance of Messaien's Meditations on the Holy Trinity, played in the Koln Dom (the first performance was streamed live; the second was a playback of the recording). if that won't work for you, then perhaps you might want to visit any of the theaters outfitted by these guys: http://www.iosono-sound.com/references/installations/ and if that still isn't enough, Nimbus Records have released several B-format recordings. --p

On Thu, Jul 22, 2010 at 7:44 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: this is about *audio*. what is looks like tells you almost nothing. in addition, you don't ned anything fancy to listen to B-format recordings, and one of the major reasons for that is fons' open source decoder that will allow you to listen to them with any jack-enabled audio player and your existing equipment. you also don't need any particular speaker setup, although clearly one of the points of ambisonics is that (up to a limit) adding more speakers enables a more accurate recreation of sound placement within 3D to be achieved.

On Thu, Jul 22, 2010 at 10:29 AM, Chris Cannam <cannam(a)all-day-breakfast.com> wrote: my limited understanding is this: the B-format data encodes the source position relative to some defined point in space. the decoder can map the "origin" used to define the positional space however it wants to. whether or not any decoders actually offer any control over this is another matter.

On Thursday, July 22, 2010 01:07:57 pm fons(a)kokkinizita.net did opine: And interesting scenario, Fons. But it leads this simple minded broadcast engineer with 45 years experience to ask a question. 1. How are the signals brought into phase such that electronically, all mic ribbons or diaphragms seem to occupy the same space, just facing in different directions? If this is not addressed, then this will lead to some interesting comb filter effects if the signals are not kept from mixing, which they will of course do in the ear. Granted, the PV of sound in normal air would require separations of inches till the stuff above high C comes into play, but at the snares and cymbals frequencies, I would have to assume some coloration of the sound from this effect alone. And of course the same concern comes into play at the speakers since they are generally placed around the listener which in no way approximates the nearly single point reception these mics will hear. In my own mind, the placement of a PZ microphone in each of the places one would place the playback speakers would seem to be a superior method, at least for a listener sitting in the nominal center, who will be so overwhelmed by (supposedly not important sonically we are told) the phasing errors that he cannot single out a single largest cause for the lack of realism. In my history of electronic repairs for a living over the last 60+ years, one instance of truly hair raising realism took place when I was about 21, and working one of the service benches at Woodburn Sound in Iowa City IA, USA. I had bought some car parts at noon, and when I left about 6 for dinner, I forgot & left them on the corner of the bench. Having a key to the back door I let myself into the back door about 8, which was pretty dark by then as only one 25 watt bulb out in the display area was on, and half way to the door to my bench area & right in the door to the front, display room, the Dukes of Dixieland marched by, going right over me. It seems that Woody and Saul Marantz were out in front, had pulled a 2nd JBL Hartzfield speaker out of Saul's econoline van, setting it just inside the front door opposite to ours in the other front corner of the display floor, along with a Berlant/Concertone tape deck capable of running at 15 and 30 IPS. And the tape was the master that had cut the Dukes then current hit record, running at 30 ips. SNR was a good 70+ db, and there was no tape hiss audible unless you walked directly in front of the JBL 075 ring radiator tweeters that had been added to both our Hartzfield and to the one Saul was carrying around. No tone controls, and only a 30 watt Marantz stereo amp., those Hartzfields were then, and may be yet, the most efficient speakers ever made, never used more than 3 or 4 watts/channel to get SPL's that would have done Joshua's trumpets at Jericho proud. Truly a total immersion in the sound, from about 35hz to nearly 30khz. Those tweeters could do a fairly good job of reproducing a 25khz square wave. It took till I had been introduced to Saul Marantz and shook hands, and for that tape (on 14" NAB reels) to run out before the hair on the back of my neck was truly relaxed. Saul it turned out was an endless source of technical knowledge sprinkled with BTDT stories. And needless to say, I did not manage to get that hydromatic transmission I had just stick shifted back together till a day later. Yeah, I'm a JOAT. :) -- Cheers, Gene "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) The cost of living is going up, and the chance of living is going down.

On 07/22/2010 08:42 PM, Ralf Mardorf wrote: that is a bogus statement. phasing happens in the brain as well. just put on your favourite pair of headphones and wire one side out-of-phase. instant nausea. (of course, if you keep it on for a few days, your brain will adapt - presto: you'll be hurling all over the real world when you finally put them down.) a terribly chosen analogy indeed. since when do the eyes care what phase an incoming photon is? unless you're staring into a laser, each photon will have totally random phase. next error: there *are* two images, and they do need to be synced. phase is irrelevant, though. you can get terrible phasing, and not just in the center, but pretty much everywhere. that's why some people stagger the timings of the loudspeakers a bit, to smear out the phasing until it is more or less masked by the content. but it should be noted that stereo has the very same problem. now if method A produces a 60° soundstage with phasing at N units of obnoxiousness, a method that produces 360° surround is entitled to 6N UoO phasing. in practice, ambisonics does better than this, but there is no denying the issue. one thing that often gets overlooked: people have learned to accept stereo (or, in some circles, 5.1) as the gold standard, and its shortcomings have grown into desired features. it's very hard to compete with a method that does a few things very well and doesn't even try to reproduce most of the auditory cues of, say, a live experience. the main ingredient that makes any sound reproduction system sound good is your brain. the trick is to nudge it into sympathy with carefully chosen cues.

On Fri, 2010-07-23 at 08:51 +0200, Ralf Mardorf wrote: PS: Not only because of the position of the speakers, but started with the recoding, resp. master mix, that's why ... ...:

On Fri, 2010-07-23 at 10:02 -0700, Fernando Lopez-Lezcano wrote: That gives me hope that I'm wrong about abilities of surround sound :). If I'm mistaken, it will be a win for me :). - Ralf

On 07/22/2010 08:15 PM, Gene Heskett wrote: well, you can't do that :) two approaches: if you only care for horizontal surround (and there is no significant z axis content), you can stack an omni and two fig8s on top of one another, so that they are in phase for signals arriving along the horizontal plane. this is often called the nimbus-halliday array, after the guy that introduced it at nimbus records. sounds very good, but no height. if you do want height, place 4 sub-cards or cardiods as close together as possible, arranged on the faces of a tetrahedron. then with some clever filtering, you can get very good results, and the comb filtering at hf can be worked around with a slight treble boost. comb filtering at hf looks worse than it sounds, usually. that's not the point. the idea is this: you measure the soundfield in one spot, and you aim to reproduce it in this one spot only (that's the maths). so the speakers should be really really close together. the good thing is: it also works when the speakers are far apart, which means we get room for listeners :) the interesting aspect of first-order ambisonics is not how it works (it only works in a point volume, which is not terribly useful), but how gracefully it fails outside this volume. it might seem so, but in practice it does not really work. say you have a regular hexagon of microphones, with a radius of 2m. assume that a singer is standing at the "north" mike. sound reaches this mike instantaneously. it takes about 12ms to reach the south mike. that is a comb filter that reaches way down where it makes the sound tinny, unpleasant, and worst of all, unrepairable. and consider what the effect during playback would be: loud, correct sound from north speaker after 6ms, bogus echo from south speaker after 18ms, combining into coloration, giving a wrong spatial cue of a back wall that isn't there, yet arriving too soon for the brain to be able to separate it as a bogus event and throw it away. the fun thing is, if you had several thousand microphones (and speakers), the results would be excellent. but before shouldering the expenses, you might be tempted to cut some corners. ambisonics is a very usable shortcut imho.

On Thursday 22 July 2010 16:29:01 Chris Cannam wrote: wrote: It makes sense to reduce B-format to stereo. But the target is important, if you aim at headphones, there are decoders that create an binaural signal. If normal stereo-systems are the target, you will do a decoding similar to any ambisonics setup but only use two speakers in the correct stereo positions and decode to file... The headphone-version gives more of the ambisonics feeling, but the normal stereo signal also benefits from the recording done in ambisonics. Have fun, Arnold

On Wed, Jul 28, 2010 at 10:47:03PM +0200, JohnLM wrote: If you use 2nd order AMB and a good AMB->5.1 decoder the results should be very good. The combimation AMB panner + AMB->5.1 decoder is in fact a 5.1 panner, but quite a sophisticated one. You could put one of theae in each channel, but since the second part is the same for all it's more effficient to do it after the mixing, in other words use an AMB mixing bus. Compared to conventional 5.1 pairwise panning the result will be more even, without emphasising the speaker locations as would be the case otherwise. In other words, the sound will be much less seem to originate in the speakers. Ciao, -- FA There are three of them, and Alleline.

On 2010.07.28. 22:06, fons(a)kokkinizita.net wrote: I stumbled on your own AMB-Plugins. :) They seem to do pretty much what I need. Mono->B-Format panner provides only angle controls. So from what I can understand, if I apply doppler effect, distance attenuation and other pre-process filters to a mono input and then push it through the panner plugin I get fairly correct representation of sound in 3D space. Right? I failed to find anything discussing mixing in AMB. Can I just sum the channels of all sounds, like I would do it to any "direct-speaker-to-channel" formats?

On Thursday, July 29, 2010 08:52:04 am Jörn Nettingsmeier did opine: This use of 'doppler' I would call incorrect, because the doppler shift is a shift in the apparent frequency of the sound rising at it approaches, and decreasing as it leaves. I'm sure you have a term for what you mean, but doppler isn't it. That police officers radar gun measures your speed by listening to the echo from your vehicle, and comparing it to the signal its sending, which to simplify, results in a beat frequency which is exactly your speed if approaching or departing exactly to or from the radar guns position. That is why it is often called doppler radar & the weather guys us it also. Because they don't stand directly in front of you to take a reading, there is some small vector error in your favor. Someone else was trying to describe the distance vs square law change in the apparent volumes. So let me try from the broadcast engineering field to explain that better. Imagine a point src of energy, be it light, sound, or other radiation such as a radio or tv signal. Measured at distance x, you will get your reference signal, call it 0 db in this case. Now, without changing anything else move your measuring instrument to a point that is now at a distance of 2x. You don't get half the signal, but 1/4 of it, because the same energy that was hitting a square of any arbitrary measurement, say a square inch, has in addition to being spread twice as wide at distance 2x, it is also twice as high. So the new reading will be -6 db compared to the original '0' db. That is why we call it the square law. The only way to get that back is to make the receptor itself 4 times bigger. But while I have observed that there are quite wide variations in ears, I have not seen an individual with expandable ears (yet) :) -- Cheers, Gene "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) I call them as I see them. If I can't see them, I make them up. -- Biff Barf

On Thu, 2010-07-29 at 17:01 +0200, JohnLM wrote: You notice a Doppler effect? So your speakers are far apart from each other, while you are moving very fast from one speaker to the other? Nothing else is called the Doppler effect. There might be a sound similar or equal to a Doppler effect, but of course using this term is bad.

On Thu, 2010-07-29 at 10:00 -0700, Fernando Lopez-Lezcano wrote: At the moment I'll don't read the links, I'll do it another time (ASAP). I guess it was Jörn, who FIRST said "doppler would come into play only if you change the distance of the sound, not when you move it on the sphere, as the distance remains constant." And I do agree. So, for stereo only (or even 'complete' 2D), we do have at least effect of; the 'loudness is the same for all stereo positions' and 'loudness will increase when going near to the centre ', regarding to the mixing console, when using the pan pot. And we also could have some kind of 'out of phase' effect. But a Doppler effect would cause an audible detune effect. As far as I guess to understand it, Doppler is completely out of importance.

On Thu, 2010-07-29 at 19:32 +0200, Ralf Mardorf wrote: PS to the German speaking people (perhaps understandable on English too): Imagine that you don't record the "TatüTata" of the police car, but you have got a sampled sound of this noise and you wish to move it by using a pan pot. We would miss the Doppler effect :D.

On Thu, 2010-07-29 at 17:01 +0200, JohnLM wrote: When I'm in the cinema I sometimes notice unbearable phasing, while a sound is panned from one 2D position to the other, this isn't a Doppler effect, but an issue regarding to transit time or something else. IMO out of phase vs detune.

On Thu, 2010-07-29 at 11:05 -0400, Gene Heskett wrote: One of the best mixed recordings I know is THE CRY OF LOVE (J. Hendrix), but unfortunately they did some very annoying pannings (for stereo only).

On Wed, 2010-07-21 at 22:58 +0200, Ralf Mardorf wrote: It's unimportant what are my thoughts about those links, I'm unable to listen, because I'm reduced to stereo, for good reasons. But I would like read what people who do have 3D equipment do feel, while listening to those perfect 3D recordings. I do agree, that 3D could be good, but it never ever will sound natural. There is no perfect building to the context for stereo or mono and there also is no perfect way to use 5 or 7 audio channels. A lot of people, even a lot of those who you respect, tried it and I never meed someone who preferred mono or stereo to surround. Btw. sometimes I do fail too, so I don't insist that I'm not mistaken, to the contrary, it would be a win for me too, if I should fail with my opinion.

On Thu, 2010-07-22 at 01:04 +0200, Arnold Krille wrote: Ok, I noticed this. Fon's AmbDec is the player for the files from http://www.ambisonia.com/?! I guess a friend who lives near to my home has at least 8 outputs for an Envy24 sound card and at least a JBL 5.1 setup, so the 5.1 speakers could be used for a minimalist ambisonics. Bad sentence construction don by me. Cheers! Ralf

On Thu, 2010-07-22 at 14:01 +0200, Ralf Mardorf wrote: Yes. Fons ships an ambdec configuration for playing in a 5.1 setup (it is not optimal because the back speakers in a 5.1 setup are too far apart but it works - I use it at home). Of course it assumes that the 5.1 setup is correct. This link has a diagram of the speaker angles: http://de.wikipedia.org/wiki/5.1 -- Fernando

On Thu, 2010-07-22 at 20:22 +0200, Ralf Mardorf wrote: I'm not sure I understand. The 5.1 ambisonics decoder in Ambdec can decode to that arrangement and it is pretty good. You could try that before moving the speakers around. Yes, you could move them into a square, use only four and use the square decoder. A 5.1 setup can be used to play up to second order ambisonics, a 4 channel square can only play first order ambisonics. Most of the recordings in ambisonia.com are first order (but I think there are a few higher order as well). See this for another example of ambisonics at home: http://stackingdwarves.net/public_stuff/linux_audio/ambi_at_home/joern_nett… -- Fernando

On 21 July 2010 22:18, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: Basically, just as long as you post twenty thousand emails to every goddamn linux audio related mailing list per hour you're happy.

On Wed, 2010-07-21 at 23:21 +0200, fons(a)kokkinizita.net wrote: Yes, I could be mistaken, but I guess it#s nearly impossible. Correct reduce it to the last 10 years, but 30 years. I guess you make a fool of your self, perhaps not to Linux lists, but indeed to professionals who have to do hard jobs all the day. At least we still have all that phased stereo recordings, that we aren't allowed to send on radio in Germany.

On Wednesday 21 July 2010 23:27:59 Ralf Mardorf wrote: Sorry for getting personal (you started it Ralf!), but: From what I have heard and seen from Fons' past and his present work both in terms of software and audio compared to what I could read from you, he is _far_ less likely to make a fool of himself anywhere in the audio/mixing/dsp/programming field. Neither in research nor in "hard daily work". Arnold

On 07/22/2010 12:58 PM, Fernando Lopez-Lezcano wrote: Oh yeah, Fons is wonderful and never makes a fool of himself. His shit smells sweet too! ;-P We need to erect a monument to the almighty, allwise and all powerful Fons. Nay, forsooth a religion shall be founded to worship at the feet of the 50 Foot Golden Standing Fons Statue. A giant Microphone and sound system shall also be erected in his honour and we shall sing in his name on imporatnt dates like when sent his first email and farted in melody ;-P Legends will be told and passed down through the generations and all practising members shall be required to install 30 point ambisonics in their living rooms as testiment to their faith. -- Patrick Shirkey Boost Hardware Ltd

On Wed, Jul 21, 2010 at 11:38 PM, Patrick Shirkey <pshirkey(a)boosthardware.com> wrote: this was completely pointless. and offensive. and rude.

On Thu, Jul 22, 2010 at 01:38:22PM +1000, Patrick Shirkey wrote: Too much. just my birthday will do, it's the first of april. You may think I've been to harsh to R.M., but one has to draw a line somewhere. A few weeks ago two religious fundementalists (don't know which denomination) rang at my gate to discuss the fallacy of Evolution. One of them asked "Did your grandpa look like this ?", showing a picture of an ape. I responded "No, not at all. But let me ask you, did _your_ grandfathers _think_ like apes ?" - "No" - "So why do _you_ ?". And that ended our little discussion. Stating that we have only two ears and that consequently any audio system reproducing 3D would require direct injection into our brains is about at the same level of ignorance. Ciao, -- FA There are three of them, and Alleline.

On 2010.07.22. 1:30, fons(a)kokkinizita.net wrote: I may have bitten more than I can chew on. :) Currently I seek to make "stupid" solution. Something like the way every decent game engine does it. Except my solution is going to be non-realtime and not (sound) hardware dependent. I reckon something like this already exist in FOSS, but I haven't been able to find anything (I have no time examining code of every other audio project out there). Thanks for the info. Good to know I'm heading the right way. Indeed 'near field' sound sources were ones I never actually gave much thought on their implementation in software. This will go next on my TODO list. Regards, John

On Wed, Jul 21, 2010 at 01:26:40PM -0400, Paul Davis wrote: ??? At least the first of these makes no sense at all, and the second applies if the dB value indicates a digital peak level. More accurately: 0dB digital peak corresponds to a sample value of + or - 1. For RMS or average, 0dB usually means the level of a sine wave with amplitude 1, i.e. having a peak value of +/- 1. It can be confusing as the RMS or average levels of such are signal are not unity. An audio signal represents pressure variation as a function of time. Multiplying it by two will give 2 times the pressure, and 4 times the power. The subjective result is another matter. Ciao, -- FA There are three of them, and Alleline.

On 2010.07.21. 20:05, Paul Davis wrote: Thanks! This starts to make some sense and helped a lot.

Excerpts from Paul Davis's message of 2010-07-21 19:26:40 +0200: I think the word loudness is a problem here. Afaik it usually refers to how it is perceived, and twice the amplitude doesn't mean twice the perceived loudness. It may mean twice the sound pressure level, energy, or intensity (if we ignore analogue anomalies, as you wrote in some other answer). It may be nitpicking, but I found it important to differentiate between the physical phenomenon and the perception. I like Musimathics by Gareth Loy for the good explanations and clear use of terms. It actually succeeds in explaining everything in clear and easily understandable terms, I can recommend it for the writing style alone. -- Regards, Philipp -- "Wir stehen selbst enttäuscht und sehn betroffen / Den Vorhang zu und alle Fragen offen." Bertolt Brecht, Der gute Mensch von Sezuan

Excerpts from fons's message of 2010-07-22 02:24:04 +0200: I had a brief look at the section about loudness in musimathics and it mentions 10 dB based on the work of Stevens, S.S. 1956, "Calculation of the Loudness of Complex Noise" and 6 dB based on Warren, R. M. 1970, "Elimination of Biases in Loudness Judgments for Tones.". I think I've encountered the 6 dB more often in texts, which doesn't mean it's closer to the truth, if that's possible at all. Knowing a 'correct' number would be nice for artists and sound engineers, but if it varies wildly from person to person, as Gareth Loy suggests (no idea where he bases this on) then this simply isn't possible. Picking any number within or around this range is probably as good as any other. I never thought about that to be honest. It's immensely complex. It might have to do with each persons hearing capabilities, for example the bandwidth of loudness perception or the smallest discernible loudness difference. If it really is very different from person to person, then an explanation that takes the different hearing capabilities into account could be sensible, don't you think? Good night :) -- Regards, Philipp -- "Wir stehen selbst enttäuscht und sehn betroffen / Den Vorhang zu und alle Fragen offen." Bertolt Brecht, Der gute Mensch von Sezuan