At the Stereophile Show in NYC this weekend, Audio Alchemy was demonstrating their EDR*S remastering process. If you gave them a CD, they would process a track or two for you and give you the CD-R (for free) to try on your own system. If I am correct, EDR*S reads the data of a CD, uses eight DTI Pro-32's to increase the resolution (I believe to 24 bits; they split the audio band into seven segments and process each segment with a single Pro-32, and then use the eighth to sum), and then dithers down to 16 bits for transcription to a CD-R that can be read by any CD player. I invite Mark Schifter of Audio Alchemy to correct or elaborate on my description of the process, if necessary.
The results? Pretty amazing. I was expecting an improvement, but not an enhancement that was so *consonant with the music*. The soundstage wasn't just larger, but each instrument came to occupy more of its own space in the acoustic, so that there was a harmony of individual musicians playing together rather than just sounds coming from different places. Detail was improved, with instrumental decays sounding more natural (*not* just more evident), timbres more realistic with greater body (if appropriate), and greater dynamics. Clarity of line was greatly enhanced -- it was much easier to follow each instrument. Overall, quite a welcome surprise. And again, it wasn't just like adding a reverb sound effect or re-equalization or something like that. The improvements were uniformly realistic, appopriate and complementary to the music.
A couple oddities, though all relatively insignificant and not necessarily inherent to the process. First, on one track I had processed, the channels seem reversed. Don't know why this would happen, or why it only happened to one track. Second, there are a couple "blips" on the transcription, perhaps due to the double speed CD-R machine they were using. Third, my CD player sometimes has trouble reading the discs when they are first loaded (then again, my CD player -- Micromega Stage 3 -- *sucks* as far as disc toleration is concerned). Other than the first problem, I don't think these really have anything to do with the process itself.
A couple people I have spoken to have trouble with the concept of the process as far as information theory is concerned (if you only have 16 bits on the regular CD, where are the new bits coming from?). I'm not familiar enough with the theory behind this or the process itself to draw conclusions here. Regardless, I find the changes to be appropriate improvements, so it doesn't concern me terribly.
Also, I probably wouldn't get discs processed where I already know they are good recordings. For example, Gabe's PGM recordings, along with being expertly recorded using the best associated equipment, are originally 20 or 24 bit and then carefully noise-shaped to 16 bits, possibly in a similar manner to how EDR*S comes back to 16 bits after the resolution enhancement. It seems EDR*S in this case might be redundant here, and the 20 or 24 bit recordings in this case are true original information, so no problem with information theory. Still, the vast majority of recordings out there (especially older ones, which seem to particularly benefit from the process) are not audiophile quality.
So anyway, check this out! It's worth a listen. Contact Audio Alchemy to find out who to call for more information (I have the guy's card, but it's at home and I'm not).
There was a problem with the show equipment where the left and right channels could get swapped - this will of course be fixed.
Also, did you get your disk remastered on Wednesday morning? The CD-R was defective and many disks made on that day are defective. This is the second time that the CD-R has broken - they do not seem to be very reliable.
Finally, there have been reports in the press to the effect that disks made on CD-R's occasionally have compatibility problems with regular CD transports, although I have not come across this myself.
Your comments on the improvements due to EDR-S are very welcome, and in line with other feedback we have received. Thanks.
: that can be read by any CD player. I invite Mark Schifter of Audio : Alchemy to correct or elaborate on my description of the process, if : necessary.
I would also like to invite AA to do this.
You can't increase the resolution of a given recording.
You can copy it.
You can reduce its jitter (when it's jitter caused by irregularly spaced pits on the CD) by copying the data to a pre-wobbled CDR-disc.
You can do whatever you want to make the CD sound nicer.
But you can't bring back or 'invent' what simply isn't there anymore.
In article <4p89ma$...@biosun.harvard.edu>, Daniel Baker <ba...@cvrc.med.upenn.edu> wrote:
>A couple people I have spoken to have trouble with the concept of the >process as far as information theory is concerned (if you only have 16 >bits on the regular CD, where are the new bits coming from?). I'm not >familiar enough with the theory behind this or the process itself to >draw conclusions here. Regardless, I find the changes to be >appropriate improvements, so it doesn't concern me terribly.
It troubles me. But it is my job to be a skeptic until something is proven to me to be worthwile.
I will preface the following by saying that while I have seen the system and have spoken briefly to Keith Allsop (its creator), I have not yet heard the system and have been hesitant to do so until I have some questions answered.
Let us approach this question from the point of view of information theory. In the realm of information theory, we have the concept of a channel. A channel in this context is simply a pathway over which information of any kind can be sent. Every channel has a certain bandwidth, and while we needn't go into the whole question of what defines a channel's bandwidth here, let it suffice to say that the bandwidth regulates the amount of information that can be moved over that channel in a specific amount of time.
Let us take the case of a 20-bit recording that I have made using a 20-bit A/D and a Nagra-D. That recording contains a certain amount of information on it, and in fact the sheer quantity of information present is far greater than the CD's ability to carry. As a result, prior to my placing that information on the CD, I have to throw some of that information away. Now, I can use all manner of neat-o tricks to preserve as much of it as I can...to encode it into the bandwidth that I have. I can use noise shaping or spread-spectrum hidden bitstreams or any number of techniques. But in the end, I have unequivocally reduced my channel bandwidth from 20 to 16 bits. What I am left with is decidedly a 16-bit recording. I will never get those 4 bits back. Once they're gone, they're gone.
This is where my problem with the AA process comes in. No matter what kind of algorithm it is, there is no way that any process can look at those 16 bits of data and produce a different 16 bits of data that is somehow closer to my original 20 bits. Think of it this way. If I have a 5 decimal-place number, and I reduce it to 4 decimal places and throw the fifth away, there is no way that anyone can look at those four decimal places and deduce the fifth, and furthermore there is no way for anyone to look at those four places and suddenly provide a DIFFERENT four decimal-place number that is somehow "more correct."
No matter what you do, you cannot get a more accurate 16-bit number from another 16-bit number. To do so would violate entropy, and this was proven by Claude Shannon in his seminal paper on information theory, "The Mathematical Theory of Communication," Bell System Technical Journal, October 1948. Besides, it's common sense.
And thus I am left to wonder what the AA process is actually doing. I have not yet heard it run on one of my recordings, but I would be very, very surprised if the 16-bit processed disc sounded closer to the original master tape than my 16-bit original disc. Once again, without knowledge of what's on the master tape, there exists no information from which they can discern data that isn't there. You can't reverse entropy. There are a lot of DSP processes that one could run on 16-bit audio to produce another 16-bit recording, but as of this instant we have no reason to believe that the AA process is taking us closer to the original.
When I visited Marc Schifter and Keith Allsop at Hi-Fi '96, they were friendly but not wholly willing to reveal what it is that they're actually doing. This is understandable given the high-end market, but they have a ways to go if they want to gain the acceptance of their process as a legitimate method of processing. Notice how all the well-respected extant commercial processes (UV-22, SBM, Meridian, etc) are published documents which you can read about. Even those who want to be industrious can go read the HDCD patent application. Keith, how about a paper at the next AES convention?
The algorithms need to be explained before I'll be convinced that it's something I want done to my audio.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
In article <4pufs9$...@decaxp.HARVARD.EDU>, "Gabe M. Wiener"
<g...@pgm.com> wrote: >... >This is where my problem with the AA process comes in. No matter what >kind of algorithm it is, there is no way that any process can look at >those 16 bits of data and produce a different 16 bits of data that is >somehow closer to my original 20 bits....
They don't claim to be creating 16 bits that are closer to your original 20 bits. They are claiming to create 18 to 20 bits that may be closer to your 20 bits than the a CD's 16 bits are. To turn their 20 bits back into 16 requires the same tricks that you used in the first place to reduce 20 to 16.
My experience is that on some material (mostly older CDs) it is more effective than other (mostly recent, real high quality CDs.) For sure it IS signal processing and as such, it does make things sound different, but then, that's the whole point. What amazes me is how I'm generally not inclined to bypass it on newer recordings as I would expect to be logically.
-- Bob Olhsson Audio | O tongue, thou art a treasure without end. Box 555,Novato CA 94948 | And, O tongue, thou art also a disease 415.457.2620 | without remedy. == Jelal'uddin Rumi == 415.456.1496 FAX |
ogi...@imec.be (Werner Ogiers) writes: >You can't increase the resolution of a given recording.
This is not strictly true. If you know the signal follows a certain set of rules, you can theoretically reconstruct it. A bunch of people in the 70s did this for pictures, and you can also ask some math friends about analytic continuation.
ogi...@imec.be (Werner Ogiers) writes: >You can't increase the resolution of a given recording.
But music is not random bits. The same is true of images. This is why image compression can work. (see for instance Fractals Everywhere, by Michael Barnsley , by the way I'm thanked in the preface!)
In article <4q69bj$...@tolstoy.lerc.nasa.gov>, Andre T. Yew <and...@alumnae.caltech.edu> wrote:
> This is not strictly true. If you know the signal follows a >certain set of rules, you can theoretically reconstruct it. A bunch >of people in the 70s did this for pictures, and you can also ask some >math friends about analytic continuation.
I still maintain that better results will be obtained if you start with a higher-resolution master and use an intelligent approach to reduce it to 16 bits, rather than using an algorithm (no matter how good) to go from 16 bits to 16 bits.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
tommor...@aol.com (TomMorley) writes: > ogi...@imec.be (Werner Ogiers) writes: >> You can't increase the resolution of a given recording. > But music is not random bits. The same is true of images. This is > why image compression can work.
Would you care to elaborate on this? This statement, in and of itself, doesn't seem to say a heck of lot. Just because music isn't random, doesn't imply that you know enough about the underlying continuous signal, given a limited resolution approximation, to reconstruct it at some higher resolution.
[[ James W. Durkin -- j...@graphics.cornell.edu ]] [[ Program of Computer Graphics -- Cornell University ]]
TomMorley (tommor...@aol.com) wrote: > ogi...@imec.be (Werner Ogiers) writes: > >You can't increase the resolution of a given recording. > But music is not random bits. The same is true of images. This is why > image compression can work. (see for instance Fractals Everywhere, by > Michael Barnsley , by the way I'm thanked in the preface!)
A better original statement would've been that you can't increase the INFORMATION CONTENT of a given recording. 16 bits is 16 bits is 16 bits. You can use the available bandwidth more efficiently, but you can never pack more than a certain amount of information in to a given channel. Image compression, or for that matter ANY compression scheme, works either by removing redundant information (which can be a lossless process), or by throwing away information that isn't redundant but which we don't think we care much about (which by definition is a lossy process). But in no case can you get (a) more information out of the process than was present in the original, or (b) somehow get around the theoretical limits of the channel capacity.
To get back to the original point, there is simply no way you can pack 20 bits of honest-to-good useful information into 16-bit samples without doing some form of compression which results in an average LOSS of information. The result may or may not be better than what you'd hear having gone straight from the analog signal to 16 bits in the first place. I suppose the one advantage I can see in trying to pack 20 into 16 is that in effect, you would be doing some dithering which could be controlled a little better than otherwise.
Bob Myers KC0EW Hewlett-Packard Co. |Opinions expressed here are not Workstations Systems Div.|those of my employer or any other my...@fc.hp.com Fort Collins, Colorado |sentient life-form on this planet.
James Durkin <j...@graphics.cornell.edu> writes: >tommor...@aol.com (TomMorley) writes: >> ogi...@imec.be (Werner Ogiers) writes: >>> You can't increase the resolution of a given recording. >> But music is not random bits. The same is true of images. This is >> why image compression can work. >Would you care to elaborate on this? This statement, in and of >itself, doesn't seem to say a heck of lot. Just because music isn't >random, doesn't imply that you know enough about the underlying >continuous signal, given a limited resolution approximation, to >reconstruct it at some higher resolution.
I believe that he is talking about Fractal compression. From how it has been explained to me, it takes the random data, and then tries to find a fractal equation that closely APPROXIMATES the data. It only compresses part of the picture in an given equation, of course, but then the stored equation is much smaller then the data it represents. When compressing the data,you specify the target compression ratio, and the how much time the compression program should spend finding the best fractal equations. It supposedly works very well if you give it a lot of time, but as the more compressed and the less time you spend,the more noticable artifacts from the compression become. You can actually perceptually increase the quality of a bad picture, and enlarge it a bit without a problem.
The problem with this is that it is ONLY an APPROXIMATION of the data. Another problem is that it is very proprietary,and the inventors, seeing a good thing, charge large licensing fees.
In article <4q9qv9$...@agate.berkeley.edu>, James Durkin
<j...@graphics.cornell.edu> wrote: >tommor...@aol.com (TomMorley) writes: >> ogi...@imec.be (Werner Ogiers) writes: >>> You can't increase the resolution of a given recording. >> But music is not random bits. The same is true of images. This is >> why image compression can work. > Would you care to elaborate on this? This statement, in and of > itself, doesn't seem to say a heck of lot. Just because music isn't > random, doesn't imply that you know enough about the underlying > continuous signal, given a limited resolution approximation, to > reconstruct it at some higher resolution.
I would think that truncation might well follow enough of a pattern to be at least somewhat "patchable."
It's still really got to be a crap-shoot whether the result is more or less accurate, however, the effect IS pleasing, and a lot more like that of having used a 20 bit converter instead of a 16, at least to my ears.
Redithering is a new artform that certainly has promise at least for older material.
We ought not to get overly hung up on accuracy. Certainly it is a useful tool and it IS very important to remain conscious of the difference between accuracy and an inaccurate but pleasing enhancement effect.
One must always remember that pleasing inaccurate enhancements are the name of the commercial recording game. Nobody ever sits in the typical mike's catbird seat yet no one would consider requiring that mikes only be used between 3 and 6 feet off the ground in an audience seat. Likewise some of the most famous classical recording venues in the world are routinely "sweetened" with artificial reverb. Nobody pays much attention because the recordings made there WORK for their audience which is the only truly valid criteria. When an effect distracts because of heavy-handed use is when it becomes objectionable.
-- Bob Olhsson Audio | O tongue, thou art a treasure without end. Box 555,Novato CA 94948 | And, O tongue, thou art also a disease 415.457.2620 | without remedy. == Jelal'uddin Rumi == 415.456.1496 FAX |
In <4q69bj$...@tolstoy.lerc.nasa.gov>, and...@alumnae.caltech.edu (Andre T. Yew) writes:
>ogi...@imec.be (Werner Ogiers) writes: >>You can't increase the resolution of a given recording. > This is not strictly true. If you know the signal follows a >certain set of rules, you can theoretically reconstruct it. A bunch >of people in the 70s did this for pictures, and you can also ask some >math friends about analytic continuation.
1. Would you care to give us one example from that "set of rules"?
2. Let us assume that we know that set of rules. How can this knowledge be used to convert one 16-bit stream into another 16-bit stream that is closer (in some sense) to the original signal even without knowing how the original signal was transformed into a 16-bit signal?
3. I know what people do with pictures (image estimation / reconstruction / processing) and it bears no resemblance to what is being discussed here.
>>You can't increase the resolution of a given recording. >1. Would you care to give us one example from that "set > of rules"?
No I, but surely bits 17 through 20 as "reconstucted" are not compleatly arbitrary. You may have lost Forever the 2dn violinist in the third row, fourth over from the right shuffing his feet, but the ambience from the flute that just dropped below bit 16 can (in principle) be guessed at.
>2. Let us assume that we know that set of rules. How can >this knowledge be used to convert one 16-bit stream into >another 16-bit stream that is closer (in some sense) to >the original signal even without knowing how the original >signal was transformed into a >16-bit signal?
I agree what can 16 to 16 do? One possible answer is that this would take advantage of know properties of D to A converters. This smacks of the various failures in the vinyl? (Help be out here RCA?? late 60's???)
>3. I know what people do with pictures (image >estimation / reconstruction / processing) and it bears >no resemblance to what is being discussed here.
But 16 to 20 bit might be analogous to image restoration or enhancement.
General comment: A great deal of the CD's on the shelf next to this computer are acoustic blues from the 20's and 30's. The CD are taken from old rare 78's often in wretched condition. Noetheless, dispite the above comments, I find most current attempts at retoration (such as CEDAR) unsatisfying and unmusical, especially when used with a heavy hand. I usually prefer (at this point in technology) a straight 78 to CD transfer, finding the best of the (say) 4 known copies of the 78.
There is something vaguely perverse about listning to Charlie Patton's High Water Everywhere (part II) through state of the art (inso far as my budget will permit) equiptment.
TomMorley <tommor...@aol.com> wrote: >General comment: A great deal of the CD's on the shelf next to this >computer are acoustic blues from the 20's and 30's. The CD are taken >from old rare 78's often in wretched condition. Noetheless, dispite >the above comments, I find most current attempts at retoration (such >as CEDAR) unsatisfying and unmusical, especially when used with a >heavy hand. I usually prefer (at this point in technology) a straight >78 to CD transfer, finding the best of the (say) 4 known copies of the >78.
I too have been seriously bitten by the 78rpm bug, and more recently old, old, old R&B.
My introudction into the world of tube amps came at the beginning of my 78rpm collecting kick (about 10 years ago). I had dug out my dad's old early 1950's Bogen mono-integrated-tube-amp for use in playing back 78rpm disks. This amp had adjustable equalization for both treble rolloff and bass turnover frequencies.
A correct playback equalization is essential when playing archival material. Even Columbia "LP" pressings from the pre-RIAA days benefit greatly.
In any case, experience with the lush sound of a classic 6L6 push-pull tube amp got me going on the high-end kick. Home-brew 12AX7A pre-amp and slightly modified Dyna ST-70 later; the Magnepans are happy as are the 78s.
Speaking of 78->CD transfers, have you actually spun 78s? Charley Patton is rare stuff.... but the sound of even a worn 78 is alluring. I've never heard a remastering that fully captures the "sound" of a 78 (neither cassette nor 7.5ips open reel does justice). A 78 seems to have a very dynamic sound, despite the poor s/n ratio (though clean 78s are nearly as quiet as an LP).
The simple, uncompressed microphone feed used on many 78rpm era recordings also contributes to the unmistakable liveness.
Some of the most "78"-like remasterings I've heard are those on the Russian Melodiya label. Alas, they did not reissue much in the way of Blues legends 8^).
>There is something vaguely perverse about listning to Charlie Patton's >High Water Everywhere (part II) through state of the art (inso far as >my budget will permit) equiptment.
Bob Olhsson <o...@hyperback.com> wrote: >They don't claim to be creating 16 bits that are closer to your >original 20 bits. They are claiming to create 18 to 20 bits that may >be closer to your 20 bits than the a CD's 16 bits are.
And they have not yet explained to us how this is possible. It flies in the face of the most basic concepts of entropy.
>To turn their >20 bits back into 16 requires the same tricks that you used in the >first place to reduce 20 to 16.
And the net result is that they are taking a 16-bit dataset and attempting to supersede it with another 16-bit dataset. This strikes me as a little bit of a stretch, to put it euphemistically. If the AA guys would like to give a cogent technical explanation of what's going on, I'm all ears.
One can go out and read technical papers (AES or otherwise) on nearly every commercial encoding process around.....be it UV-22, SBM, Meridian, what have you.
When manufacturers refuse to explain what's up, that's when I begin to get concerned.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
In article <4ppf6n$...@eyrie.graphics.cornell.edu>,
Werner Ogiers <ogi...@imec.be> wrote: >You can reduce its jitter (when it's jitter caused by irregularly >spaced pits on the CD) by copying the data to a pre-wobbled CDR-disc.
Let's be correct in our terminology here. You can re-make its pit geometry by copying it to a CD-R. Whether this re-making results in less jitter on decoding...and whether this has any effect on sound quality...is entirely a function of what hardware you use to play it back. Geometrical differences are not ipso facto translated into jitter, but evidence suggests that they can be under certain circumstances.
>But you can't bring back or 'invent' what simply isn't there anymore.
Well, one could "invent" whatever one wants. But in no way can you recover that which has been discarded. You can use whatever heuristics you wish in order to synthesize or interpolate data, but that does not in any way suggest that this data resembles the original.
Let us not forget the most fundamental rule of entropy. Once you have lost information entropically (i.e. by collapsing it), you cannot get it back.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
>When manufacturers refuse to explain what's up, that's when I begin to >get concerned.
Gabe...
Our company holds some 14 patents (via Peter Madnick and others)... and YOU of all people should be "smart enough" to realize that once these sorts of things are published they become "road maps" for others to follow (read steal)...
I'm going to ask YOU to sign a non-disclosure document and then ask Keith to visit you and make some re-mastered copies (and explain the process) for you. I'll be curious as all get up to hear about your reaction.
We've made several hundred re-mastered copies for truth seekers like yourself (self proclaimed or otherwise)... and EVERYONE we've made copies for (of those that have reported) have been startled by the outcome. This IS a fact. So it's probably time to explain things a bit further to one such as yourself.
Mark Schifter <LWBF...@prodigy.com> wrote: >Our company holds some 14 patents (via Peter Madnick and others)... >and YOU of all people should be "smart enough" to realize that once >these sorts of things are published they become "road maps" for others >to follow (read steal)...
I am well aware of the downside of patenting one's technology. But I am also aware of the fact that without a frank and open dialogue with the audio engineering community, it is a difficult thing for those of us who use the technology to consider its use for professional purposes.
>I'm going to ask YOU to sign a non-disclosure document and then ask >Keith to visit you and make some re-mastered copies (and explain the >process) for you. I'll be curious as all get up to hear about your >reaction.
If I am presented with a non-disclosure document, I will sign it and will not discuss the nature of the algorithms or the functioning of the technology itself.
But as I'm sure you can appreciate, what I will do is report quite honestly and fairly what my subjective findings were, after evaluating the technology and the effect that it has on my software.
I consider myself one of the fairest testers I know. I have no loyalty to anyone except he who makes the best products. Every so often I come across a product line (like Prism, for instance) who puts out consistently superior equipment. But even there, I always say to Graham (director of Prism), "Be careful when you ask me to run a shootout of your converter against someone else's, because you have to be prepared for the possibility that you might lose."
Of course, in 2+ years, they haven't yet, but my point is that if I feel a technology is good, I will say so, and if I feel it isn't, I'll say so too.
I await the non-disclosure document. After I sign it and return it, please ask Keith to call me at work to make an appointment to visit.
And, folks, there we leave this issue until after Keith's visit.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
In article <4qcjkb$...@eyrie.graphics.cornell.edu>, Bob Olhsson
<o...@hyperback.com> wrote: > It's still really got to be a crap-shoot whether the result is more > or less accurate, however, the effect IS pleasing, and a lot more > like that of having used a 20 bit converter instead of a 16, at > least to my ears.
If it's a crap-shoot, then I can assure you that the odds are weighted toward the less accurate.
I have many DSP units that take in 16-bit inputs, do a little processing (whatever algorithm might be running) and output 20 or 24 bits. But that does not mean that I have any more information about my original recording. It means that the DSP algorithm just so happens to have a higher precision output than my original signal!
> We ought not to get overly hung up on accuracy. Certainly it is a > useful tool and it IS very important to remain conscious of the > difference between accuracy and an inaccurate but pleasing > enhancement effect.
No, we ought to get exceedingly hung up on accuracy. The goal of audio equipment...at least in my work...is reproducing the musical experience, not editorializing on it. If the performance is warm and voluptuous, or cold and stark, I want that rendered on the CD.
> One must always remember that pleasing inaccurate enhancements are > the name of the commercial recording game.
Au contraire. Many (myself included) strive to add as little coloration to the recording as possible, and to let the music speak for itself.
> Nobody ever sits in the typical mike's catbird seat yet no one would > consider requiring that mikes only be used between 3 and 6 feet off > the ground in an audience seat.
Well, excepting the fact that the binaural folks do this all the time, let us remember that we place microphones the way we do in order to compensate for the fact that the recording still has to be reproduced through loudspeakers, which means that the sound gets radiated twice. Once from performer to microphone, and once from loudspeaker to ear. Contrast this with the single radiation that occurs for a live performance. If everyone recorded binaurally, placing the mics six feet up in an audience seat would be precisely what we'd do.
> Likewise some of the most famous classical recording venues in the > world are routinely "sweetened" with artificial reverb.
I for one would never be caught dead in a venue that needed reverb unless someone put a gun to my head.
> Nobody pays much attention because the recordings made there WORK > for their audience which is the only truly valid criteria.
Do they? Most such recordings, in my experience, sound appalling, and nothing like real music in a real hall. Artificial reverberation and excessive processing may be fine for pop music, but as far as I'm concerned it has no place in classical recording, or any recording genre where the goal is not to create sounds but to re-create a live musical event.
-- Gabe Wiener Dir., PGM Early Music Recordings |"I am terrified at the thought A Div. of Quintessential Sound, Inc., New York | that so much hideous and bad Recording-Mastering-Restoration (212) 586-4200 | music may be put on records g...@pgm.com http://www.pgm.com | forever."--Sir Arthur Sullivan
Gabe M. Wiener wrote: > And the net result is that they are taking a 16-bit dataset and > attempting to supersede it with another 16-bit dataset. This strikes > me as a little bit of a stretch, to put it euphemistically. If the AA > guys would like to give a cogent technical explanation of what's going > on, I'm all ears.
Gabe:
Why don't you, one of the most refreshingly open-minded engineers I've had the pleasure of meeting, just go listen to the damn thing first. At least go listen to their DDTIPRO32. Then put it on the test bench - as I'm sure you could measure what it does :)
> One can go out and read technical papers (AES or otherwise) on nearly > every commercial encoding process around.....be it UV-22, SBM, > Meridian, what have you. When manufacturers refuse to explain what's > up, that's when I begin to get concerned.
In article <4qjoeb...@decaxp.HARVARD.EDU>, "Gabe M. Wiener" <g...@pgm.com>
wrote: > When manufacturers refuse to explain what's up, that's when I begin > to get concerned.
They have explained that they are interpolating more bits somehow. They only are not talking about what kind of algorhythm they are doing it with.
I think AA is too small a company to be able to afford the several years of secret development and non-disclosure agreements that preceeded most of the AES papers and explainations you are referring to.
The Apogee UV-22 and Meridian noise-shaping technology were both derived from previous academic research. Neither company has told anybody HOW they do what they do, only that they are accomplishing what the academic research suggested might be do-able ten years ago. There hasn't been a comparable public feasability discussion preceeding the AA work however I can't see blaming that situation on Audio Alchemy.
It would be interesting to compare a 20 bit recording with an interpolated 20 bit result from the same recording reduced to 16 bits. Hopefully I can get some time to try it.
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In article <4q9qv9$...@agate.berkeley.edu> James Durkin
<j...@graphics.cornell.edu> writes: > Would you care to elaborate on this? This statement, in and of > itself, doesn't seem to say a heck of lot. Just because music isn't > random, doesn't imply that you know enough about the underlying > continuous signal, given a limited resolution approximation, to > reconstruct it at some higher resolution.
You can't do that. Once you've added noise, you're stuck with it, unless you have other knowledge of the exact signal structure.
But you can reduce the bit rate, by using source-coding techniques (source modelling, rate-distortion coding, noiseless coding) to reduce the bit rate while maintaining a 1:1 input to output.
It is conceivable that one could encode on a CD whatever resolution (it would be time-varying, but not worse than 16 bits) could be represented in 16 bits, by using some sort of very flexible source coder.
There are, of course, better ways, one might also run a very-high-rate perceptual coder, use that knowledge of perception that the high-end proponents in this group adamantly reject, and get perhaps 20 bits of input 'resolution' where it matters. (one could get more by compression, but getting meaningful bits to the input might be a bit tough)
A look in Akansu and Smith "Subband and Wavelet Transforms, Design and Applications", Chapter 9, might unconfute some of the differences between source coding gain, perceptual coding, and "what works".
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In article <4q9r1k$...@agate.berkeley.edu> my...@hpfcla.fc.hp.com (Bob
Myers) writes: > To get back to the original point, there is simply no way you can pack > 20 bits of honest-to-good useful information into 16-bit samples > without doing some form of compression which results in an average > LOSS of information.
Whoops. Gotta pick at a nit here, sorry.
If you have a full 20 bits of real, non-redundant information, you're right.
But that's not what most audio signals look like. Even the least-redundant audio signal I've ever measured (gets out Akansu and Smith again), has about -30 dB give or take, spectral flatness measure, indicating that a purely rate-distortion coding method can remove about 5 bits on average from that signal, WITHOUT CHANGING THE VALUES OF THE DECODED SIGNAL.
Now, these two statements are not contradictory, what that -30dB spectral flatness measure shows is that the signal, while it might have 16 or 20 bits input, has redundancy (i.e. each sample can be predicted from the past samples) that can be extracted via well-known methods and utilized for coding gain. (That gain at a 64 sample interval, to be clear.)
In other words, the amount of "information" in a 20 bit signal is at MOST 20 bits/sample. It's possible to have much much less.
> The result may or may not be better than what you'd hear having gone > straight from the analog signal to 16 bits in the first place.
Well, if the signal is typical of audio signals, it's likely to be possible to do a 1:1 20 bit encode into 16 bits for nearly all samples of the signal, and very likely all samples given the averages observed on most audio signals. (A signal that could not be compressed in this way would be white noise. I'm not sure we care all that much, but I must be clear, compressing "white noise" is an oxymoron.)
> I suppose the one advantage I can see in trying to pack 20 into 16 > is that in effect, you would be doing some dithering which could be > controlled a little better than otherwise.
For most, if not all audio signals, you could probably pack the full information in the 20 bit signal into 16 bits, in a real process, that restores the same numbers at the output. While it's certainly possible to make a signal for which this wouldn't work, it's not very common in "real music" signals.
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qaq...@biostat.sph.unc.edu writes: >1. Would you care to give us one example from that "set of rules"?
How about three? A signal that is bandwidth-limited, and point- sampled can be completely reconstructed by a sinc function. Both signals still have 16-bit resolution, but the reconstructed signal comes closer to the original signal than the point-sampled version, answering your question 2.
Knowing the inital trajectory and velocity of Voyager, and the positions and velocity of the various planets in our solar system, and applying a set of rules known as "Newtonian physics", we get to derive the complete trajectory at every point of Voyager.
Knowing the starting point and algorithm of a pseudo-random number generator, you can retrieve a message hidden deep in otherwise innocuous noise sent to you by a friend encoded with the same PRN generator. For examples, spread-spectrum communications or encryption.
Have you asked a math friend what analytic continuation is? If so, you get a fourth example for free!
>3. I know what people do with pictures (image estimation / >reconstruction / processing) and it bears no resemblance to what is >being discussed here.
No, it doesn't. However, if you reread my original post carefully, you will see that that wasn't what I was trying to say. To summarize for you, someone said that you can't end up with more bits than you started with, in effect, no more information than before. I said that this is not strictly true, if you assume that the signal follows a certain set of rules. Conservation of information is still going on here, since you have brought in extra information for the reconstruction by "knowing" that the signal followed a certain model. Whether that assumption is valid or not is the interesting part, no? And shouldn't that be the real question that we should be asking Audio Alchemy?
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