On Sat, December 22, 2012 7:42 am, Ralf Mardorf wrote: It would remove two conversions from the stream. Nothing against analog mixers, but starting with digital, converting to analog and then back to digital has got to loose something. On the other hand it could be a pain having digital audio coming in at 48k on a 44.1k or 96k project. I am sure the synth has one internal sample rate and to add more would mean resampling to make it sound right. The problem with ADAT is that it came from the ADAT (no kiddin') and is seen as a 8track street. What would make it a lot more useful (and expensive?) is that every adat unit would do through work. So a 2 channel adat IF would have an adat in and allow routing however many tracks come in around its own tracks. So in the case above with the 6 channel synth there should be an adat in where two more channels could be routed to 7+8.... But that is not part of the spec. -- Len Ovens www.OvenWerks.net

On Sat, Dec 22, 2012 at 12:39 PM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net>wrote;wrote: and what card would that be? the usual ralph misinformation. does it ever stop? ADAT supports 44.1kHz and 48kHz as-is. With the S-Mux "standard", it can route 88.2kHz and 96kHz signals, but each channel is split across two ADAT channels, causing a 50% reduction in the number of available channels.

On Sat, 2012-12-22 at 19:00 +0100, Ralf Mardorf wrote: PS: I'm a human and I make mistakes. However, I'm still waiting for your reply. Perhaps you noticed Paul, that last time you tried to blame me, it was you who was mistaken. This was a thread about the -Xalsarawmidi switch. You might want to read this one http://comments.gmane.org/gmane.linux.audio.users/78891 I can post several mails were I was mistaken, but I can post double as much where you was mistaken. I guess we shouldn't do such stupid things. You very often abuse people when you think that you are right, not only me. Perhaps you can stop this in the future? I don't care that much about it, but I know at least two other people who were hurt by you. Why are you doing this that often? However, I'm not hurt, it's ok for me, so don't worry. Regards, Ralf

On Sat, 2012-12-22 at 16:53 -0500, Paul Davis wrote: That's correct. I was mistaken, but you was mistaken too ;). Please, lets forget about this "battle". IMO it's more important how we can use Linux with what interfaces ever, to get the best quality regarding to the sound, at less costs. I need 48 KHz with the best bit rate I can get. I don't like less than 48 KHz and I'm unable to hear that > 48 KHz does improve something, excepted of my 2 TerraTec cards, for them 96 KHz does sound better, might be a converter issue. Regards, Ralf

On Sat, Dec 22, 2012 at 08:53:10PM -0500, Thomas Vecchione wrote: [ about using 44.1 kHz sample rate ] Some people believe the strangest things. That doesn't make those things true. SRC amounts to applying a low-pass filter. That's it. There is no other magic involved. The fact that this filter is producing output samples at a rate which is different from the input sample rate does not change it in any fundamental way. It just means using different coefficients, which are no better or worse than those used by a 1:1 rate filter. The operation will add some round-off noise, but no more than a simple EQ or even a fader. That noise will be at a level at least 40 dB below what will be added by reducing to 16 bit for a CD. Which means it makes a difference of less than 1/1000 of a dB, if anything at all. It does not add distortion or degrade the signal in any way. If you're looking for that, you'll get a zillion times as much by using even the simplest dynamic effect. Given that most audio interfaces will perform better at 48 kHz than at 44.1, if there is any significant difference at all you're probably better of producing a CD at 48 kHz and converting it to 44.1 right at the end of the chain, the same place where you reduce to 16 bit and add dithering, for example in Ardour's export dialog. There is no good reason to use 44.1 kHz *ever*, except for the signal stored on an audio CD. True. In particular electro-acoustic composers using Max/MSP on their MACs all seem to use 44.1 kHz, it's a continuous source of trouble at the LABEL and the CdS. Must be something with the defaults of that SW or system. In some cases they weren't even able to switch to 48 kHz. -- 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)

On Sun, Dec 23, 2012 at 5:45 AM, Fons Adriaensen <fons(a)linuxaudio.org>wrote;wrote: http://xkcd.com/285/ I've always had fine results running my Delta1010s at 44.1kHz, since I only do audio destined for CD and wanted to avoid the re-sampling. Why would running them at 48kHz make them perform better? -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa

Gibbs effect ripples aren't a filter artifact. They're a feature of any bandlimited signal. Using a less-sharp filter does not eliminate or mitigate the effect, it simply changes the appearance. It's no different from a little kid spreading his peas all over his plate so that it looks like there's less there and he's eaten some. Spreading the transition band over more space to make it less noticeable is a) irrelevant once the transition band is above the hearing range and b) has nothing directly to do with the ripples. Ringing from an IIR filter with significant internal energy storage also happens, but that's a different phenomenon. Monty

On Sun, Dec 23, 2012 at 10:05:22PM +0000, Fons Adriaensen wrote: So artifacts (transient peaks) on the trailing or leading edges of a square wave is not a form of ringing? I remember years ago when I was working in Telecom (then Post Office) and me and another guy were interested in checking out some VHS VCR's, so we "borrowed" some models which we would return on Monday. This other guy knew what he was doing and we also had access to quite a bit of test equipment. One test I remember we did was to feed a square wave (which was available as a reference signal from the actual o'scope we were using) into the VCR and view the output on the o'scope. On one model (Sharp, I think) it was distorted to hell, it looked more like a bad sine wave than anything resembling a square wave. The weird thing was that it sounded clearer than the other models, you could hear paper rustling and clothes brushing against things very clearly. On another model, the output was not a bad representation of the input signal, but there were noticeable peaks on the trailing edge of the square wave. If my memory serves me correctly, he referred to this as ringing, caused by the filtering stages in the circuitry. Also I remember that he said that if the peak was on the leading edge it was filtering on the input stage of the circuitry and if it was on the trailing edge it was caused by filtering on the output stage of the circuitry. That test alone made me realise that a simple listening test is not sufficient. I'm guessing that the Sharp model would produce "listening fatigue" long before the other model which had minimal distortion on the output squarewave. -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X

Well, actually, that's not entirely true... you're also potentially trading off time resolution for frequency resolution. Beyond the scope of the discussion, but important for anyone actually following along in detail. Monty

On Monday 24 December 2012 10:37:40 Fons Adriaensen did opine: Absolutely. However, the response of a passive filter is 100% predictable. Enough math applied to that distorted waveform can tell you very accurately what the bandwidth and group delay characteristics of that path the square wave followed from the input jack to the output terminals. There are many other things involved, and there can be, in some amplifiers, such as the high powered klystron used in tv broadcasting back in the day of 2 cents a kilowatt electricity, a delay that varies with the power level precisely because of the relativistic effects imposed on the electron beam that does the amplifying. It is a beam velocity modulation and because of the e-mv2 relationship, the positive swing of the input cannot accelerate the beam quite as much as it can slow it down on the equal negative swing. So that device is electrically longer at the high power levels of the sync tip portion of the old ntsc signal. It is also called incidental carrier phase modulation or ICPM in shorthand speak. And it is not easily corrected, so the sync waveforms were horribly distorted by having the high frequency edges moved as much as 200 nanoseconds to the left. It was also the cause of about a 20 db loss in the aural SNR in the viewers receiver because the receivers used the intercarrier frequency only in detecting the sound, that imposed that ICPM as a background buzz in the audio. I didn't fully understand just how bad it was till a circuit breaker running the coolant pump failed, single phasing that 3 phase 10hp motor, and the electron beam burned a hole in the klystrons collector, filling it with water. That hole was a $150,000 hole, and 2 months running at low power because I wheeled the 2200 lb aural dolly with an old tired klystron in it into the visual cabinet, tuned it even more broadband than usual and, using an N 't", fed some of the aural drive into the visual tube. So we had about 25% of our normal power on the air. Because with both signals being subjected to identical ICPM effects, and the receivers operating on the frequency difference only, the normally 45 db down audio 'sync' buzz was gone, and the on air aural SNR was suddenly close to 72 db! When the new tube arrived, I almost didn't want to go back to full power because I knew that &*^*& buzz would be back. I did get comments on it from the viewers too, they noticed it right away. Yes you can, particularly from the aspect of how good the anti-aliasing filters are. 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) My web page: <http://coyoteden.dyndns-free.com:85/gene> is up! "The Amiga is the only personal computer where you can run a multitasking operating system and get realtime performance, out of the box." -- Peter da Silva I was taught to respect my elders, but its getting harder and harder to find any...

On Thursday 27 December 2012 08:01:09 Ralf Mardorf did opine: Ralf, if he isn't competent with a hot soldering iron, he's a 'paper' engineer only. I have known a couple like that, I wouldn't let them in the same building as a soldering iron with an intact line cord. I hope Santa was good to you. 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) My web page: <http://coyoteden.dyndns-free.com:85/gene> is up! Dammit Jim, I'm an actor, not a doctor. I was taught to respect my elders, but its getting harder and harder to find any...

On Sun, Dec 23, 2012 at 01:07:19PM -0500, Joe Hartley wrote: The usual combination of technical and economic factors. Every AD or DA needs an anti-aliasing filter. Say you want a flat response up to 20 kHz, and at least -120 dB at half the sample rate. At 44.1 kHz that means roughly -60 dB/kHz, at 48 kHz -30 dB/kHz for the transition band of the filter. Now the complexity of an HP filter (the number of coefficients) is proportional to those numbers. That means that for the same specs at 44.1 kHz you need twice the number of gates, chip area, power etc. as at 48 kHz. That is a simplification, if you take everything into account the ratio is even bigger. This, plus the fact that 44.1 kHz is considered to be a 'consumer' rate anyway means there is a strong incentive for AD/DA chip designers to cut corners at 44.1 kHz. And that is exactly what happens in almost all cases, with very few exceptions. The alternative, which makes sense, is to forget about HW support for 44.1 kHz and let the driver resample from 48 to 44.1 if the user wants to use that rate. This is also valid for higher sample rates. At 96 kHz, assuming you want a flat response up to say 32 kHz, you only need 7.5 dB/kHz. Since the _delay_ of the filter is also proportional to the transition band slope, such a filter can have much less latency than at 48 kHz, and that's the reason why e.g. digital snakes and mixers designed for PA use operate at 96 kHz. Low latency is essential for PA, in particular if in-ear monitors are used. 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)

On Sun, Dec 23, 2012 at 4:35 PM, Fons Adriaensen &lt;fons(a)linuxaudio.org&gt;wrote;wrote: If this is the case, why would we have several different resampling algorithms with varying levels of quality? All very true, and I certainly don't dispute this, but also I am not sure as to why exactly it is relevant to this conversation either. Seablade

On Sun, Dec 23, 2012 at 05:05:47PM -0500, Thomas Vecchione wrote: The usual tradeoff between quality and CPU use. Today it shouldn't matter at all for stereo. But if you have 64 channels to resample in real time it still does. 'Quality' is a bit of a misnomer in the case of e.g. zita-resampler and the 'sinc' options of libsamplerate. The only difference is in the frequency response for frequencies near half the sample rate. You mentioned that resampling would add noise. It does, but as shown above the noise you have anyway is already at a much higher level, so anything added by resampling will not make any difference. 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)

On Sat, 2012-12-22 at 20:53 -0500, Thomas Vecchione wrote: Now you spread misinformation ;). Is there any scientific source for that claim? My impression is (might be scientifically right or wrong, it's my experience) that - < 48 KHz sound can be ok for some sources, but it also can become very bad - > 48 KHz you don't hear an improvement, IOW 48 KHz is all that's needed - There is some gear that does sound better at > 48 KHz, but if you compare that gear with other gear at 48 KHz it doesn't sound better I don't have experiences with conversion from 48 KHz to 44.1 KHz, when I did it (doing something a few times doesn't lead to experience), the sound becomes less good, that's the nature of 44.1 KHz. I don't know any scientific reason that a conversion should cause issues, but producing at 44.1 KHz does cause issues, since the sound is less good, mixing is harder to do. I was asked how the sound quality is, when I use 192 KHz with the RME card, but I can't answer this question, because I don't record at 192 Khz. I guess the bit depth is important, especially for production, perhaps less important for listening to the finished recoding. YMMV! Regards, Ralf

On Sun, Dec 23, 2012 at 12:39:17PM -0500, Thomas Vecchione wrote: Yeah, but what about harmonics? For truly PRO work, analog should be used. CD's came out as the poor mans quality stereo, it was a compromise for high quality vinyl. Just as McDonalds is a poor compromise for a quality restaurant. But with the proliferation of advertising, huge product selection, the rising cost of the real quality goods; digital and McDonalds soon became the norm. I believe it was Shannon's law which basically stated that the sampling frequency should be twice as high as the highest frequency (or bandwidth, if the lowest is zero). This was used in telecomunications where a 4kHz (actually 3.4kHz but usually rounded off to 4kHz) bandwidth was considered to be ample to convey human speech at a quality which was considered acceptable. Therefore, an 8kHz sample rate was used. I believe the same reasoning was used when they started fabricating CDs. In all fairness, if the source of the music is digital, then digital mastering throughout the whole process is acceptable (the DDD) on the cover of the CD usually (used to?) indicates this. Others are AAD and ADD, where the A stands for analog. So, consider a musician playing a Stradivarius violin. You'd want to be able to reproduce all the tonal richness, and harmonics during the recording phase. Some people say the ear can't hear above a certain frequency anyway, so it doesn't matter whereas there are others who can pick the difference between a Stradivarius violin and an "ordinary" violin. I also understand that vinyl is now increasingly (very slowly though) becoming the preferred medium for listening to music. Remember: If you sample at an infinite frequency you have analog, and isn't the idea of quality digital to have a high sample rate. -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X

On 12/23/2012 10:14 PM, Monty Montgomery wrote: I was gonna reply to him, but decided to do a face-palm, and have a cup of cocoa instead.

and practical reasons too (compatibility with existing technology). See: http://en.wikipedia.org/wiki/44,100_Hz

On Mon, Dec 24, 2012 at 9:27 AM, Fons Adriaensen &lt;fons(a)linuxaudio.org&gt;wrote;wrote: although i'm skeptical, i'm willing to leave a tiny amount of doubt in the air. i think the evidence is clear that we cannot hear these frequencies. there is some slightly woo-ish stuff about how they might still interact with us physically and contribute something to the experience of being "in the presence of" the live instrument(s). i suspect its totally bogus, but i also don't think that the science has been done to clearly establish that it is, and that 22kHz is a hard limit for human experience, not just human hearing.

On Mon, 2012-12-24 at 09:48 -0500, Paul Davis wrote: At least inaudible low frequencies seem to have impact to the humans cognition. If inaudible frequencies are important for audio productions is another issue. At least inaudible frequencies are noticeable. Regards, Ralf

What about http://www.pnas.org/content/early/2012/01/02/1114999109.abstract ? The study was criticized (eg. http://www.violinist.com/blog/laurie/20121/13039/) but still, people today build violins and some professional players like those new ones, just like they like the old stuff. That said I'm sure a violin 10,000 euros is way better than a 100 euros one. But from there to "a stradivarius is way better than what violin makers of today build" (if that's what you meant), there is a lot of hype and magic thinking involved, which is very surprising from you.

On Mon, Dec 24, 2012 at 10:40 AM, Fons Adriaensen &lt;fons(a)linuxaudio.org&gt;wrote;wrote: the guy from the chicago symphony who makes carbon fiber violins still seems to have all of his hair and digits, and i believe they are selling reasonably OK. a luthier i talked to in berlin told me that most of the magic of old instruments isn't the techniques but simply the "physics" of what happens to wood as it ages. the lignin changes its structure over long periods of time, as do other components. he told me that the hardest part of becoming a luthier is that it takes at least a decade to get your supply chain established, a supply chain that lets you build instruments with wood that has already aged for at least a decade. he said that the changes in the wood structure continue for at least 100 years, but they become much smaller with each passing decade. there is very little difference at all between a 200year old violin and a 100 year old violin, but a huge difference between one made today and one made 100 years ago, even if the techniques are identical. i liked the guy a lot, his studio was beautiful and he had a degree in physics and mechanical engineering from a major US university. i chose to believe him :)

On Mon, Dec 24, 2012 at 10:46:05AM -0500, Paul Davis wrote: For your amusement on Xmas evening: <http://www.luisandclark.com/introducing-the-violamatic/> Cheers ! -- 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)

On 12/24/2012 10:40 AM, Fons Adriaensen wrote: I read about this guy in 1983, in a science magazine. I think he cracked the code: http://www.nagyvaryviolins.com/

On Mon, 24 Dec 2012, Thomas Vecchione wrote: Actually I think the thing with analog is that while it's not sonically accurate, the inaccuracies are often musical and pleasant (though that may be more true to older people who are used to the sound of tubes and tape than younger people who are used to MP3's and digital aliasing artifacts). A lot of the warmth that people associate with an analog signal is harmonics that were not even in the original signal to begin with -- but few can argue that they sound good! That's the whole point of tube compressors and preamps...not to sound accurate, but to sound even better than accurate. After a lifetime of listening to tape saturation and harmonics from tube coloration, accuracy is so...boring. The really nice part about that (especially for people like me who actually have damaged their hearing in some frequency zones due to unwise loud playing of the ride cymbal bell and such things as that) is that fortunately, most of the frequencies that really make people groove aren't up in the >10kHz range anyway. People really respond to bass and mids, even upper midrange, but if you give people too much high treble, unless it's really smooth, it can be described by so many fairly negative adjectives: metallic, harsh, cold, tinny, abrasive, clangy. Most people like warm recordings. Those near ultrasonic frequencies usually just aren't that big a part of the music. Actually, the original poster is correct -- vinyl is the connoisseur's prefered format...but for reasons that most of the people using it don't understand themselves. One would be correct to point out that it's an inferior medium. Its frequency response is poorer, it wears out every time you use it, the stereo channel separation is awful, it's plagued with background hiss and rumble, and it has limited dynamic range. But... * Recording and mastering from multitrack tape to vinyl is a process that is over eighty years old. There has been time for masters to learn from masters, and for their masters to learn from other masters. It's an art and a science in itself now. By the time the zenith of the analog era came (late 70's to early 80's), there were recording engineers who were positively wizards at making that format sound good. Consequently, most of the LP's one might ever hear sound absolutely marvelous in every way that musically counts, regardless of whatever shortcomings the actual medium might have. * Digital recording and mastering to CD on the other hand has been a comedy of errors since the day it came out, even if for no reason that's actually a fault of the actual media format. In the 80's, people were using terrible analog/digital converters, often handled audio at 16-bit depth from start to finish with no headroom for errors, and were directed by the studio lords to hastily convert as much of the labels' back catalog of albums to digital as fast as possible (which they did...sloppily). Consequently, to this day, half the best albums of the 70's and 80's are best heard on vinyl, because the CD versions were done with no love (or brains). * In the 90's, analog/digital converters continued to be horrible, and digital recording was still a learning experience even for the "pros." Another trend I remember from that era was the practice of putting lots of digital reverb on almost every track...probably a bad thing to do during an era when digital reverbs were...not exactly the best I've ever heard. Though that part in particular had nothing to do with CD's themselves, that too probably contributed to making albums sound even more grainy and metallic than they were already bound to do. Also, one word: ADAT. Enough said there... * Now, in the 2000's, we have the so-called "loudness wars," in which every commercial production house seems mentally driven to make dynamic range a thing of the past. Since the albums of this era are seldom put out on vinyl, this guarantees that the follies of the 2000's will not taint vinyl's good reputation for excellent production (even if the reasons have nothing to do with vinyl). What we have here is going on almost thirty years of a good format almost never done well, and an older poor format almost always done well. The listener only cares that s/he gets an album that sounds good. Regardless of the fact that vinyl is an inferior format, are the listeners really wrong to choose the format that gets them their music in a way that sounds right? The fact that they're wrong about the reasons doesn't take away from the fact that, in the end, a vinyl record is far less likely to be a butchered up product than a CD. Wow...that ended up being longer than I thought. (And let's not even get into why people like analog synthesizers better. That's a related but still different conversation...) -- + Brent A. Busby + "We've all heard that a million monkeys + Sr. UNIX Systems Admin + banging on a million typewriters will + University of Chicago + eventually reproduce the entire works of + James Franck Institute + Shakespeare. Now, thanks to the Internet, + Materials Research Ctr + we know this is not true." -Robert Wilensky

These two things are not unrelated. :-) Also, the loudness wars are nothing new. Everyone noticed long ago that louder recordings sold more records. Even in the 1950s, pop acts pushed mastering engineers to make a recording as loud as possible. Bob Ludwig tells some good stories about this. The only reason it didn't get really bad until the 1990s/2000s was limitations in the technology. Digital techniques took the brakes off. [...] The early days of tape and vinyl were no monument to best practices. Digital is still suffering from engineers practicing out of superstition rather than knowledge, but it's still come of age much faster than vinyl did. The convertors were not as advanced, but they exceeded the preceeding analog technology in every way. I rather disagree. They're not as textbook perfect overkill as they are now, but the technology was already fully modern by the mid 90s. Setting aside bugs of course. Eg, ESS stumbled badly a few times in the early days. Monty

On Tue, 2012-12-25 at 14:33 +0100, Jeremy Jongepier wrote: A friend switched to CDs only and he e.g. can't get "The Cry Of Love" from Jimi Hendrix on CD, since production has stopped. He easily could get the same old vinyl I've got. I experienced, that e.g. the "I against I" and "Rock for Light" from the Bad Brains do sound much better on the original records, than on CDs I heard. Regards, Ralf

On Mon, 2012-12-24 at 15:29 +1300, Chris Bannister wrote: 44.1 was a compromise to get enough minutes on a CD, later when consumer DAT was introduced it wasn't needed to take care about the length, so they came with 48 KHz. Vinyl is another issue. I prefer vinyl, because records are similar to living beings. They get old, they get scratches, but there's no data loss, the mechanical principle allow it to play them with a needle on paper, there isn't the need to have a device with special encoding. The cover art is much nicer. Merry Christmas! Ralf

On 12/24/2012 10:12 AM, Ralf Mardorf wrote: Psychoacoustics are a very interesting and highly complex field :D Some people love vinyl (which is also a bandlimited medium, sice the record is non rigid and the needle has a non zero mass and some other reasons, too) because it sounds "better" to them. That has nothing to do with whether a vinyl record represents a signal more faithfully than a CD and a good D/A converter. I play guitar as my main instrument. I like tube amps better than most transistor amps simply for the reason that they add something to the signal (all these complex non linear components working together to produce something very pleasant to me). A simple transistor amp is measurably so much more faithful in amplifying my guitar's pickup outputs than a tube amp of the same complexity. The imperfections in this case are not undesirable, but rather they are experimented with and fully embraced by musicians (and audiences).. The same argument can be made for analogue equipment in the recording and production domain. Tube mic preamps, analogue circuits in mixing boards, etc.. And there's nothing wrong with that at all :D Some people really like the imperfections introduced by a vinyl playback chain. Here's a little thought experiment: Get the best available digital recording and playback chain, and make a great recording of your vinyl player's output for some vinyl records (preferably some that you are not familiar with). Now make a double blind test between direct playback from vinyl and the digital recordings of the vinyl. My hypothesis is that you cannot discriminate between the two in such a way that you'll find the direct vinyl playback better than the digital recording of the vinyl (in a statistically significant fashion). Or even reliably discriminate which is which (i.e. which comes directly from vinyl and which from the recording). Now do it the other way around: Take some digital masters of albums and cut vinyl records from then. Now do a double blind test. My hypothesis is that you will be able to discriminate the vinyl from the digital playback for the simple reason that the vinyl cutting and playback chain introduces loads of imperfections that will be clearly audible. You might prefer the sound of the vinyl in a sense similar to a guitar player preferring the sound of a tube amp over some transistor amp. But that doesn't tell us anything about the faithfulness of reproduction. If you like the vinyl cut of the albums better than you simply like the artefacts and imperfections. Analogue equipment has its place and knowing when to use it and when not and being clear about the reasons is a good thing. Claiming that analogue equipment is better in audio reproduction is just simply wrong as a blanket statement.. Flo -- Florian Paul Schmidt http://fps.io

On 12/25/2012 02:24 PM, Jeremy Jongepier wrote: Yes, that was my impression, too.. For a good vinyl cut you have to have a different mastering to work around the limitations of the medium.. Actually I worded that second point rather badly. The point was that the vinyl cutting process (including special/professional mastering) will introduce many more audible artefacts than the 44.1khz/16bit downsampling export for a CD. Vinyl is an audibly inferior medium to a good digital chain. So much worse that you need highly trained professionals to work around the limitations to create a bearable product. Some people might still prefer the sound. But that has nothing to do with vinyl being the measurably better medium. Flo P.S.: I still like my vinyl collection of records. The haptics. The nostalgia. The big pictures on the covers.. The handling of a pair of Technics MK1210 and a good mixer is still awesome for mixes, etc.. -- Florian Paul Schmidt http://fps.io

On Tue, 2012-12-25 at 14:41 +0100, Florian Paul Schmidt wrote: Not needed all the times, I own a JVC QL-A5 and removed the semiautomatic thingy. For the sound quality you only need a good pickup/needle and IMO start-up, pitching range, isn't that important, especially start-up is ok with less expensive turn tables too, when using a slipmat etc..IMO the Technics are only needed for DJs. Btw. many home users prefer belt drive. The main reason to use the Technics is, that crap as e.g. semiautomatic tends to get broken, but for my record player it's possible to remove the semiautomatic. Regards, Ralf

On Thu, Dec 27, 2012 at 07:09:51AM -0800, Len Ovens wrote: Except that it would turn the wrong way... When I was in broadcasting the turntables were EMT. They had a slip coupling between the main (heavy) platter and the actual (light) one supporting the disk. So with the motor turning you could stop / cue the disk by holding one finger to the rim of the upper platter. When released it would start 'instantly'... 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)

Something I recently wrote about 44.1, sound quality and esoteric home audio equipment: http://oldmixtapes.blogspot.com/2012/11/ultrasonic-speakers-hi-fi-high-samp… Neil -- DJ Dual Core's Blog http://oldmixtapes.blogspot.com/ Order without government; Peace without violence.

On Thu, December 27, 2012 2:37 pm, Neil wrote: I'm only surprised you didn't mention Aural Exciters, which used to be (don't know if they still are) used to add high frequency information (distortion) to the sound to make it more "exciting". As I recall (It's been a while) they tried to add even harmonic distortion. -- Len Ovens www.OvenWerks.net

On Thu, Dec 27, 2012 at 6:23 PM, Neil &lt;djdualcore(a)gmail.com&gt; wrote: There are even LV2 plugins that provide aural excitement (the Calf plugins provide one, for instance), although you can somewhat emulate it with a HPF and light distortion on that that filter. -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi

On Thu, Dec 27, 2012 at 03:46:49PM -0800, Len Ovens wrote: IMHO in most cases they sound horrible, and the sound changes all the time unless they are kept at exactly the same distance. I've started giving singers etc. a Beyer M58 instead. This is an omni dynamic, originally designed for ENG/EFP so it can take some abuse. It works very well for singers, and when used outdoors it picks up much less wind noise than an SM58. 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)

On Fri, Dec 28, 2012 at 9:38 PM, Len Ovens &lt;len(a)ovenwerks.net&gt; wrote: Actually monitoring is only one reason directional mics get used on a stage. The other is more simply preventing bleed and phase problems. Try close micing two rack toms for instance and you will rapdily discover the problem. Or try keeping the cymbal bleed out of the vocal mic. This list can go on for quite a while. Using Omnis on stage generally isn't a good idea, but can be done. For instance due to proximity effect, similarly to what Fons mentions, is why you will find musical theater always uses omni-directional microphones. Then again I don't see many concert mixer mixing a concert like a theatrical mixer mixes a live musical (At least a good theatrical mixer). You are literally changing levels every line. However you might get more detailed in other ways in concert mixing, for instance with effect processing and similar, but personally I find theatrical mixing a huge order more difficult than most concert mixing I do. Likewise I just this past week did a concert with a second drum kit on stage that my music director went to for one song, which was a collection of toms and a snare for a more high energy version of 'Little Drummer Boy'. They were all grouped together so I used a single omni mic to pick up the entire setup, it worked really well. Seablade

On Fri, Dec 28, 2012 at 06:38:24PM -0800, Len Ovens wrote: A directional mic wil give you a few dB advantage, but only if it is and _stays_ oriented correctly - something you can't always count on. OTOH an omni can be used very close more easily. Another thing that affects feedback and is often overlooked is the frequency response (of all components in the chain). Narrow peaks of a few dB (which will not even affect the sound quality) will negate any advantage there is in using a directional mic. 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)

On Sat, Dec 29, 2012 at 09:48:34AM -0500, Thomas Vecchione wrote: True if you consider only the direct sound, which may be valid in the case of monitors close to the mic. OTOH, if the feedback is mainly due to indirect sound which will come from all directions then a cardioid provides just 4.8 dB advantage. Which of course could make a difference. 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)

On 12/29/2012 09:55 PM, Len Ovens wrote: ... problem identified :) solving it is beyond the realm of the technical and may necessitate the use of blunt instruments. boosting is usually the wrong approach for pa filtering (unless you really know what you are doing). try to attack the boxiness instead (for me, the term "boxy" pretty much means 800 hz give or take a third). still - it's nice to have a decent mike on a good voice, where it will make a difference. will be solved by the blunt instrument mentioned above. at the risk of sounding like a teacher: try to avoid thinking in such imprecise terms. your analysis seems to imply that the midrange needs boosting. from what you described above, it's evident that the midrange is ok (the room even has a tendency to boxiness), it's just that the smilie-shaped main eq will over-emphasize bass and treble. so the way to tackle that is _not_ to boost the mid-range, but to correct the crooked curve on the pa. i honestly don't know why a 30-band eq lets you boost stuff. 99% of users would be much better off having an eq that only provides attenuation. makes sense, unless you have a little headroom on the pa and a guy at the controls during the show, in which case it's nice to reduce the amp volume a bit and give him/her a few dbs of actually working with the bass (for me, even as little as 3db SPL raise over the amp sound gives me a lot of freedom to shape the sound). plus for the house sound, heavy compression on the bass is often useful, and bass players are naturally reluctant to use it (and they shouldn't, because playing into heavy compression leads to sloppy technique, so keep the amp unsquashed). when i have to amplify the kick drum, i will put the bass on the pa as well. with small speakers, it's very difficult to create a kick drum that doesn't sound like a cardboard box thrown out of a window, but if the bass locks tightly, the combined sound can be quite impressive. to balance that, i need both instruments on the pa. when i work with bass players in small venues (where most of the house sound is from the amp), i let them set up their preferred tone, and then if the amp has a good eq, i will approach them during the soundcheck to correct house sound problems directly at the amp (again, attenuation only). often, it's just a few frets on a particular string that are playing up. honestly, an omni mike won't help you. fons does have a point in saying that omni mikes have some advantages and could be put to good use outside the realm of classical recording, but until someone sorts out your lead singer problem, that's academic :) get yourselves a sound guy to run your show who is thick-skinned enough to tell the bossman go pee, wait for the audience remarks about how good the sound was for a change to sink in, and then after one or two shows, you have yourselves a reference sound which you can try to recreate even without a sound guy at the controls. get a mixer with two semi-parametric mids, problem solved. very decent ones are available nowadays at very low cost. a year ago, i got the club i work for to invest in a new board (an a&h gl 2800 for about 3k euros plus taxes, which is really negligible for a professional venue), and the musicians have been more than happy with it. a board with that feature set and sound quality would have been high in the five digits 15 years ago. nowadays, there's really no need (and no excuse) to work with inadequate mixers if you're serious about your music. an omni has absolutely no advantage there. your singers will want to work on their microphone technique and exploit the near-field effect in a musical way. shotguns are not useful for music stages. they do have better side rejection in theory, but at the cost of a very irregular off-axis response. the few times i used shotguns in the theatre, i found they would give me even less gain before feedback than a standard supercardioid, and required insane filtering. -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

Am 29.12.2012 21:55, schrieb Len Ovens: it may be too far off to take into consideration, but you could test microphones by Bob Heil. He invented the talk box long ago but made some microphones that are used by some serious acts like Alice In Chains, Charlie Daniels, ZZ Top, Joan Baez, Kansas and Stone Temple Pilots as well as Coast Hills Church and Northpoint Church to name a few. I recently bought the Heil PR 35 for me as a singing drummer and also recommended the PR 20 and PR 22 as an announcer mic to my local church. Heil's PR 20UT model http://www.heilsound.com/pro/microphones/pr-20-ut costs $101 and is worth a try to the concurring SM58 at nearly the same price point. Your experience might be different, but they sound natural without needing much EQ. Just my 2 cents and God bless, Marius

On Sun, Dec 30, 2012 at 4:35 AM, Roger &lt;gurusonic(a)gmail.com&gt; wrote: Honestly Sennheiser is often a better choices than the standards of Shure(SM58, etc.), however I am finding people tend to vastly overstate them. Personally I am just not as happy with, particularly the low end 'e' series mics. The 9 series isn't to bad, but the 8 series I could do better than fairly easily in my experience. I also don't have specific brands mics I use all the time across the board, I tend to use whatever I find works best. Sometimes that is Shure, sometimes Sennheiser, sometimes Rode, sometime Audix... this list goes on for a while:) Particularly when dealing with vocal mics, it can be very important to have a mike that matches the voice. Otherwise the SM58, Beta58, and even Beta57, while not the best sounding mics in the world, you should be able to get a decent sound on just about ANY voice with. That is why they are still used. Seablade

On Sun, Dec 30, 2012 at 11:19:30AM -0500, Al Thompson wrote: It all depends on the circumstances. I never wrote that you could blindly replace all your SM58s with omnis. I wrote that I've done it in some cases, and with good results. Also note that the miniature DPAs often used on violins, horns and others (including singers) are omnis. 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)

On Thursday 27 December 2012 19:04:41 Len Ovens did opine: They weren't that tough. Our 5 car news staff used up a 10 pack of sm58's a year, and thought nothing of it. The old "not my car" attitude as they crawled out of the wreck. I never could figure out how they did it either, everytime I checked the car kits, the mics were reasonable well stowed. Cables were generally a mess though. This is the same news crew that would get smack in a deputies face as he was escorting the suspect through a hallway at the court house, get slammed out of the way and shortened a $7,000 camera lens about a quarter inch on a door frame, all in one 10 foot piece of video tape used. Then they'd bring it to me, complaining the back focus must be jammed, it wasn't adjustable any more. They had, with their attitude of get the story at all costs, just did $10,000 worth of damage to a $21,000 DVC-PRO camera, because when you jam the lens that hard, the prisms that separate the colors were usually cracked too, and that was a $5,500 item. After the second such in 3 months, we put them on notice that the next one doing it was canned, on the spot, and would be sued for the damages. They, with one exception to prove we were serious suddenly were a lot more careful. Where the screens tend to get plugged with dried spittle. I've run a bunch of them through the dishwasher. Not the mic of course, just the screens. 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) My web page: <http://coyoteden.dyndns-free.com:85/gene> is up! Don't SANFORIZE me!! I was taught to respect my elders, but its getting harder and harder to find any...

One more note (not a criticism at all, just more information); It's common for reconstruction anti-imaging filters to allow some aliasing in the reproduced transition band... that is, they don't slam to zero between 20050 and 22050, they actually use 4kHz, from 20 to 24kHz or so. Since the aliasing happens in the folded transition band, it's still all up above 20kHz. Embedded hardware is normally more constrained than software on full sized CPUs, and every extra gate is money (so I'm told)... OTOH, I'm also used to seeing modern Sigma-Delta designs do ~ no output filtering at all. You can't use them in demos with oscilloscopes :-) Monty

Thank you to everyone for the feedback. I have made a couple of corrections to the article based on the e-mail I got tonight. I try to do my homework when I write things like this but sometimes I reach beyond my expertise. Thank you for the help! http://oldmixtapes.blogspot.com/2012/11/ultrasonic-speakers-hi-fi-high-samp…

On 12/27/2012 02:37 PM, Neil wrote: Nice, thanks... Just a thought: a speaker with an extended frequency response (ie: able to reproduce ultrasound) could sound "better" because the high frequency driver is more linear and better behaved in the _audible_ range, and not because it is able to reproduce sound above our hearing limit... -- Fernando

On Mon, Dec 31, 2012 at 2:14 PM, Neil &lt;djdualcore(a)gmail.com&gt; wrote: It's a simple reason if you have the math background: signals with compact support in the time domain have infinitely long tails in the frequency domain. By having more high frequency range, you improve the temporal response of the transducer. At the ends of the frequency range, the group delay becomes significant and shifts some components of the signal more than others. A wide-band transducer with high damping is better able to reproduce the signals that you feed it, but also loses some of its capability to deliver power.

On Mon, Dec 31, 2012 at 3:17 PM, Neil &lt;djdualcore(a)gmail.com&gt; wrote: Yes, in general. In the case we're describing, the top end of the passband is outside the range of hearing. The phase shifts might still be audible at the low end of the transducer's range. However, phase shifts aren't much of a problem by themselves--they're all over the place in any typical multi-driver system and placement in the room matters just as much. The loudspeaker crossover introduces the same effect. Subwoofers are much more prone to phase shift problems. You notice it a lot when the bass seems to lag behind the beat--and this results from not having a continuous frequency response that goes down to 0 Hz (the ideal case) or bad resonance characteristics. Really clean sounds have a lot to do with the tightness of the temporal response--but there are of course tradeoffs in any kind of design choice like these. Chuck

On Mon, Dec 31, 2012 at 5:01 PM, Folderol &lt;folderol(a)ukfsn.org&gt; wrote: Complicated. To the best of my recollection... The hair cells are half-wave rectifiers and trigger on the positive zero crossing. They can fire, at fastest, about 1000 times a second. So, at lower frequencies, the hardware is in fact transmitting a mostly-phase-locked response to the incoming signal. This is part of how low-midrange frequency sounds are localized. Above 1000Hz the firing pattern becomes more like a power integral and phase information is ~discarded. At low frequencies, it appears that the neural hardware only pays attention to phase in the sense of comparing phase offset between the ears. The transition is gradual. So the ears hear absolute phase < ~ 1000Hz but it's not clear absolute phase is actually passed up the perception chain. Simple t-test should do it :-) Monty

On Mon, Dec 31, 2012 at 5:44 PM, Tim E. Real &lt;termtech(a)rogers.com&gt; wrote: turn up the rest of the audio system. The way our ears work, what frequency we actually hear shifts slightly with intensity. There is a great book on acoustics by F. Alton Everest that goes into this a bit in one of the earlier chapters. I am still reading through all of it, but it is an interesting phenomenon. Pitch is in actuality a subjective form of Frequency, not objective, and this is reflected in this phenomenon. the pitch goes down as the level of sound is increased. At high frequencies, the reverse takes place—the pitch increases with sound level. Locations 1553-1555). McGraw-Hill. Kindle Edition. There is a nice experiment involving two oscillators they use to demonstrate this. The book is well worth the money if you can get it and read English natively(Not sure what translations exist or how good they are). I am still working my way through it but it will likely take me two or three times through before I fully understand most of it. Seablade

On Tuesday 01 January 2013 14:17:22 Al Thompson did opine: Humm, I wonder if this is why much of Johnny Cash's work has him 25 to 50 cents flat. He was back on key for a while after the tonsillectomy. 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) My web page: <http://coyoteden.dyndns-free.com:85/gene> is up! My views <http://www.armchairpatriot.com/What%20Has%20America%20Become.shtml> If it's Tuesday, this must be someone else's fortune. I was taught to respect my elders, but its getting harder and harder to find any...

On January 1, 2013 07:26:42 PM Al Thompson wrote: Turn on the "Auto-Ruin" ! Tim.

On Wed, 2013-01-02 at 03:23 -0500, Al Thompson wrote: They wouldn't have used auto-tune that disgracefully. Auto-tune only is an issue for the most popular mainstream music nowadays. Take a look at the appearance (clothes, body language) of today's most popular mainstream musicians who misuse auto-tune in a very extreme manner. Just a special kind of "artist" is misusing auto-tune again and again. Happy New Year! Ralf

On Thu, December 27, 2012 9:19 am, Fons Adriaensen wrote: Ya, thats why it was a joke. It seems to me we pointed that out to whoever told us too. They were tough anyway. It's like the two inch video cart machines... the magazine never turned 1/2 turn to the required slot, but got there in time that they could switch from cart to cart every 5 seconds. The first ones could fit two carts in the space of one... but I could stand on one without damaging it. I saw that done too, but It seems to me it was just the mat slipping. The method used depended on the operator. -- Len Ovens www.OvenWerks.net

On Fri, 2012-12-28 at 15:56 +1300, Chris Bannister wrote: Most amps only support MM inputs and MM IMO is ok. The cartridge's company seems to be important. I still couldn't pay for a new cartridge, so I'm using an Audio Technica, because the old needle for a better cartridge gets broken. The Audio Technica does sound ugly, no bass, so I talked to one of those Technics MK2 users and he experienced the same. With a good cartridge my record player does sound better than my CD player. With the Audio Technica cartridge I can't stand to listen to my records, the CD player does sound better. When they introduced CD players, they gouged holes in CDs and then they demonstrated, that this doesn't effect the sound quality. They simply were lying, as we all know today, it's more likely that a CD is visually perfect, but anyway it can't be played any more. The reason to introduce "better" technologies usually is to make money. It's seldom that it's really better technology with a suitable better quality. I still own records from my childhood and I could hand down those records. So why buying all the I records I already own as CD, a media that won't last as long as a record does? Assumed the sound quality of new CDs is better, when comparing it with a new record. How will the sound quality be comparing a 60 years old CD with a 60 years old record? It's not important, since the top ten in 60 years are not the same top ten of today :D and the sound quality of the mobiles in 60 years is less good than what we already need to listen too, if we use public transport and have to listen to all those mobiles playing crap as Madonna and Black Eyed Peas by the mobiles "speaker". I noticed a loss in consumer audio quality and btw. consumer video quality might be better today, but video/audio sync is out of style. A better viewing quality with audio quality that still is ok for video isn't an improvement for me, when there is no lips sync any more. Today good consumer equipment is unpayable, there's no need to use the cheapest crap, but to get a real impression if e.g. a CD or a record does sound better, I guess we should compare an averaged good record player from the past, with an averaged good CD player from today. Comparing professional equipment is useless, since nearly nobody owns professional equipment. 2 Cents, Ralf

On 12/28/2012 07:25 AM, Ralf Mardorf wrote: Which cartridge do you have? I'm planning to have an Audio-Technica fitted in my SL1210 as I read some good reviews about them. Jeremy

On Tue, Dec 25, 2012 at 02:41:54PM +0100, Florian Paul Schmidt wrote: Really? I'm sorry, but if you judge a stereo system by the flashing lights and the number of knobs and dials available, then we sit on different sides of the fence. I believe in listening to music as the artist intended it. Don't get me wrong, there are heaps of crap vinyl recordings out there, that's for sure. -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X

2012/12/28 Florian Paul Schmidt &lt;mista.tapas(a)gmx.net&gt; Florian's mail is "The killer mail" for all of you guys writing of porno-technology... B-) One of C.Parker famous quote is "Master your instrument, master the music, and then forget all that bullshit and just play" but...according to this thread it seems that many are locked at the first step ;-) -r

On 28 December 2012 14:50, Chris Bannister &lt;cbannister(a)slingshot.co.nz&gt; wrote: This is closer to the opposite of the truth (counterintuitive as it may seem). But I think there have been two or three good expositions about this from Monty and others already in this thread. I like my records. I have a pretty good record player that's about the same age as I am, and I use it a lot. I like the physical objects, and there are recordings I have on both LP and CD and prefer to listen to from LP, either because the mastering is different or because I happen to like the vinyl sound in that music. But I can't tell the difference between the original vinyl and a recording of it, using a decent soundcard, burned back onto a CD again. I've tested it well enough to convince myself. All those attractive distortions are faithfully preserved. Chris

On Fri, Dec 28, 2012 at 05:31:31PM +0000, Folderol wrote: That can't be denied. But it's only part of the story. Keeping the signal in the analog domain is no guarantee that its quality will be preserved. Simple fact is that even switch and potentiometer contacts are not perfect - they *do* generate distortion as well. I can't count the number of times I've measured distortion in analog circuits that went away when switches or pots were either replaced or cleaned. And of course they can be replaced by solid state switches and VCAs, but these are not perfect either. 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)

On Fri, Dec 28, 2012 at 09:03:32PM -0500, Al Thompson wrote: The only 'justification' is that such maintenance can be extremely expensive and in many cases almost impossible or impractical, in the case of single PCB mixers for example. If you want regular maintainance, the extra cost of modular equipment pays off rather fast. Also a switch or pot doesn't have to be really defective, in many cases this is a sort of degradation that goes unnoticed for a long time. The point I wanted to make is that much supposedly superior analog equipment is real life less perfect that it is believed to be. There's nothing 'musical' to the type of distortion caused by a dirty switch. 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)

On 28 December 2012 17:31, Folderol &lt;folderol(a)ukfsn.org&gt; wrote: Well yes, the electrical signal from a mic or whatever is analogue, but I think the problem with this argument is that "later down the processing chain" is basically as soon as you try to do anything meaningful with the signal, such as (say) record it to tape. You might get back a high resolution signal, but it's not the same one you started with. If you're really bothered about accuracy rather than artistic effect or whatever, surely you want to digitise as early as possible. Chris

On Fri, December 28, 2012 4:47 pm, Jörn Nettingsmeier wrote: Len's rules for buying a stereo system: - spend your whole budget on speakers (listen to them, sometimes cheaper speakers even from the same manufacture sound better). - go to the good will store and buy whatever amp you can get for $30. - Be happy. -- Len Ovens www.OvenWerks.net

On 12/29/2012 09:54 PM, Florian Paul Schmidt wrote: true, i should have stated that explicitly. but quantisation noise is just that, noise. every analog system has noise, too, and yet people love it. the important thing is to make the noise signal-independent, because signal-dependent noise is very very nasty even at very low levels. this is achieved by proper dithering. you can even push the noise where it is least obvious. to summarize, there are two physical constraints, and they are common to analog and digital signals: bandwidth and dynamic range (or, more precisely, signal-to-noise ratio). in both respects, digital systems out-do analog ones easily, by orders of magnitude, if desired. so any preference for analog has nothing to do with inherent superiority or fidelity. either it's particular artefacts that are considered "euphonic", or it's a sense of conceptual simplicity and elegance, or, most often, the user's inability to understand digital systems combined with incomplete understanding of analog systems. -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

On Mon, Dec 31, 2012 at 12:54:07AM +1300, Chris Bannister wrote: No, both terms are ambiguous, as are many others ('power', 'impedance',... ) Usually the context will make it clear what exactly is meant when they are used. 'Dynamic range' can mean the range of levels a signal takes (which is more or less your definition if I understand it correctly), but the same term is also used as meaning 'the maximum available signal to noise ratio'. Also 'signal to noise ratio' can mean an actual value given some conditions, or the maximum value for a given system. So both terms could in some cases mean the same thing. It's always necesary to understand the context. For example, in microphone specs you could find a figure called 'signal to noise ratio'. In that case this is just 94 dB SPL minus the self noise, by definition. It's the S/N ratio you get when recording a signal of 94 dB SPL at the mic, assuming your preamp doesn't add noise. And it could be unweighted, or in dB(A), or be measured using the CCIR 468 method, and those values could be more than 10 dB different. 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)

On Mon, 2012-12-24 at 10:45 +0100, Florian Paul Schmidt wrote: Friends and I made tests with best studio equipment available, but we never made double blind tests, those tests were blind tests or they were completely not blind. It's possible to get the same results with analog and digital equipment and only to get the wanted colouration. For home studios analog has got the better colouration, regarding to tape saturation, you can't control neither digital nor analog colouration for a home studio. Tubes don't need to colourize the sound. Other than Neumann, the Brauner VM1 e.g. doesn't cut sound by path filters, the sound quality of the VM1 is caused by the capsule, that's why not all microphones do sound equal, some have colouration, others don't colourize the sound. If somebody ordered a stereo pair of the VM1, Mr. Brauner and I selected the microphones without doing blind tests, without measurements, just by singing and speaking into the microphone. If you have experiences in listening, this is easy to do, at least no customer ever was unsatisfied. For guitar amps there's colourisation, because the amps are build this way, e.g. by using Celestion speakers. That a lot of tube equipment does colour the sound that much is regarding to bad tubes and bad circuits. I guess there are several articles from Mr. Brauner in the VDT-Magazin, however, he also has his knowledge from books, such as Winfried Knobloch's "Röhrentechnik ganz modern", Pflaum Verlag München, ISBN 3-7905-0660-5. Unfortunately it was stopped to continue knowledge about tubes around 15 years ago by e.g. Elektor-Verlag, but they provided some good books about tube circuits. I guess they even stopped to sell Horowitz/Hill "The Art Of Electronics" ("Die Hohe Schule Der Elektronik"). It's often written that digital should be better regarding to the "neutral" sound quality, but I recognized that the knowledge about discrete analog circuits is gone. There are studios that still record analog, they don't use samples, but provide Melotrons and other gear. So using digital and than using samples that emulate Melotrons, Moogs etc., to add a saturation plug etc. never ever will sound as good as an analog production using a Meltron, Moogs etc. and a tape recorder. People often claim that digital should sound better, more neutral, but than they try to get all kinds of emulations. To "produce sound" we want colouration. It's possible to use professional analog gear, to get no colouration, but we want colouration. 2 Cents, Ralf

On Mon, Dec 24, 2012 at 10:13:01AM -0500, Ricardus Vincente wrote: And those of today, that hasn't changed. 44100 was chosen for a mix of reasons, the wikipedia article referred to earlier explains all of them very well. 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)

On Mon, Dec 24, 2012 at 10:13 AM, Ricardus Vincente &lt;wizardofgosz(a)gmail.com&gt; wrote: There was another more mundane reason than that. 44.1kHz allowed engineers to store an even number of samples in either an NTSC or a PAL scanline when using UMatic video recorders to record digital audio. Sony had supposedly even planned a line of products based on the idea, though they never came to market. The lowpass filters of the day weren't yet capable of cutting off cleanly in only 2kHz (well, 4kHz, it's ok if the transition band aliases) as they were all-analog and oversampling wasn't used yet, but designers correctly decided that technology would catch up. The CD had also originally been 14 bits because that was the biggest DAC that Phillips made at the time; Sony pushed hard for 16 bits [they had a 16 bit DAC in the works :-]. Several of the original CD players only played back 14 bits. Anyway, like most engineering, the tradeoffs were interrelated and all came into account. Monty

On 12/24/2012 06:01 PM, Neil wrote: Those were 12 bit, weren't they?

On Mon, Dec 24, 2012 at 07:34:26PM -0600, Neil wrote: I remember such a thing from when I worked in broadcasting, must have been late 80s. We even had a setup with two Bmax VCRs controlled by an Apple II that allowed to do simple assembly editing. It was an horrible thing to use. Edits would fail regularly. If you were lucky you could retry them, but quite often the already assembled part would be damaged at the edit point, forcing you to go back. It was used for a year or so, until something better (the first hard disk based systems) became available. 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)

On Sat, 22 Dec 2012 18:39:29 +0100, Ralf Mardorf wrote: Indeed they are analog circuits, so not as much conversion as Len's point. They're stereo so 12 inputs required if panning is to be preserved. 1.5 ADA 8000s etc. All good points, thank you. Two ADA8000s here so far. No complaints on the sound quality yet. True. :) but I now have about 24 mic. inputs and I expect I'll have many more before my home studio is finished. I guess I must pay for versatility I don't need. Better tailored products would perhaps be more expensive, as they would cater to a smaller market. -- John.

On Sat, 2012-12-22 at 13:00 -0800, Len Ovens wrote: When I was a friend of Dirk Brauner and I worked for his manufactory (we are from the same school, played in the same bands, before he falls out with our clique), we recommended the elCheapo ART tube devices. They are used by professional studios, in combination with the VM1. At least the old ART tube thingy, I guess it's from the 90s, isn't a toy, it's a very good pre amp.

On Sun, Dec 23, 2012 at 9:50 AM, John Murphy &lt;rosegardener(a)freeode.co.uk&gt;wrote;wrote: If it works for you have fun. I will say that I haven't been to impressed with what I have seen in as far as reviews of other SMPro equipment if it is the manufacturer I am thinking of, but I have not used any hands on to be certain. Yes, I worked with a keyboardist for some time that never knew what it was till I taught him what it was and how to use it. Not sure he ever used it. Seablade