the importance of analog sound in prog |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 14 2012 at 04:12 | |||
No, I'm quite a bit younger than Pedro (Moshkito) but still older than practically everyone else here including you and José. Am I correct in guessing you didn't read all of my post where I mentioned my age?
1's and 0's have every bearing on accuracy all by themselves, that's kinda the whole point. iPlods (even though I hate, hate, hate the brand) actually sound very good and are of audiophilist quality in every measurable parameter. Our resident audiophilist (Oliverstoned) does not own a turntable but he does own an iPlod and he is far more passionate about audiophilary and valve-sound that anyone I've ever crossed swords with on this subject.
That's confusing fads and fashion in musical styles with technology.
Of course it is a coincidence. They stopped making great albums long before digital studios were the norm. Each band and each genre of music has a finite life where they are at their peak of creativity - how many bands ever "got better" as they aged? Edited by Dean - October 14 2012 at 04:30 |
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Snow Dog
Special Collaborator Honorary Collaborator Joined: March 23 2005 Location: Caerdydd Status: Offline Points: 32995 |
Posted: October 14 2012 at 04:16 | |||
So he knows. I am nearly as old as you and agree entirely with your argument.
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WisdomBeginsInWonder
Forum Newbie Joined: September 25 2012 Location: The Beggining Status: Offline Points: 32 |
Posted: October 14 2012 at 04:21 | |||
I prefer Gramophone disks,they are the best
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Snow Dog
Special Collaborator Honorary Collaborator Joined: March 23 2005 Location: Caerdydd Status: Offline Points: 32995 |
Posted: October 14 2012 at 04:23 | |||
That's fine by me.
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 14 2012 at 04:29 | |||
iPod's use Wolfson DACs and most audiophilist rate that as being a "top-end" DAC. External DACs on an iPlod are not straight forward because there is no digital-out, no co-ax or toslink connection. The only way to get digital out an iPlod is via the USB so a dedicated dock is required that contains far more "computing" than an off-the-shelf external DAC and in most cases all you are doing is replacing the iPlod's WM8740 DAC with the external unit's WM8740 DAC and losing a lot of the iPlod's functionality in the process.
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Catcher10
Forum Senior Member VIP Member Joined: December 23 2009 Location: Emerald City Status: Offline Points: 17846 |
Posted: October 14 2012 at 11:59 | |||
^ I also do not use iPod, I use the Microsoft Zune which to me in a side by side sounds better than Apple product. I also use a Zune dock which has a toslink out and I have that connected to an external DAC via toslink. That too in A/B tests sounds better with more clarity, depth and soundstage. The DAC I use also is a tube based so I can switch between direct sound or pass it thru the tube buffer and again, with an A/B test it does give a slightly analog sound, it just seems to tone the harshness down for a pleasing effect on the ears.
Straight connection using the headphone out sounds like a$$ to me, you need the dock.......thin and lifeless without it. It is very subtle as the Zune can only hold up to 320kbps files....So it is 100% for convenience sake I listen in that manner....shuffle play of 2500 songs. But since the DAC upconverts to 24/192 using toslink it does help some.......But I will always prefer redbook CDs thru my NAD CDP.....no comparison IMO. Speaking of Oliverstoned...where is he in all this? This is all 100% my opinion, but I do enjoy the hobby of trying to match gear to get thr best sound possible.....I'm not a devote audiophile from the standpoint that I will spend $1000 for a set of spkr cables, there are certain things I just don't believe in. I prefer to find that gear that gets me the best sound return for my money, looking for that last 10% of something can cost you a small fortune these days in audio gear.
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Surrealist
Forum Senior Member Joined: October 12 2012 Location: Squonk Status: Offline Points: 232 |
Posted: October 14 2012 at 13:03 | |||
Analogue is not better than digital, CD is better than vinyl, solid-state is more perfect than valve (tube) but if you prefer vinyl/analogue/valve(tube) then that's fine.
vinyl cannot have more detail than digital. iPlods (even though I hate, hate, hate the brand) actually sound very good and are of audiophilist quality in every measurable parameter. Our resident audiophilist (Oliverstoned) does not own a turntable but he does own an iPlod and he is far more passionate about audiophilary and valve-sound that anyone I've ever crossed swords with on this subject. Dean, When you make claims and statements like this, you lose all credibility regardless of how much technical jargon you throw around. Please explain to the audience how a sampling of a sound wave can be better than the original. This is absolutely absurd. |
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Catcher10
Forum Senior Member VIP Member Joined: December 23 2009 Location: Emerald City Status: Offline Points: 17846 |
Posted: October 14 2012 at 13:33 | |||
Hello,
Well seems you are enjoying the sound of your Senn HD800, which yes are a very nice set of cans. I don't listen thru HP but your connection seems fine, especially if you are enjoying the sound.
I am not sure what you mean when you bypass the signal to the AUX connection on the amp.....Bypass what signal? The only improvement in sound I suppose you could make is to get a headphone amp or run thru an EXT DAC, your iPod that is. And the best option is to connect thru a digital cable like coax or toslink (optical) cable, you will need an iPod docking station with digital out capability, not just analog.
What CD are you getting that are recorded at 24/96?? CDs are redbook standard which is 16/44.4.....You can get a digital file at 24/96, for example downloaded from HDTracks.com....So I may not fully understand what you are saying.
Best thing to do is audition if you can, and let your ears decide what you like best......I think there are many options for you to create a sound you might think is better.
At the end of the day if you like it....Then great, keep listening!
Edited by Catcher10 - October 14 2012 at 13:34 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 14 2012 at 14:16 | |||
Oliver hasn't been too complimentary on TubeDACs in the past so perhaps it's best he's not here
I've often wondered about building a small ECC88 pre-amp running in class A to replicate that authentic valve-sound (your 6N11 is an ECC88 equivalent double-triode used as a stereo class-A buffer amp ... I've seen some designs using these as headphone amps but that is highly optimistic IMO) - The Marshall Valvestate 100W head unit uses ECC88s as their only valve [the rest of the circuitry is solid-state, hence the name], so the technique is proven. Since the tube is running in class-A and doesn't use a matching transformer in the anode it will still lack the true even-harmonic warmth of a true valve amplifier, but the non-linear transfer characteristics of the valve alone will be sufficient "round off the corners" off any source (even from a turntable) [the ECC88/6N11 valves are designed to work at 250V HT - all these designs run off 24V so the transfer characteristic is pretty poor down at that anode-voltage - but since we are looking for more harmonic distortion not less, that is "a good thing"]
What you percieve as harshness is simply a flat response, which gives you more trebble than you'd really like. As I have said before, when we design engineers gave audiophilists what they were searching for (ie the perfect response of CDs and solid-state amplifiers) they suddenly realised that's not what they wanted after all: (You Maniacs! You blew it up! Ah, damn you! god damn you all to hell!)
A classic James-Baxandall tone control circuit found in most analogue amplifiers is too broad to be used to compensate for this perfect flat response as they tend to start cutting off at around 2KHz which is too low (ie the scatter-gun approach), similarily a Graphic Equaliser tends to be too drastic - what we're after is far more subtle - we don't want to change the tone appreciably and make it sound like it was recorded under a goose-down duvet, we want to tone it down just a little (or "round off the corners" as I put it). The old fashioned "scratch filter" (that kicks in at around 10KHz) was a better approximation IMO, that doesn't add "warmth" but it does tone things down a little and does not make a huge difference to all the "audio", but that cannot equal what you can do with a simple little thermionic valve. [Obviously amp modelling using DSP can achieve that to such a degree that it is possible to model the exact characteristics of different valve amplifier designs, not just that of a generic valve-sound but that is going to be an anathema to an audiophilist]
And that is essentially what I've been saying all along - we've never liked a flat response even when that is what we thought we wanted - remember the tone control settings from the sound-systems found in a typical teenagers set-up, with the bass control wound right up so the speakers cabinets physically moved everytime John Paul Jones plucked an E; or all the amps and walkmans with a "bass boost" button (my car radio still has a bass-boost button). We liked lots of bass and not very much trebble and even at 55 years of age I haven't really grown out of that (even though the tone controls on my NAD amp have never moved off their centre spot in 30 years, I compensate by having huge floor standing speakers with 10" woofers)
Same here, but I'm not going to espouse one system over another because it is all subjective. You and I are on the same page here, we both put together our setups to get the sound we like. While I understand specifications and circuit diagrams and I know all the theory and physics behind how sound is recorded and reproduced, I do not set up my system with a pink noise generator and a spectrum analyser - I use my ears and my experience.
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Catcher10
Forum Senior Member VIP Member Joined: December 23 2009 Location: Emerald City Status: Offline Points: 17846 |
Posted: October 14 2012 at 14:55 | |||
^ Never used one of those pink noise dealies either....my ears are sufficient.
My DAC came with a 6N11 tube, its not a bad tube.....I rolled in a JJ Electronics E88CC gold pins, and after about 40 hrs of burn time I did not like it. It was thin on the low end, really took away the lows I was getting with the stock 6N11 Chinese tube. I then rolled in a Jan-Phillips 6922 and that was much more to my liking....the nice low end returned and just gave a full sound and better than the stock 6N11......I will say I felt the E88CC had a very black background, just lacking in the lowend.
I like rolling these tubes, they are relatively inexpensive since I only have one, like $15-$30 ea, depending on age. So I like the different sounds I get....The Jan Phillips has about 300-400 hrs on it, its really nice and I will start looking for another make, just to see what happens.
People really like the Mullard tubes, but I have not accepted spending $75-$100 for one. I just don't think I will get much more benefit.
Send pics when you build it....would be interesting to hear your thoughts.
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Snow Dog
Special Collaborator Honorary Collaborator Joined: March 23 2005 Location: Caerdydd Status: Offline Points: 32995 |
Posted: October 14 2012 at 15:05 | |||
No he doesn't and it absolutely is not absurd. Why should he repeat ad infinitum all the points he has made?
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Surrealist
Forum Senior Member Joined: October 12 2012 Location: Squonk Status: Offline Points: 232 |
Posted: October 14 2012 at 17:33 | |||
Dean,
Please explain to the audience how a sampling of a sound wave can be better than the original. |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 14 2012 at 18:55 | |||
I'm not throwing around technical jargon (even if you think that some of what I have written is "jargon") - I am refuting statements you have made that contain technical jargon that I believe is used in (or made in) (or contains some) error or has been misinterpreted by me.
My credibility is always in question, that's why I have a tendency to prove myself more than people who make glib statements and do not expand on them.
Now by "original" I assume you mean the original acoustic sound waves and the answer to that is blindingly obvious - it cannot and neither can an analogue electrical signal be better than the original sound wave. It's a dumb statement that I have never made, nor would I ever make.
Now if by "original" you are asking how a sampling of an analogue electrical signal can be better than the original analogue electrical signal then the answer is again blindingly obvious - it cannot. And again it is a dumb statement that I have never made, nor would I ever make.
Now if by "original" you are asking how a sampling of an analogue electrical signal can be better than the recovered analogue electrical signal from a vinyl recording made from the same "original" analogue signal source, then my answer is less blindingly obvious, and an awful lot more long winded and jargon-filled. But you did ask...
So if the "audience" will indulge me further I will re-iterate and perhaps expand on some of what I have written thus far.
Analogue sounds are not "natural"
An electrical signal captured by an microphone is an approximation to the audio (acoustic) sound wave that the human ear hears, it is an analogous representation of that sound wave made in another medium (this time electrical waves in a conductor rather than sound waves in air) - this is why it is called an ANALOGUE signal - it is not the sound wave, it is an electrical wave that is an analogy of the sound wave.
With all the best will in the world that electrical signal is not a perfect representation of the acoustic sound wave, regardless of how you perceive it, it is still an approximation. There will still be some elements of the sound wave that the microphone could not capture - some sub-sonics, some supersonics, some subtle nuances, some very very faint sounds, some phase-shifts, reflections or reverberances that the human ear can hear that the microphone cannot. Human hearing is stereophonic, directional and our appreciation of sound is not just limited to the sound that enters your lug-holes, as you have noted, we can feel it, not just in the low frequency vibrations that resonate in our chest or that we can feel through our feet but also in the minute fluctuations in air pressure on our skin -all of these add to the experience and thus to the perception of sound; the best designed microphones in the world cannot capture all of that - as anyone who has some experience of studio work will affirm, there is no universal general purpose microphone that is perfect for every situation, each instrument and each type of sound and each different environment requires a specific kind of microphone that is best suited to that requirement - you don't use a Sure SM58 dynamic mic where a Neuman or a Røde condenser would better capture what you are after. Of all the audio spectrum that the original acoustic wave was composed of, only some of that will be converted into an electrical signal by the microphone even when you use the right microphone for the sound you are trying to record. Therefore the analogue signal is neither natural nor is it perfect.
Now once we accept that analogue electrical waves are not perfect then we can look at digital capture (which is a misnomer, it is digital conversion, we're not "capturing" anything). The digital conversion process takes the electrical approximation to the acoustic sound wave and converts it into a stream of numerical values, the number bits in that conversion determines the dynamic range of the reconstituted electrical signal after it has been converted back and the sampling rate determines the frequency range that can be converted. We all understand that in principle, and this is the tricky bit, we all think we know what that means in reality.
Now I said a few posts back that if all else in a hi-fi system is equal (is same amp and same speakers and same interconnects) then all we are comparing is the DAC to the stereo cartridge, so lets do just that:
If we have 16-bits then we have 65,536 different numerical values in which to encode the voltage of the electrical approximation to the acoustic sound wave ... and 65,536 sounds a lot because it is a lot - if 65,536 equates to 1 volt of input signal voltage then 1 would equate to 15.2µV - to put that into perspective if the maximum output from your super-duper moving-coil cartridge is 0.5mV and we call that 0.5mV "65536" then "1" would be the equivalent cartridge output voltage of 7.63nV (7.63 x 10-9). That last value is a total mind-flip but unless you appreciate just how small that is it will not mean anything to you, so here is a little bit more physics:
So lets look again at the super-duper cartridge - its internal resistance is typically 7ohms - this is not the impedance of the coil (which is much higher) but the simple dc resistance of the copper wire in the coil - the Johnson-Nyquist thermal noise created by that 7ohms of wire is 47.9nV ... 6.3 times more than the 7.63nV that is equivalent to 1-bit output from the cartridge. [postscript: this 47.9nV gives a SNR for the cartridge of 80dB, which is as we know is much better than what is on the vinyl - plugging 80dB back into the effective-bits formula gives the cartridge an equivalent bit resolution of 13-bits - obviously better than the equivalebnt 10-bits information on the vinyl but still a lot less than the 16-bits of CD].
What this tells us is 16-bit conversion is not bad and 65536 different levels to convert an electrical analogue of an acoustic sound wave is better than the best stereo cartridge in the world can ever replicate. I have already explained why the electrical signal reconstituted from a vinyl record is the equivalent of 10-bit digital signal, now we see that even if vinyl and CD were completely equivalent (ie both equivalent to 16-bits) then the cartridge could not reproduce all those bits, the finer detail would be lost in the inherent noise in the cartridge itself. Digital is capable of representing a far wider range of voltage approximations to the acoustic sound wave than either the vinyl media or the stereo cartridge was ever capable of reproducing.
Now lets address the issue of the sampling frequency, which (we all should know by now) is 44.1 kHz and Mr Nyquist (that bloke again) tells us that this sampling frequency limits the maximum frequency that a CD can convert is 22.05KHz. But that isn't strictly true because of sub-sampling, which means that signals greater than 22.05KHz will be converted but not at the right frequency (they will be aliased), so we need to employ anti-aliasing to limit the bandwidth of the electrical analogue approximation to the acoustic sound wave.
The pro-analogue argument goes that these "missing" supersonic frequencies are "important", not because we can hear them (because unless you are of a non-human species, you cannot), but because we can "feel" them. I'm not 100% convinced of that but I'm prepared to go along with it, I think it is more likely that what we can "feel" is actually those above-audio tones beating with other tones to produce beats we can hear, but as I say, I'm prepared to go along with the idea that for whatever reason, these above-audio supersonic frequencies can be "felt" or heard in someway or other. The problem I have with that is the bandwidth of tape and vinyl recordings is also bandwidth limited, so even though vinyl is capable of recording frequencies as high as 45KHz, those frequencies are not present on any commercial vinyl recording. (there are technical reason for this - firstly to allow more disc surface area to be used for audible part of the signal, and secondly to prevent the cutting stylus on the disc cutting lathe from overheating). Now if my idea of beat frequencies is more true than the touchy "feely" idea then this bandwidth restriction is not a problem, because the beats (being audible) would have been detected by the microphone, and as any radio-ham will tell you, once you've got the side-band you can remove the carrier and not lose any information. So yes, in the real acoustic sound wave world supersonics are important (may be), but once you've captured those effects with a microphone you don't need them anymore. And anyway, just because the vinyl platter can theoretically record frequencies above 20KHz it does not imply that the rest of the system can. Making amplifiers work at frequencies greater than 20KHz is not something that design engineers waste time and money on because it makes the amplifer unstable and prone to all kinds of problems, audio amplifiers are bandwidth limited by design and this is controlled by negative feedback that prevents the amplifier from goinf unstable (and this goes for valve(tube) amps too - also triode valves are dreadfully poor at high frequencies and so are the output transformers, which are designed to operate across the audio spectrum of 20Hz to 20KHz and no higher).
Since we are on the topic of frequency response and bandwidth, lets drag the super-duper stereo cartridge back into the arena for another look under the specs. The bandwidth of CD and DAC is 20Hz to 22.05KHz ±0.01dB (as near as dammit) - the bandwidth of a Goldring Legacy is 20Hz to 22kHz ±3dB ... now that ±3dB seems so insignificant but decibels are logarithmic and ±3dB in terms of linear voltage is a factor of 2 ... the output level of the cartridge can vary from 2.5mV to 5mV across the frequency range and still be regarded as "in spec" and "flat". That doesn't look to be "perfect" to me. When I compare that to the values from a CD I would have no choice but to say that CD is better. The Goldring specification also says the channel separation is 25dB - what that means is that for a sound output on the left channel we will "hear" 1/18th of that on the right channel as crosstalk. On a CD the channel separation is infinite - crosstalk is impossible, again I would have to conclude that CD is better in this respect.
I heard you the first time, these posts don't write themselves.
Edited by Dean - October 15 2012 at 08:23 |
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Surrealist
Forum Senior Member Joined: October 12 2012 Location: Squonk Status: Offline Points: 232 |
Posted: October 14 2012 at 22:25 | |||
Ok..
I do appreciate you taking the time to discuss the science and theory. Could you explain to me why an 8 track Alesis ADAT recording sound or a Digital FIREPOD with 8 digital inputs recording a room full of music sounds absolutely dreadful when played back compared to say a Tascam 8 track reel to reel. Same mics.. same music.. it's NOT subjective. It sounds flat, cold, metallic, unnatural and utterly lifeless. Then you spend the next month trying different filters and plugins, amp simulators.. all kinds of stuff that try to get you a sound you can live with. I have done this... and where I recorded a comparison between digital and tape machine and have played this back to people and it is utterly shocking how poor the digital is.. again .. same mics, placement.. room.. board, playback, speakers.. I all seriousness I would love to hear your explanation. Thanks |
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Surrealist
Forum Senior Member Joined: October 12 2012 Location: Squonk Status: Offline Points: 232 |
Posted: October 14 2012 at 22:59 | |||
A vinyl record is an analog recording, and CDs and DVDs are digital recordings. Take a look at the graph below. Original sound is analog by definition. A digital recording takes snapshots of the analog signal at a certain rate (for CDs it is 44,100 times per second) and measures each snapshot with a certain accuracy (for CDs it is 16-bit, which means the value must be one of 65,536 possible values).
This means that, by definition, a digital recording is not capturing the complete sound wave. It is approximating it with a series of steps. Some sounds that have very quick transitions, such as a drum beat or a trumpet's tone, will be distorted because they change too quickly for the sample rate. In your home stereo the CD or DVD player takes this digital recording and converts it to an analog signal, which is fed to your amplifier. The amplifier then raises the voltage of the signal to a level powerful enough to drive your speaker. A vinyl record has a groove carved into it that mirrors the original sound's waveform. This means that no information is lost. The output of a record player is analog. It can be fed directly to your amplifier with no conversion. From the graph above you can see that CD quality audio does not do a very good job of replicating the original signal. The main ways to improve the quality of a digital recording are to increase the sampling rate and to increase the accuracy of the sampling. |
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Surrealist
Forum Senior Member Joined: October 12 2012 Location: Squonk Status: Offline Points: 232 |
Posted: October 14 2012 at 23:56 | |||
Why Vinyl Records Sound Better Than CDWhy does it sound better?Some
people have a bit of an obsession with vinyl. Maybe it's just because
they think it is vintage and therefore cool. Maybe they are just old
school and not want to move on. Maybe they are complete audiophiles and
love their music that much. Digital EncodingIn
order to encode a similar signal onto digital media such as a CD or
DVD, a conversion from analog to digital is required. No matter how good
the conversion is, there will always be losses which occur through the
transition. Technology may get better and better, reducing these losses
but it is effectively impossible (at least in modern times), to
reproduce an analog signal exactly with digital data. Take our previous
example of a gradual change between two frequency values. Now, we have a
series of discrete changes between them. Imagine if you were told to
sing from as low as you can up to as high you can using only three
different tones. You would sing low, a medium note and then high. This
is a very extreme example, but shows what we mean by discrete steps.
Improved technology would allow you to change frequencies more often in
the same period, say 5 times. Now you have a closer reproduction of the
original smooth variant, but it still isn't great. This is shown by the
second graph in the picture above. |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 15 2012 at 02:35 | |||
Oh my giddy aunt. I don't know whether to laugh or bloody cry. Wherever you copied that from you evidently didn't scroll down and read the inevitable barrage of comments that should have accompanied it. The brief text contains so many factual inaccuracies it makes me want to weep.
First off I defy you to tell the difference between a sine, square and triangle wave at 10KHz on any audio system because you cannot. This is not subjective or based upon ABX listening tests, it's a fact of life. A 10KHz sawtooth (which is what that depicted CD "output" is tending towards) is rich in harmonics (it has more information in it than a 10KHz sine wave - it has converted the "complete" sine wave and then added some more - this means NO INFORMATION IS LOST) - those harmonics start at 20KHz (the first harmonic) then go up in doubling steps to infinity, so the next would be 40KHz and the next 80KHz and the next 160KHz etc etc. Regardless of whether CD or vinyl can replicate those harmonics above 20KHz (they cannot) the average human ear cannot hear the harmonic at 20KHz. All you can hear is the 10KHz component and that is a sine wave, the same sine wave as the "original" analogue approximation to the acoustic sound wave. All CD players use anti-aliasing filters, these remove all those harmonics so the output from your CD player looks nothing like that stepped waveform - it looks like a 10KHz sine wave, the same 10KHz sine wave of the "original" analogue approximation to the acoustic sound wave and that's exactly what it sounds like. If you don't believe me try it - record a 10KHz sine wave, digitise it to [email protected], burn it onto a CD and then look at the output of the CD player's DAC with an oscilloscope - then listen to that on your bestist headphones in the whole world ever and then compare that to the original. I guarantee you will hear no difference.
Secondly you cannot hear the grooves in a vinyl record (I think I've said this before) - you have to convert the mechanical grooves into kinetic energy in the stylus and then convert that into magnetic energy the magnetic field of the cartridge and then convert that into electrical energy in the electrical coil of the cartridge - which is fed to your amplifier. The amplifier then raises the voltage of the signal to a level powerful enough to drive your speaker. Just because you are doing these three conversions in the analogue domain does not mean it is perfect, far from it - if it were perfect then no one would ever spend $700 on a cartridge (and those still are not perfect) - perfection means accuracy, so since the output from a cartridge is not perfect it is not an accurate representation of the original acoustic sound wave either.
Thirdly, if people are going to claim "Some sounds that have very quick transitions, such as a drum beat or a trumpet's tone, will be distorted because they change too quickly for the sample rate." then prove it - it really is a trivial thing to do - produce a spectral analysis of the drum beat or trumpet's tone and prove that those very quick transitions are not only higher than the Nyquist frequency of the digitiser, but also that vinyl can recover them. Don't make specious comments without proof.
Edited by Dean - October 15 2012 at 06:31 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 15 2012 at 03:48 | |||
Oh my poor aunt is now whirling like a dervish on 78 ... much more of this and she's going to be ill. Whoever wrote that (certainly not you) has made some factual mistakes and while it reads as very convincing to the uninitiated, it is really no better than that previous cut'n'paste article you posted. And its tone is really rather patronising, which is extremely disappointing. *sigh* The article makes a gross error on what digital quantisation does. He is claiming that the digitiser quantises the signal in the frequency domain - it does not - it quantises in the voltage domain and that completely destroys the point he is making and any argument he has to support it. Sorry, but it really is that bad. It isn't scientific at all, its totally erroneous. If you do not understand what I have just written and cannot see the error yourself then I'm at a bit of a loss as to how to explain that in simple terms, but I shall try. In his singing example he said "Take our previous example of a gradual change between two frequency values. Now, we have a series of discrete changes between them. Imagine if you were told to sing from as low as you can up to as high you can using only three different tones. You would sing low, a medium note and then high. This is a very extreme example, but shows what we mean by discrete steps" Fortunately that's not how it works, the digital signal does not contain discrete steps of frequency, it contains the exact same gradual change that the singer produced when modulating from one note to another and that transition will be as smooth as any analogue representation. Digital sampling does not quantise frequency. The transition from one note to the next is slow - no matter how good a singer or player you are the transition from one note to another will take seconds, this is the equivalent of a very low frequency modulation (which is why key-changes are sometimes called "modulation" and where the "M" in FM radio comes from) - note that I said "equivalent" there - that very low frequency change does not exist in the waveform as a discrete frequency component. If you were to produce a spectral analysis of that change in tone you would only see frequencies between the two tones - nothing above and nothing below, and if you digitised that change and then converted it back into analogue and repeated the spectral analysis then the same set of frefrequencies would still be present. His graph is also a little misleading since it does not show what he is trying to illustrate, but never mind, he was wrong anyway. His comment about the resolution of the human ear is also wrong but that's by the by - the human ear can discerner power-level changes of slightly less than 3dB ... this figure is very important and is used a lot in the audio world (analogue and digital) because it is the smallest change the human ear can detect - 3dB represents twice the power coming our of your speakers - that's how low-res the human ear is.
Edited by Dean - October 15 2012 at 04:35 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: October 15 2012 at 04:27 | |||
Sorry to disappoint but this is the epitome of "subjective" ... this is exactly what subjective means. The words "flat", "cold", "metallic", "unnatural" and "utterly lifeless" are totally subjective terms - there is no objective definition for any of them. Obviously I cannot give an explanation because I have not heard this subjective comparison. I'm a little puzzled by your use of the phrase " say a" but not enough to press it further - you either used a Tascam 8 track or you didn't, but it matters nothing either way since I wasn't present at the listening so cannot comment. I would ask whether that was a double-blind ABX test, whether the "people" knew which source was which, whether they had any preconceptions prior to the comparison and whether you could have subconsciously biased the examples (it happens, body language, tone of your voice, general attitude - not accusing you of doing that, but I have to raise it as possibility) but all that is irrelevant because as an engineer if the difference is that obvious (and I mean statistically proven that everyone can tell the difference every time .. and for that I do not mean that most people can tell the difference, or everyone can tell the difference most of the time¹) then as an engineer I would want to know what the differences were and where they came from, because if this is scientifically and technically demonstratable and repeatable then I could completely rewrite every textbook on this subject. However, after 30 years of this no one has done that and if it was that bleedin' obvious then someone would have by now. ¹ This proviso sounds harsh but that's how it is - if you asked people to separate a horses from cows and they got it right most of the time or most of the people got it right all of the time you would not call that a great success, in fact you would call it a failure.
Edited by Dean - October 15 2012 at 06:06 |
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What?
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Snow Dog
Special Collaborator Honorary Collaborator Joined: March 23 2005 Location: Caerdydd Status: Offline Points: 32995 |
Posted: October 15 2012 at 05:01 | |||
Don't you get sick of doing this every six months or so Dean?
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