Do you hear the difference between a 128 and 320 kbps mp3? (test yourself inside!) - Page 2

Its very real, analog sounds different. Whether its 'better' is subjective dont you agree? Im perfectly fine with people saying they prefer the sound of analog, because what they prefer is up to them. However their reasoning is rubblish.

They prefer the sound of analog because of the way the circuitry mangles the sound!!!!


well of course its much more complex than I said, but im not going to write 20 pages....However the fundamental principles are sound. (sic) hehe. Every instrument bar a sinewave generator has harmonics and overtones, thats what gives a sound its timbre, but you already know this being a muzak teacher....


well, getting technical, its because a piano is not tuned mathmatically perfectly that makes it a pain to sample, the resonances are close, but the effect of imperfect intervals is that it creates a chorus effect, ie periodic modulated slight detuning.

ok, im just trying to explain things simply....vibration of strings, resonances, how they work, forgetting the specific mechanics of a piano, this is how it works - and as far as were concerned - trying to establish if ultrasound effects audio, then the piano string model is easy to visualize....

However - I am perfectly right when I say that you cannot accurately capture the subtle resonances of a piano when sampling it a note at a time. Here's why

If I strike a single note - I set the string vibrating, and on its own it contains its characteristic set of harmonics that define it as a piano - However part of the characteristic of a piano is the resonance that occurs when a string is struck that sets other strings vibrating, regardless of whether they are mathmatically in tune....harmonics tend to wander from mathmatical perfection as you hit higher order ones anyway.

Now say I sample a single key - lets "symbolically" call this A, and suppose strings B, C, D, E resonate with it. I record this

A---BCDE

Now I sample another single key lets call this A1---and strings B1,C1,D1 and E1 resonate with it.

Now in my sampler, I play A and A1 together - and as i've captured the resonance of these strings, Im also reproducing resonances B,B1,C,C1,D,D1 and E and E1 as you would expect.

A perfect capture of the essence of a piano??? Not quite.....

In a real piano, the soundwaves of A and A1 would modulate eachother - something like the effect of beating on closely matched tones, and another frequency and harmonic set is created, A2---B2,C2,D2,E2 which in turn causes other resonances in strings, and thus we have in effect a characteristic of the piano that has not been captured by our sampler when we sampled every key...because playing back the 2 samples simultaneously does not recreate A2,B2,C2,D2,E2.

Lets forget the effect on timbre and harmonics by hitting the keys harder or softer...

Now, not much music is simply 2 keys of a piano, so imagine the very complex interplay between numerous notes being played and the resonances created in the physical world..its just not possible to sample it one key at a time.

However, like the ultrasound effect we are trying to solve, the effect of this is so small and virtually indistinguishable, that you can sample every key of a piano and fool 99.9% of the listeners...if you are very clever and know how to process the sound to get around the pitfalls of sampling a piano. Its really not as easy as sampling every key.



well it cant hurt to have them present can it?



The 'something' that is being lost is actually - what is being added to the analog recording that makes it perceptibly sound better.

The real problem with digital is that it suffers from being too accurate and the slight distortions it does introduce are not pleasant sounding to the ear. In contrast analog is not very faithful to the source, but includes by default distortions that are pleasant to the ear.

Really, if you dont believe me that distortions make things sound better, get an SPL vitalizer or suchlike, its a psychoacoustic processor, It distorts the crap out of sounds and they take on all the properties that people are claiming make analog sound better. Better stereo seperation, high frequency distortion to add in near ultrasonic harmonics...etc. It really isn't accurate to source in any sense of the word, but it works.

my research effort to find evidence to back me up has produced this...


see....


http://www.postpiano.com/products/K2.htm

well atleast the problem is being addressed!

Yes.


Depends on with whom you're speaking I guess.


Again, depends on who you talk to.


<head explodes>

No really, I guess that makes sense. But aren't you really just confirming the point about digital not being as accurate, especially for certain types of music? After all, one of the most common hi-fi listener comments is that string instruments sound better with analog. Hmmm.



Exactly...and digital is not recording them.


It's not being "added"--that's the point. It's simply there, whereas with digital it must be added or left out.


I agree with part one...not necessarily with part two. It depends on the quality of analog recording.


I do believe you. I just thing distortion is not the only factor.

well I used the piano example in the context of recording to make a sampled instrument that can be played in a synthesizer, if you actually record a real piano performance to digital you capture everything that enters the microphone.

I chose the piano example for a reason...It shows that even in the audio range, with closely spaced harmonics that essentially cross modulation resonance is irrelavent - you backed this up when you claimed a piano can be sampled 1 key at a time. What is masked, ie the missing information is 99.9% irrelavent. And this is in the audio range.

Now if were talking about ultrasonics having the same effect on audio - the harmonic spacing is much greater and the energies much diminished - so the effect is going to be magnitudes smaller - to the point that I accept that it does not matter and it is not this effect that gives analog its warmth.




I had a quick look up the technical specs of vinyl and decks. It would seem that a perfect vinyl cut on the best equipment available can record frequency up to 75khz. There is a problem though, because of the mechanical contact between the grooves in the vinyl and the stylus, after a single playing of the record, your stylus has 'cut' away any frequencies in the ultrasonic. Typically after a few playings, vinyl has a frequency cutoff about 18khz and subsequently deteriorates.

If you use a bit of common sense guesstimation, knowing that the stylus has mass and the laws of physics, you can see that after the grooves in the vinyl have become a bit slack, there will be distortions created by tracking errors of the stylus. Knowing that momentum is subject to acceleration and decelleration forces, you can visualize that tracking errors actually produce rounded square wave clipping distortions - now a bit of synthesizer theory will tell that rounded square waves are produced by a square wave passed through a low pass filter and that a square wave produces lots of harmonic distortion in the odd periods, while the rounding produces even harmonics and attenuates the levels of the odd harmonics.

In essence this creates a very fat warm sound - thats analog synthesizers were hot property for dance music producers in the 90's. Interestingly though with the advent of powerful modelling algorithms on computers, recreating the analog synthesizer sound digitally is actually in some cases better, ie fatter and warmer than the analog counterpart!

I think that if there are ultrasonic sounds present in the replay of a vinyl, that they are created by harmonic distortions of high frequency sounds already present - and thus they are artificial, and not part of the original sound. I've mentioned the SPL vitalizer - this doesn't work that way, but there is something called the Aphex Aural Exciter which works in exactly that way - it adds sparkle warmth and depth specifically by distorting the high frequencies and adding the harmonics back in to the original sound. Completely artificial, but again sounds great.

Incidently, I looked up what instruments could produce ultrasonic sounds naturally, most of them do, but the % of energy they create above 20khz is less than 2% in all cases and most cases is less than 0.5%. There is one instrument that creates 40% of its sound above 20khz....Its the cymbal.

So if you were looking for evidence of ultrasonic deterioration in digital music, your best chance of finding it is to look at the digital recording of cymbals. There is a snag however...as there always is....cymbals are enharmonic!

Which means that essentially a cymbal creates not much but high frequency tuned noise, and even if ultrasonics resonated the noise, because the human ear does not distinguish timbre in high frequencies, you will still only detect the presence of energy - which is already there anyway! - ie its impossible for the human ear to resolve the effect of resonance on a cymbal, you'd only detect it using spectral analysis on a computer!!



but it is being added. It comes from the lack of resolution in the stylus, in the groves of the vinyl, in the signal path. It just is not there in the original source instruments. Its 'simply there' because vinyl is not an accurate and fortunately the type of distortions it makes to the source sound pleasing. With digital, its just far more accurate to the source, and if you want the nice-sounding 'gloss' you have to add it in.


I think by the time you've added up all the factors of mechanical and electrical distortions of the vinyl, from tracking errors in the stylus, to the magnetic hysterisis and saturation in the moving magnet, to the distortions of components in the signal path - I think you can explain vinyl's warmth.

Incidently, digital audio is a pretty flat frequency response media. Vinyl recordings are actually passed through a device that emphasises the high frequency and attenuates the low frequency (like dolby noise reduction on cassettes) - on playback the deck contains circuitry that reverses this - it boosts LF and attenuates HF.

Now seeing as the circuitry for this is going to be some kind of multistage broad range equalizer - and as such equalizers by default introduce phase distortions - (this is how the spl vitalizer works) its not too hard to imagine that if you buy an expensive deck, that the EQ circuitry tends to lean towards the phychoacousic type of equalizer rather than the less expensive passive band pass type. You might actually find that expensive decks use some kind of psychoacoustic processing on them to give them the kind of warmth and stereo seperation that you like....hmmm... completely artificial, but nice!

Great. <tapes head>


Source? It certainly sounds reasonable.


I understand...but yes, it's guesstimation. I think you're potentially overestimating all of those errors in terms of physics and mass. Also, the best vinyl equipment compensates for some that, with laser-guided tracking and various other means.


But those frequencies are present, and perhaps enough so to make a difference. Also, I don't think you're using "enharmonic" correctly, unless you mean that cymbals sound equal to one another in terms of pitch.


But that "noise" will be present in some recordings and not others. It's not clear what effect the energies have on our auditory perceptions.


OK, that's a good point. Again though, you've mentioned that the frequency response of vinyl is far higher than digital, and I'm just saying that might have something to do with the perception of warmth, space, etc. On the other hand, most speakers can't produce those frequencies at all, so


Or a good portion of it, I would agree. The question is whether that can explain all of it.


Imagine how nice it sounds with vacuum tube amps!

no, i think im right!

http://www.soundonsound.com/sos/aug0...ecrets0802.asp




I have gone to pretty great lengths to explain why it is not an issue, perhaps you could provide some evidence to state why it is an issue. All I'm hearing at the moment is regurgitation of quarter-educated audiobuffs imaginations!


The thing is, in a continuous passage of music, the brain cannot detect changes in volume until it hits about 3db, if ultrasound were creating that much resonance in the audio spectrum, it would be pretty obvious all of the time.


I can imagine! However do you have any idea of the amount of distortion created when using tubes? They use these things specifically because they have non-linear response curves, and as soon as your talking non-linear - youre talking total harmonic distortion.

I wonder if you're getting a bit hung up over the use of the word distortion? Maybe lets use a different word, as distortion has many negative connotations attached, usually it is something to avoid. But used properly 'controlled wave shaping' is the tool engineers use on everything to add life, warmth, depth, sparkle, clarity, etc.

It exists everywhere in the recording chain, companies reputation hinge of the back of the way their products shape the waves.

http://en.wikipedia.org/wiki/Gramoph...onse_and_noise

I really don't get the context it's being used in. The only definition I'm aware of is the one I've used in my studies (e.g A# is the same pitch as Bb).
Looking up the word gives on that definition.


Actually, you've speculated, though quite plausibly and reasonably. In any case, I'm not the one who brought up these "energies" to begin with.


Detecting changes in volume is not the same as the "inaudible" portions being perceived, even if it's only the vibrations being picked up unconsciously.


No...I was making a tongue-in-cheek reference. Obviously tubes create distortion. I was really just sort of mocking the quarter-educated folks you referenced!

But its still distortion... ie a deviation from the original waveform!
Engineers in the studio use all manner of devices to alter ie distort the waveform of a signal, to correct a sound, and/or enhance it, in contrast to 'high fidelity'... the most accurate possible reproduction of pre-recorded music. In the studio of course, there is a "high fidelity" section of the signal path .. ie the monitoring system post mixing board. Everything before that is subject to distortion of one kind or another... even the signal path within the most expensive consoles will generate some degree of distortion, especially analog boards, and when that distortion is multiplied in the multitrack environment (120+ tracks and channels of discrete mixing).... Some of the most expensive and desirable "outboard" equipment in the studio control room are vintage era microphone preamps, limiters, compressers,and eq networks, which on paper (scope), have quite "dirty" signal paths and by rights, should sound none too great.. some being vacuum tube based and others early transistor circuit designs. Not to forget, the greatest degree of deviation from the original signal happens whenever any transducer is involved.. and that is at the beginning and end of every audio chain. Speakers, mics, pickup cartridges for vinyl disks, etc etc introduce more color (ie distortion) to the signal path than any other (properly designed) component without moving parts.. then not to forget the acoustic properties of the room where the listening happens.. the listener will be hearing the reverberation quality of the room mixed (to a greater of lesser extent) in with the music. 99.99% of listeners (?) do not have a pair of time aligned monitors, a reference quality amplifier in an acoustically designed living room!
My impression is that the electrical specifications of audio equipment on paper can be very misleading when it comes to the final test, ones ears and brain.

ok, but analog has that 'sound' and it is something you can audibly hear - the warmth, depth, sparkle etc. This is not subconscious, so we dont need to be looking for subconscious effects to be causing it.

I think you'd agree that if you placed a high quality 4 pole low pass filter tuned to 20khz cutoff between the deck pre-amp and the amplifier, then you wont perceive any difference in the character of the vinyl character. You would still have the analog warmth, depth etc, but would knock out all the ultrasonic. - Therefore you can conclude that ultrasonic is not the reason vinyl sounds the way it does.


I guess the decision to go analog or digital comes down to this option...

Do you want the recording medium to color every signal by default beyond control of the audio engineer, or do you want to leave it to the audio engineer to color the signal by judgement and have the recording medium accurately reproduce the signal?

In a studio environment the ultimate goal is to reproduce the final output of the mix as flat across the audio spectrum as possible so you can faithfully hear what you are doing. Then you patch in some pretty average Yamaha NS-10's to see how it sounds in the average living room! This is not in any sense what audiophiles call hi-fi, and you cant get away from the fact, that what passes as audiophile is tuned to some degree to make the sound as pleasing to the ear as possible while keeping the sound accurate.

You've probably noticed that generally, the manufacturers of audiophile equipment dont have a presence in recording studios, and manufacturers of studio control room monitoring dont really have a presence in the audiophiles living room - obviously some exceptions.

I can tell you now, that if you pulled your audiophile equipment from the living room and replaced it with flat control room equipment, it would be slated as dull and lifeless and people would absolutely hate it. 2 things for 2 different jobs.

You're right - electrical specifications are generally meaningless - they can give you a very rough overview of the quality of a piece of equipment, but until you hear it you just wont know.

I don't agree with that at all. The job is to create great sound. They need excellent reference quality equipment to do that. They don't try and make it flat just so someone else can change it with their playback equipment later.


Yes, but who is doing the tuning? The audio and recording engineers...that's who. They "sweeten" the recording to make it sound better. They certainly don't try to make it just plainly and blandly accurate.


I don't see why you're making that point. Obviously the applications are different. It doesn't mean that the studio engineers intend the mix to sound flat. It's just that they are more concerned with the actual editing than they are filling a room with sound at the moment.


Absolutely true.

Now can someone do the same thing comparing say 256kbps AAC and Lossless? THAT would not be so easy. I'd like to know if anyone can reliably tell the difference and what equipment you need to do so!!!

what I meant is that in a pro studio environment the equipment they monitor with is as flat as possible, not that they try to make the mix sound flat.



As above. This is why you need flat reference equipment, so you can hear what you are doing. ie its no good adding a 3db pad to sweeten the vocals because the crossover circuit of your 'hi-fi 'speakers is attenuating the 2-3 khz range by 3db to give a hi-fi speaker a more tuned bass and top end. Thats why studio monitoring is flat - as in frequency response, because you need to hear what you are doing accurately. You then make slight adjustments using a pair of 'crap' speakers - like the NS-10's to get a feel of how it sounds on more modest systems.

Of course audio engineers try to make a mix sound sweet - and to do that properly, you use high-quality flat response reference monitoring.



well the point is - is that what passes as reference equipment in audiophile land is still tuned to flatter the ears of the listener, whereas studio grade monitoring is designed to be fully accurate so that the engineer knows how much 'sweetening' is required


I'll give you something too. I reckon the use of 'enharmonic' was wrong, both by me, and in the article I linked to, and a few others I didn't show also.

Which is exactly what the aim of the audiophile....


I am very familiar with NS10 speakers (and the controversy about which variety of toilet paper works best placed in front of the hf units to lessen the NS10's emphasis in the upper-mids and high end!). The NS10s are in no way "audiophile" quality, but they are a handy monitor to mix on, provided they are turned down low, and one be pretty sure that if something sounds fair (listenable!) on the NS10s, then it should sound fine on other systems. That was the NS10s strong point in the studio. However, I have never heard any domestic speakers which sound anything like NS10s, probably for good reason! I find them so harsh they rip my ears off!


.

For sure. Some audiophile equipment isn't even electrically compatible, in terms of headroom, impedance, voltage output levels etc etc, with the professional studio environment.. even the types of plugs and sockets installed can be incompatible, and the last thing you want in either environment is adapters....


I disagree there. Both control room and audiophile equipment should be "flat" (frequency vs amplitude) and transparent in every respect. The speakers should be as colorless as possible, and the amplifier should be the classic "wire with gain". Both engineer and audiophile want to hear the music, rather than the equipment. If music sounds dull and lifeless on high end equipment either in the control room or in the audiophile's living room, then the chances are there is a problem with the mix.

We could set it up as a poll and see how many people get it right!

OK, I see where you're going with all of that.



Thanks. I looked again and I can only find the definition I was talking about initially.