Hi Paul, thanks for replying! I'm shaken with the depth of your knowledge, as always! :) How very detailed... Thankya! Yeah... yeah, yeah, yeah I'm hanging that one up over my bed. Carlo

Hi Loki, thanks for replying! Couldn't agree more that it's about what you do with the hardware you have way more than which hardware you have. Carlo

Joe Hartley wrote: About the bit depth, I always use 24bit. In this way if the sound level of a recorded track is not high enough I have more dynamic range to use when I mixdown in 16bit. c. -- www.cesaremarilungo.com

On Thu, 26 Jan 2006, Carlo Capocasa wrote: It's not resolution but frequence range. A 192k card can recod/play audio frequencies up to ~96 kHz (in theory) which is about 5 times higher than any human being can hear. Best regards, Hannu ----- Hannu Savolainen (hannu(a)opensound.com) http://www.opensound.com (Open Sound System (OSS)) http://www.compusonic.fi (Finnish OSS pages) OH2GLH QTH: Karkkila, Finland LOC: KP20CM

On Fri, Jan 27, 2006 at 10:53:45AM -0800, Florin Andrei wrote: Steve's Decimator plugin simulates this. -- Paul Winkler http://www.slinkp.com

Yeah, the good old economy, isn't it... Thanks for pointing out! Carlo

On 1/26/06, Joe Hartley <jh(a)brainiac.com> wrote: I saw a 400GB 1394/USB2 drive at Fry's yesterday for $219... ;-) - Mark

On Thursday 26 January 2006 14:23, Carlo Capocasa wrote: It doesn't take well trained ears, Carlo. I'm 71, with "Carhart Notches" over 100 db deep in my hearing, mostly from wearing out 3 rifle barrels at the target range before I discovered ear-muffs. But, I can feel uncomfortable, even pain yet, instantly, from the effects of the old acoustic doppler motion detector type burglar alarms that run at about 44 kilohertz when I enter a building equipt with such alarms. I don't make it a habit to stick around and discuss the weather with the occupants when I encounter them things. -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2005 by Maurice Eugene Heskett, all rights reserved.

Michael T D Nelson escribe: [1]This is the best article I've ever read about this issue 1. http://www.prorec.com/prorec/articles.nsf/articles/8A133F52D0FD71AB86256C2E… Cordially, Ismael -- Tout fourmille de commentaries, d'auteurs il en est grande chert� http://lamediahostia.blogspot.com/ http://www.flickr.com/photos/ivalladt/

Michael T D Nelson escribe: In the case of classical music almost as important as melody, harmony or rhythm themselves. Even there's a lot of minimal electroacustic music out there where dynamics take the main part in expressivity. Cordially, Ismael -- Tout fourmille de commentaries, d'auteurs il en est grande chert� http://lamediahostia.blogspot.com/ http://www.flickr.com/photos/ivalladt/

On Fri, 2006-01-27 at 14:01 +0000, Michael T D Nelson wrote: Just curious - what do you think about the bass response on the M1 Active? -- Florin Andrei http://florin.myip.org/

On Friday 27 January 2006 10:07, Michael T D Nelson was like: That really is a moot point. I'm of the opinion that once behind the microphone, it's the musician's job to perform their part with all their heart and soul. Having to pull your punches because the technology can't handle your dynamic range does not necessarily produce good results, and anyway, that's the engineer's job, surely? Too much control on the part of the performer tends to lead to sterile recordings IMO. I like rattle and hum, I enjoy the effect of pushing certain bits of technology beyond their factory specs (mostly in the analogue realm, it has to be said). I like to hear the squeaky chain on a bass drum pedal. For me, too many studios are, like hospitals, airless pits where music goes to die. The engineer serves the music _not_ the other way round. </rant> :) -- cheers, tim hall http://glastonburymusic.org.uk/tim

On Saturday 28 January 2006 13:08, Michael T D Nelson was like: Sure. Yes, and the engineer can always place them a foot away from the microphone with a bullet-proof pop-shield. ;) Having read the further responses. I don't think I'm really in any disagreement with the general point. The less you mess, the better. Look me up if you're passing through. -- cheers, tim hall http://glastonburymusic.org.uk/tim

On Saturday 28 January 2006 13:44, Ismael Valladolid Torres wrote: Humm, I would have thought that this would have been the subject of the first weeks lesson in Music Production 101. With refresher course sections on mondays thereafter. :-) -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2006 by Maurice Eugene Heskett, all rights reserved.

On Fri, 2006-01-27 at 09:59 +0100, Ismael Valladolid Torres wrote: Yeah, that's a classic. ;-) -- Florin Andrei http://florin.myip.org/

Wolfgang Woehl &lt;tito(a)rumford.de&gt; writes: If so, you have already recorded it, haven't you?

On Thursday 26 January 2006 15:47, Wolfgang Woehl wrote: In a recording situation, there is always the possibility of aliasing, and once thats in the digital data, it cannot be seperated by any other means than a resample with a canceling phase signal. As the original phase is unknown, there isn't much chance of ever getting it back out of the data. Aliasing is what happens when you record something with a doppler motion detector running, and the recorder doesn't have a brick wall filter thats at least 80 db down at half the sampling frequency and above. The resultant playback will have a nominally .1 to 5 kilohertz tone superimposed, and thats the diff between the 44.1khz sampling frequency and the doppler burglar alarms normal running frequency. Admittedly thats an extreme demo, but its a pretty good one if you'd like to try it. It doesn't need more than a 5 dollar microphone either. Personally, I find aliasing distortion so unpleasant that I'll waste the bandwidth and data storage to sample well above such man made racket. But thats just me. YMMV. -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2005 by Maurice Eugene Heskett, all rights reserved.

On Thu, 26 Jan 2006 16:11:48 -0500 Gene Heskett &lt;gene.heskett(a)verizon.net&gt; wrote: A fact often forgotten is that the human ear is itself non-linear in response. It can be reasonably argued that (assuming no other distorting factors exist) well out of band signals could interfere 'mechanically' inside the ear and produce audible difference signals. What is more, these signals will vary from person to person. An experiment I have tried that seems to show some non-linear behaviour, and gives different results for different people is to produce an apparent pitch change to a constant frequency tone. Play a 440 Hz sine wave (A) to someone at very low amplitude. Let them get comfortable then suddenly increase the level about 40dB. Some will say the pitch dropped, a few that it rose and some that there was no change. It seems to work best if you can get the amplitude change to take place at the zero crossing point so there is no audible click. -- F

On Tuesday 07 February 2006 04:26, David Cournapeau wrote: [...] I've taken the liberty of a spelling correct above s/on/one to make it correct. My thoughts on this run toward a 1 bit sample at 256khz (or more since we could do it at many megahertz today) driving both the output data stream, and a 24 to 32 bit counter whose input is the same clock, and the direction bit is this 1 bit a/d's output. The catch here is that the 1 bitter is comparing the output from a d/a driven by this counter with the next sample to make its data bit that then slews the counter one way or the other by this one bit. Such a device would be slew rate limited at the higher frequencies, but should not generate any aliasing products whose magnitudes would be greater than the bit ratio allows, and its input anti-aliasing filter could be very simple. Its output would already be pretty well compressed, particularly as the slew rate approaches the limit, and further digital processing of the 32 bit range down to 24 bit might offer a considerable amount of additional compression (you cand read that as noise reduction) to be done. It would of course assume a counter driving the d/a that did not overflow in either direction, and some sort of circuitry to attempt to ac couple the signal by doing a very slow slew toward the 50% point exerted over a t=rc of about .2 seconds, which should be ignored for the most part by the reproduction system. Such a devices nyquist limitations and the subsequent aliasing would be a long way below the human ears range of being able to detect them under any conditions I can imagine, even by those who can hear an acoustic doppler burglar alarm. That said, its also an old idea/technique, waiting only for sufficiently well behaved electronics which we've had for what, 2 decades or more now? Or has this quietly occured in the utilitarian devices we buy today without being an important enough detail to make it to the propaganda on the box? Are these then considered to be implementation details that are considered proprietary/trade secrets & best not mentioned, yadda yadda? I don't know, so you all tell me please. -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2006 by Maurice Eugene Heskett, all rights reserved.

Some notes on 96khz: I tried working exclusively with 96Khz for about a month. The dynamic range is so extreme that the faintest of sounds are picked up. "Silence" no longer exists. ANYWHERE. Ladspa plugins like gates don't go below -70db, and they need to. A plus was I felt free to sell all my outboard compressors. My analog synths sounded fatter, voice more robust, but at least some of that was the result of turning my knobs down from "11" to give me 10DB or more of headroom. (I'd fallen into a bad habit) The 96Khz Bardsdown bosendorfer does sound awesome at 96khz (even though linuxsampler only does 16 bits, not 24). No background noises. I do use 96khz when I'm just playing piano, no vocals. I went nuts during this period quieting my studio, cutting various noise sources by about 8 db overall. (now saving up for a sound booth) I was still unsatisfied with listening directly to the 96KB recorded results - I could hear leaves in trees blowing outside (I don't exaggerate)... but the 44khz mixdowns were nice, all that extra stuff dropped out. I went back to straight 44khz/24 bit afterwards and although the sound seems "deader" I don't hear all that extra stuff all the time, and the savings in disk space and mixdown time are worth it. Note: I have a good ear (can hear up to 22khz in one ear - compensates for the other which doesn't go above 4k) So having hires audio is good for debugging your studio, and pure 96khz sound sources, but not so hot for development or analog work. The primary reason I remain interested in using it is to encode down higher quality surround. 192Khz strikes me as complete overkill, except perhaps in that instance. On 1/26/06, Florin Andrei &lt;florin(a)andrei.myip.org&gt; wrote: -- Mike Taht PostCards From the Bleeding Edge http://the-edge.blogspot.com

Yeah! So you're talking about a possible switch from consumer equipment to higher bitdepths/frequencies? Yeah... Law of diminishing returns at work? As for the mastering I'm wondering if it's worth it to get really good at it myself or if that's a lot like learning another musical instrument and best left to the pros. (Not meaning people who band together in 'professional organizations' but people who do it a lot :) I would assume using 32bit is similar in how much 'bang for the buck' you get than using 192 kHz? Thanks a lot for sharing, Florin, appreciate your insights and differentiated response! Carlo

On Thu, 2006-01-26 at 21:52 -0800, Florin Andrei wrote: Why would a consumer format EVER use 192KHz? Will people's hearing improve according to Moore's law? Lee

On Fri, 27 Jan 2006, Florin Andrei wrote: I think one of the earlier posters in this thread got it exactly right when he said that it's not so much that we want to reproduce frequencies that nobody can hear, it's that we'd like to have enough resolution to reproduce the waveforms of the ones we can hear well enough to be able to tell what shape they were supposed to be. As you go into the higher octaves, having only a few samples-per-wave resolution gives horribly approximated versions of each wave. Sure, a nice filter can smooth off the peaks and nastier bits, but you're still just trying to repair a wave that's had some of the information taken out of it -- and this for frequencies that are within our hearing range. It never makes sense to sample the analog world as "just enough" resolution. Good flatbed scanners scan at 45-bit color so they can give you 24. Good audio gear samples at 24-bit resolution so you can master at 16. It's never a bad idea to leave some room on sampling frequency. Things like that contribute to that "dithering noise" metallic sound that analog recordists never had to worry about. They were cursed with more background hiss and a lower dynamic range, but since the human ear already has to deal with noise and signal strength issues in daily use of our sense of hearing, I think things like that on analog recordings have always been easier for a listener to psychologically tune out than that metallic "sheen" in the upper octaves that bad digital recordings (especially over-processed ones) seem to always have. Nature never has that particular kind of distortion, so it's harder for the listener to overlook the way we've always forgiven phonograph and tape recordings in the past. -- + Brent A. Busby, UNIX Systems Admin + "It's like being + + James Franck / Enrico Fermi Institute + blindsided by a + + The University of Chicago + flying dwarf..." +

On Sat, Jan 28, 2006 at 01:30:54AM +0100, Esben Stien wrote: Exactly. And it's quite easy to see why this is true. Imagine the sample frequency is Fs and you have perfectly sampled a sinewave with frequency f (or calculated the samples). The inverse of sampling consist of creating an analog signal wherein each sample becomes a very thin 'spike' with the sample's amplitude, and that is zero in between those pulses. (*) If you calculate the spectrum of that signal, you find it contains all frequencies of the form k * Fs +/- f, with k and integer. To reconstruct the sine, you need to filter all of them out except the original f (and -f). Now this becomes more difficult as f approaches Fs / 2, as in that case Fs - f will be quite close to f, ** but it is nevertheless just a matter of filtering and of nothing else **. Again very true. The main reason why some people can hear a very very subtle difference between 48 and 96 kHz seems to be that it's quite difficult to make a 'perfect' filter for 48 kHz, even digitally. There are very few DACs that get this right (e.g. Apogee, and you pay for it). (*) Of course most DACs will 'hold' the value until the next sample time, giving a 'staircase' waveform rather than pulses, but that doesn't change anything fundamentally. -- FA

On Sat, 2006-01-28 at 14:11 +0100, Wolfgang Woehl wrote: True - "beat frequency" between two very close tones may be much lower frequency than the tones themselves. However, these lower frequencies would be recorded if you are using live sources. Your room mic(s) should pick these up and they would be recorded at 48KHz. -- Jan 'Evil Twin' Depner The Fuzzy Dice http://myweb.cableone.net/eviltwin69/fuzzy.html "As we enjoy great advantages from the invention of others, we should be glad of an opportunity to serve others by any invention of ours, and this we should do freely and generously." Benjamin Franklin, on declining patents offered by the governor of Pennsylvania for his "Pennsylvania Fireplace", c. 1744

On Sat, Jan 28, 2006 at 02:11:50PM +0100, Wolfgang Woehl wrote: There are certainly instruments that produce sound well above 20 kHz. Now to get 'beats' between two of those requires something non-linear - just summing them is not enough. Our ears are non-linear at high sound levels, so in a real performance this could happen. But I don't think it will be noticeable at any sane sound levels. As to non-linearities in the mixing process, that is something that should *not* happen. A properly dithered AD or DA conversion has *no* non-lineraties resulting from the quantisation (until you drive it into saturation). Even at 16 bits it's difficult to get audible intermodulation from the quantisation, unless you are working with pure sinewaves. But some cheap equipment shows very clear non-linear effects, often from the badly designed analog parts, or in the early days of digital, from the converters. It's a long time ago, but I remember having once tested a AD converter that had large 'gaps' - some ranges of values were never used. -- FA

On Sun, Jan 29, 2006 at 01:40:46PM +0100, Wolfgang Woehl wrote: Did you send the ardour output directly to jaaa or did the signal go via some analog path ? I just tried the same with two jaaa instances, one generating 400 Hz and the other 500, summed via jack and measured in either jaaa. No traces of 100 Hz. (Side remark: if your signals go through any analog hardware, it's better to avoid 100 Hz as a beat frequency. The mains frequency is 50 Hz in Europe, and I still have to see the first audio card that doesn't show up traces of 50, 100, 150, ... Hz in jaaa.) There are several different mechanisms at work here that can make you hear 100 Hz with your test. First, in order to hear anything at all, your signal has to be reproduced by a speaker, and these *do* produce intermodulation distortion. This can be quite noticeable in such a test, and in fact that's how it is measured. The sound card and amplifier will distort as well, but normally much less than any speaker. Second, you have chosen two frequencies that have a simple ratio (4/5). Human hearing will try to 'decode' them as a single signal, in this case as the harmonics of an 100 Hz one. So you could *hear* 100 Hz even if it isn't present physically. The effect has been kwown for ages by organ builders, who use it to make 'fake' 32" stops. -- FA

Tracey Hytry &lt;shakti(a)bayarea.net&gt;et>: Thanks Tracey. Looking for this I came across http://hyperphysics.phy-astr.gsu.edu/hbase/hframe.html again and want to recommend it for it's educational value and overall freaking greatness. I need to go to school again. -- Wolfgang

On Sun, 29 Jan 2006 16:21:12 -0500 Gene Heskett &lt;gene.heskett(a)verizon.net&gt; wrote: Just had a look at it... and immediately stuck it on my bookmarks! -- F

Hallo! that should be correct ;) ! LG Georg

no lg Georg

On Sun, 29 Jan 2006 23:26:23 +0100 Cesare Marilungo &lt;cesare(a)poeticstudios.com&gt; wrote: No! It is definitely 1/c1 + 1/c2 ... The only alternative I know of is 'product divided by sum' which is often easier to use. i.e. (c1 x c2 x c3) / (c1 + c2 + c3) -- F

fons adriaensen schrieb: with 3 caps, it would be: ceq = (c1*c2*c3) / (c2*c3 + c1*c3 + c1*c2)

Folderol wrote: Yes it is. I thought I've read Ceq when in fact it was 1/Ceq. c. -- www.cesaremarilungo.com

Cesare Marilungo wrote: A little more fraction work than that, I fear. It should be the same formula as calculating the total resistance of resisitors in parallel: 1/Ct = 1/C1 + 1/C2 etc. You gotta find the lowest common denominator and convert the caps to the LCD, add them and take the recripcle of the sum of the caps. -- Phil My Home Page: http://fancypiper.info Our 2nd CD: http://www.cdbaby.com/naomisfancy Naomi's Fancy performances: http://naomisfancy.virtualave.net/schedule.html

fons adriaensen &lt;fons.adriaensen(a)skynet.be&gt;be>: wrote: ardour master bus -> spdif out / da -> pa / speakers -> jaaa So what I hear is what you say is called "intermodulation distortion" from the speakers. I don't see it in jaaa either. Point taken. Now given the non-linearities others and you (below) describe and given this is about mixing discrete signals (not a summing recording of the audible range) and given there are instruments that emit frequencies or produce harmonics well above the audible range (violins, Brian Wilson 40 years ago) it makes sense to record it, doesn't it? With higher samplerates? btw: I couldn't find the command line switch to give jaaa instances a jack-name in order to run more than 1, it doesn't seem to show up in jaaa -h? Oh dear, yes. I'm starting to "hear things" already, http://hyperphysics.phy-astr.gsu.edu/hbase/sound/subton.html notes a "trio for two flutes". That I would like to hear. -- Wolfgang

On Sun, 2006-01-29 at 13:40 +0100, Wolfgang Woehl wrote: You should also get 900Hz. They're the sum & difference frequencies. Apparently broadcast people called them sidebands? They're also the reason that barbershop quartets get "ringing" chords. I bumped into this page that explains non-linear distortion: http://www.isvr.soton.ac.uk/SPCG/Tutorial/Tutorial/Tutorial_files/Web-heari… bye John

On Saturday 28 January 2006 08:11, Wolfgang Woehl wrote: To elucidate a wee bit, my comment was meant to be refering to mixers in the electronic sense. Mixing products of just two tones will contain each signal, and the sum and difference frequencies of them will be determined by the amount of non-linearity in the mixer. Trade the term non-linearity for distortion and it might become clearer. In audio work, the target is of course no distortion, but getting rid of that last .1% is what you spend unseemly amounts of money on, and play pissing matches as to what method is better, yadda yadda. Of course in receivers, the so-called local carrier is driven into the mixer quite hard so the resultant output contains as much of these sum & difference frequencies as possible. And then we filter for the sum, or the difference, and use that as the intermediate frequency of the classic superheterodyne receiver. However, one tends to forget that the ears own response also generates its own distortion, primarily because the amplitude vs loudness is fairly close to an inverse square law function for single tones, and is why the db was invented all those years ago. Put the human ear into a multitone situation, and there can be some detected distortion/aliasing heard even for tones coming from two completely independent amplifier/speaker setups where there shouldn't be any interaction between them. I'm of the opinion that those folks who can lay claim to a golden ear, either don't have as much of this effect in the first place, or their ears have trained themselves into totally ignoreing that portion of the distortion which is ear generated. I was once such, 50 years ago, but alas, at 71, tinnitus has set in and while I can still hear bad audio quicker than many younger folks, it is not as nerve wracking as it used to be. One could say I've developed a tolerance for it I guess. But, because I have been a broadcast engineer, now almost retired, for much of my working life, I still need to hear when things aren't right, so I can fix them before those emails start coming in. :-) -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2006 by Maurice Eugene Heskett, all rights reserved.

On Saturday 28 January 2006 13:47, Ismael Valladolid Torres wrote: No, of course not, then it was phone calls & letters, but they sure do the email thing today. I think I got my first email from a viewer in the middle-later 90's after we had setup a web page at wdtv.com. About 9 years ago now. One of the first things we did was to write an arexx script that grabbed the teleprompter's script contents from the NT based ap-news server, striped the control stuffs, wrap it in html, and put it up on our web page 5 minutes after that newscast started. That caused a minor revolt in the news dept cause we got to give them heck over the typo's & shorthanded words only they knew the meaning of. We may even have been first in the country to do something like that. It wasn't much, but some deaf folks could get it by closed caption generation too. Then the commission finally made us write down every word said on the air just because of the deaf, and things went to from mearly bad, to even worse cause most of the script writers had to make so much output in a limited time that spelling & typo's got really bad, & still is. With the time constraints, its not likely to get markedly better either. Even so, I think the CC watching people are getting a better product than they were before. Some stations use voice recognition software to do that, but it makes more mistakes than the reporters do. Closed captioning labor time is now a measurable cost of running a news gathering operation today. Just offhand, I'd say we have 1.5 more people involved in an average 30 minute newscast now than we did 10 years ago before closed captioning made it to the masses, just to do the closed captioning, but video editing gear has gotten better & faster so its a wash in the actual body count. We do about 2.5 hours of news daily M-F, with a reduced schedule on S & S. Our field cameras now record on hard drives, and the news story doesn't see tape until its dumped to archival storage on dvc-pro tapes long after the newscast has been put to bed. During the actual newscast its served from apple video servers all hooked to the news dept editors by 100-base-T ethernet, with editing of stories actively going on during the cast. It used to take an editor 30+ minutes to do a 1:30 package by editing tape, now it can be, with good notes from the camera reporter (usually the same person), maybe 5 minutes to locate the right stuff in an 18 minute rambling interview and have it ready for air. Morning 5 minute cut-ins run 20-30 minutes total from 6 to 9ish, 30 minutes at noon, 1.5 hours 5-6:30, and 30 to 45 minutes at 11pm. This is all local stuff, national gets its breaks and the 6:30-7 slot. So yes, the broadcasting business today is very much a 2 way comm channel, thanks to email. -- Cheers, Gene People having trouble with vz bouncing email to me should add the word 'online' between the 'verizon', and the dot which bypasses vz's stupid bounce rules. I do use spamassassin too. :-) Yahoo.com and AOL/TW attorneys please note, additions to the above message by Gene Heskett are: Copyright 2006 by Maurice Eugene Heskett, all rights reserved.

Esben Stien escribe: Yes! These are maths, and I wouldn't mess about maths, they always win. Cordially, Ismael -- Tout fourmille de commentaries, d'auteurs il en est grande chert� http://lamediahostia.blogspot.com/ http://www.flickr.com/photos/ivalladt/

Carlo Capocasa wrote: Cool! So we can throw away all out recording equipment finally. Now we can resynthise every sound accurately at playback time... This is the future of sound reproduction! I don't know why we didn't do this years ago... :o) Michael

On Sat, Jan 28, 2006 at 11:04:37PM +0100, Carlo Capocasa wrote: But just like vectorizing a bitmap, it's an inherently lossy approach. To me this just sounds like a description of ogg, mp3, etc. -- Paul Winkler http://www.slinkp.com

Hans Fugal escribe: And when you say *perfect* you mean *mathematically perfect* which is even more perfect than *acoustically perfect*. It's top of perfection, full stop. Let's repeat it until getting tired so Google's cache doesn't get full of misinformation. :) Cordially, Ismael -- Tout fourmille de commentaries, d'auteurs il en est grande chert� http://lamediahostia.blogspot.com/ http://www.flickr.com/photos/ivalladt/

On Sat, 28 Jan 2006 at 11:27 -0800, Mike Taht wrote: I'm sorry, I don't understand this. Can you elaborate? If you have 5:1, that means you have six channels, right? How does that relate to sample rate? -- Hans Fugal ; http://hans.fugal.net There's nothing remarkable about it. All one has to do is hit the right keys at the right time and the instrument plays itself. -- Johann Sebastian Bach

On Sat, 2006-01-28 at 13:21 +0100, Carlo Capocasa wrote: To quote a friend: (tanh(sin(2*pi*(tanh(((sin(2*pi*(t+1/16)+sin(2*pi*(t+1/16)+ sin(2*pi *(t+1/16)+sin(2*pi*(t+1/16))/2)/2))+1)-2)*2)+1)*8)* (tanh(((sin(2*pi *(t+1/16)+sin(2*pi*(t+1/16)+sin(2*pi*(t+1/ 16)+sin(2*pi*(t+1/16))/2) /2))+1)-2)*2)+1)*6*(tanh(((abs( sin(2*pi*t/90-sin(2*pi*t/45)/2))-1) *2)+1)+1))/2*tanh(sin(2* pi*t/180)*20)+(sin(2*pi*t*f*2^((2*(int(cos (pi*int(t*4)/2)+ cos(pi*int(t*4)/4)))-24)/12)+(sin(2*pi*t*(f+5)*2^((2 *(int( cos(pi*int(t*4)/2)+cos(pi*int(t*4)/4)))-36)/12)))*(1-2*abs(1- t %0.5))*8*(tanh(((sin(2*pi*t/180-sin(2*pi*t/90)/2)-1)*2)+1)+ 1)) *sin(2*pi*t*2+abs(sin(2*pi*t*2+abs(sin(2*pi*t*2)*0.5))))/ 16+sin(2 *pi*t*f*2^((2*(int(cos(pi*int(t*4)/2)+cos(pi*int(t*4 )/4)))-36)/12) +(sin(2*pi*t*(f+5)*2^((2*(int(cos(pi*int(t*4)/ 2)+cos(pi*int(t*4) /4)))-48)/12)))*(1-2*abs(1-t%0.5))*4)* sin(2*pi*t*2+abs(sin(2*pi*t *2+abs(sin(2*pi*t*2)*0.5))))/2)* tanh(sin(2*pi*t/180)*20)+(tanh ((sin(2*pi*t*f*2^((2*(int( cos(pi*int((t-6/8))/2)+sin(pi*int((t -6/8))/8)))-0)/12)+sin(2* pi*t*5)/2) *(tanh(cos(2*pi*(t-2/8))*5) +1)+sin(2*pi*t*f*2^((2* (int(cos(pi*int((t+6/8))/2)+sin(pi*int((t +6/8))/8)))-0)/12)+ sin(2*pi*t*5)/2)*(tanh(cos(2*pi*(t+2/8))*5)+1)) *(tanh(sin(2* pi*t/180)*2)/4+sin(2*pi*t/180-sin(2*pi*t/180))*0.78)) /4+ tanh((sin(2*pi*t*(f+1.2)*2^((2*(int(cos(pi*int((t-6/8))/2)+ sin(pi*int((t-6/8))/8)))-0)/12)+sin(2*pi*t*5)/2)*(tanh(cos(2* pi*(t-2/8))*5)+1)+sin(2*pi*t*(f+1.2)*2^((2*(int(cos(pi*int((t+ 6/8)) /2)+sin(pi*int((t+6/8))/8)))-0)/12)+sin(2*pi*t*5)/2)* (tanh(cos(2 *pi*(t+2/8))*5)+1))*(tanh(sin(2*pi*t/180)*2)/4+ sin(2*pi*t/180 -sin(2*pi*t/180))*0.78))/8)/5)*0.9 f=440 Also audible as an mp3 at: http://www.mikseri.net/elektrojaenis (It's one of the songs, just press the download link above the formula) A bit OT though, as it's made with Goldwave in windows -- Sampo Savolainen &lt;v2(a)iki.fi&gt;

On Friday 27 January 2006 18:31, Brent Busby was like: As a relative know-nothing in this area. I have to say that arguments like this are winning me over. Anything that would reduce that digital tinniness has got to be worth pursuing. Surely? Having recently had the chance to experience 24 bit audio, I would say that although the sounds are undeniably richer and more acceptable, there are still harmonics, particularly noticeable at root note >=C5 where the harmonics sound brittle and unnatural. Of course it is possible that due to weaknesses in the signal chain, I'm not hearing 24 bits at all. Thanks for this discussion. I am slowly beginning to understand these issues better. -- cheers, tim hall http://glastonburymusic.org.uk/tim

the emerging hi-def standard for non legacy-media platforms is Intel HD Audio (the successor to AC'97) and its format is: up to eight channels at 192 kHz/32-bit quality i work at 32f/44.1khz, however when ripping nicely recorded and mastered vinyl records (tino's breaks comes to mind) the drums sound more dimensional and roomy at 96khz..blind tests have shown sounds above the range of hearing (often quoted as up to 30khz or so) affect the 'air' of a sound, and of course there can be beyond-nyquist OOB aliasing..a good analog tape deck or studio monitors can offer a response to 50khz..you will have to decide the tradeoff that suits you, now that hardware is not dictating the limits..

On 1/27/06, Florin Andrei &lt;florin(a)andrei.myip.org&gt; wrote: Actually in the case ardour/jack et al 24 bit is more convenient, since instead of 24 bit signed linear format, we use 32 bit floats, which have just enough dynamic range to fit 24 bits. 32 bit floating point is much more convenient than any fixed point format... -- Mike Taht PostCards From the Bleeding Edge http://the-edge.blogspot.com

Esben Stien &lt;b0ef(a)esben-stien.name&gt; writes: Well, it's wrong to say that. Any signal below the Nyquist frequency can be recreated, but with less values, the electronics would have to work more to render these middle values, but that should be trivial. It seems I've defeated myself and my arguments are wrong. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@ n n