apl
Stand Up Comity
The Cirque/Beatles Love project CD is noteworthy. The sounds they got off those old tapes is great, and the project would have been impossible without ProTools. So a good marriage of analogue and digital.
ausrock
Well-known member
And if I remember correctly, it was handled by George Martin and his son with full approval from the Beatles "family".
msh,
Yes it is an interesting "study", one which if in any way, shape or form is correct, could potentially have some interesting implications but that's not really for this thread.
msh,
Yes it is an interesting "study", one which if in any way, shape or form is correct, could potentially have some interesting implications but that's not really for this thread.
apl
Stand Up Comity
...but that's not really for this thread.
LOL! You gonna report him to a mod if he posts it?
ausrock
Well-known member
You misunderstood, I wasn't thinking of the technical aspects..........what possible conclusions may be drawn from that study if the results are reliable, that is what I was refering to.
ethyrvalve
Norman Nanoweber
If I may... am I the only one who thinks 'Love' was a total cheese-ball abomination? George Martin truly besmirched his good name with that steaming pile of cirque du soleil crap. it makes the bee gees' sgt pepper sound like a work of inspired cocaine-free genius by comparison. There's too much wrong with it to really say how much, if any, of the blame should go to pro tools.
apl
Stand Up Comity
You may, and you are not alone. But a lot of people do like it.
Are you complaining about the artistic aspect or the technical aspect?
T
Tim Gillett
Banned
I'll jump in. Something about serious resampling and remixing from the multitracks when two of the original artists arent there to approve seems a bit questionable. Lennon and Harrison would have approved the release of the finished mixes but the multitracks from which they were made wouldnt have been in the same category.
But also it could be seen as a mediocre producer talent trading off the probably much greater talent of the fab four.
I had a lot less of a problem with Natalie Cole's duet with her long deceased Dad, Nat on the Unforgettable track. She's an artist in her own right, and a good one too, and it was only one track on an otherwise excellent album.
Somehow seemed more right.
Tim
But also it could be seen as a mediocre producer talent trading off the probably much greater talent of the fab four.
I had a lot less of a problem with Natalie Cole's duet with her long deceased Dad, Nat on the Unforgettable track. She's an artist in her own right, and a good one too, and it was only one track on an otherwise excellent album.
Somehow seemed more right.
Tim
ethyrvalve
Norman Nanoweber
From my perspective, when you're listening to something so artistically flawed it's tough to analyze the technical aspects. it's kinda like saying "hey i know we're in a gas chamber, but that florescent lighting sure brings out the subtlety in the wallpaper pattern?"
no offense to circle du soleil fans or any gas chamber aficionados out there.
mamm7215
Member
K as a Massive Beatle fan I gotta chime in here. The Beatles Love/Cirque show was AWESOME to experience. My wife and I went to Vegas last year for our 10th and saw it. Artistically flawed? Dunno how you arrive at that but to have that opinion is fine, great even, but there was nothing flawed about it. You may not like/appreciate what they did to the music, but it sounded great and was artistically "interesting" at worst. Also, Paul and Ringo, Yoko, George's wife and George and Giles Martin all approved the work and Paul and Ringo actually had hoped they'd do more weird combo stuff with the music. Being there at the show, the sound was stunning, and the show was amazing to watch. So there's an opinion from the other side of the argument, that also has nothing to do with the original topic of this thread...
ethyrvalve
Norman Nanoweber
apl
Stand Up Comity
I had chicken stirfry for lunch.
And coffee.
And coffee.
mamm7215
Member
Absolutely no ill will. 
cjacek
Analogue Enthusiast
Lemme try again...
OK, the biggest problem with the CD is not only hard wall limiting of frequencies but as important and even more so is the low sampling that it uses, effectively throwing away large chunks of original material.
So while the CD can sample up 20khz [and every other frequency from 20hz], it does this in an incomplete way, throwing aside most of the 'meat' of those frequencies. That is why, in essence, you get a 'sample', in the strictest sense of the word.
[Speaking strictly about cassettes]:
A good cassette / cassette deck [emphasis on 'good'], while capable of even exceeding past the audible human hearing range [some cassettes / decks can do over 20khz], will never use sampling or hard wall limiting, thereby more effectively storing the original signal as it went in, even if it is [sometimes] more limited in how high or low it can go, as in storing those frequencies.
Lets say, in this example, that my cassette can do 50hz to 15khz and that the CD can do 20hz to 20khz. Going by those numbers, the CD should win hands down, as far as frequencies go. What is not talked about is that the cassette will have more 'meat' between its rather 'narrower' range than the sampled / chopped up CD, in its wider range.
To illustrate above example:
50hz _________________________________15khz [analogue - cassette]
20hz . . .. .. . ... . . . . . . . . . . . . . . . . .. . . . .. . . . .. . . ..20khz [digital - CD]
Obviously there are problems with the cassette format too but point is that it's not only about frequencies but the dramatic limitations placed on the CD, as far as sampling and frequencies.
Again, I ask, why doesn't the SACD format take over from CD? The Super Audio Compact Disc would eat, for breakfast, any of the digital formats past and present, offering the highest resolution since some of the better analogue tape decks. It could have been the last word on digital audio high resolution format for many years.
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apl
Stand Up Comity
This is not true. The sampling is discrete in time, but the resulting information is continuous in the frequency domain. Unless I really, really missed something big in Signal and System Analysis.
But, hey, I've been wrong before.
Got a link explaining it?
B
Beck
Guest
Tim,
You make a valid point, but…
To facilitate understanding it's crucial not to ascribe any and all analog arguments to every member here. It would indeed cause some confusion to assume any one member speaks for all.
For example, I don’t know what I think about frequencies that are considered to be higher than humans can hear, but my mind is wide open on the topic. However, it’s not really on my radar at this time as to why I prefer analog.
Another thing that may be helpful is to realize that those of us who can hear a clear difference between analog and digital don’t necessarily need to have it explained in technical terms. Oh sure, it’s interesting to discuss, but these discussions really boil down to those of us that can hear it trying to find a way to communicate our perceptions to those who lack the same auditory sensitivities… perhaps an exercise in futility. If only hearing could be corrected as easily as vision.
We are in a sense in the difficult position of trying to describe a color that most people can’t see.
When vinyl was prevalent it was common to roll off frequencies above 18 kHz and below 38 Hz or so in the mastering phase. Yet, LPs sound better than CD to my ear. There is obviously much more to it than frequency response.
One more quick note: I see a common error in how many people interpret frequency response as listed on a spec sheet.
Tape (even cassette) is capable of recording and reproducing frequencies in excess of the listed frequency response. The frequency response specification refers to a frequency range within plus or minus a number of dB. Thus you will see a typical reel-to-reel deck response listed as 30 Hz to 20 kHz +/- 3dB. This simply tells us that if 1 kKz is 0, the frequencies between 30 Hz and 20 kHz will not vary more than 3 dB up or down from 0.
It is a measure of flatness, and not an absolute measure of the frequencies the tape and the machine can accommodate. Since 20Hz-20kHz is an industry standard range based on the range of human hearing, many manufacturers don’t list higher or lower response even though the deck is still within +/- 3 dB above 20 kHz. Some do… for example my Akai GX77 reel-to-reel at 7.5 ips is rated 25 Hz -- 26 kHz +/- 3 dB @ 0 VU. That increases to 25 Hz – 33 kHz +/- 3 dB @ -20 VU. And if the spread were greater, say +/- 5 dB the range of frequencies could go even higher.
The bottom line – unlike CD, which has an absolute maximum 20 kHz top end, tape actually contains frequencies much higher, and those frequencies can be emphasized or deemphasized with EQ because they are there. With CD they aren’t even there. Again, I'm not sure about the role of ultrasonic frequencies, but I’m not ready to dismiss it either.
evm1024
New member
Nyquist and the real world.....
Nyquist/Shannon is often held up as the defining theory that proves that digitally sampled audio is flawless (OK so I over stated it, grant me that). All signals sampled below the Nyquist freq can be reproduced. People often do not understand what the theory actually states and also tend to forget the conditions required by the theory.
Here is a NASA paper which explores one of the distortions that occur in sampled signals well below the nyquist cutoff. Take a moment to read:
http://ntrs.nasa.gov/archive/nasa/casi.ntrs.nasa.gov/20000120584_2000177329.pdf
Please skim over the full paper even if you choose not to read it fully.
Regards, Ethan
Nyquist/Shannon is often held up as the defining theory that proves that digitally sampled audio is flawless (OK so I over stated it, grant me that). All signals sampled below the Nyquist freq can be reproduced. People often do not understand what the theory actually states and also tend to forget the conditions required by the theory.
Here is a NASA paper which explores one of the distortions that occur in sampled signals well below the nyquist cutoff. Take a moment to read:
http://ntrs.nasa.gov/archive/nasa/casi.ntrs.nasa.gov/20000120584_2000177329.pdf
Please skim over the full paper even if you choose not to read it fully.
Regards, Ethan
ausrock
Well-known member
The following is "an article" I saved off the web sometime in the past and it appears the link to it is no longer valid so I'll just copy it here.......also, I don't either agree or disagree with the contents 
................:
"Sampling and Nyquist's Theorem for Audio and Video
Updated 10/29/03
All line doublers, good comb filters, and DVD players process video as digital and must sooner or later convert it back into analog form.
This is an explanation of why video resolution and audio frequency response might not as great as Nyquist's theorem and sampling theory would otherwise suggest..
--------------------------------------------------------------------------------
In a Nutshell
All analog to digital conversion of audio and video is accomplished by sampling. Nyquist's theorem states that it is only necessary to take samples at a rate of slightly over two per cycle of the highest frequency component of the source analog signal.
That is only half of the story. The other half is that when the highest frequency content of the subject is close to half the sampling rate, the process of turning the samples (digital representation) back into a waveform or video scan line (analog representation) is complicated and sometimes not done right.
For video, we take samples by dividing up the picture into a grid of evenly spaced spots we call pixels. The highest frequency component stands for the finest details. A cycle consists of one dark detail followed by one light detail.
The fallacy in video or audio reproduction is that recovering the analog signal might be done simply by "connecting the dots or samples". I believe that the terminology used is "analog domain" for the original waveform and "digital domain" for the conglomeration of samples (pixels in the case of video). Proper conversion functions, I believe they are called sinc, not sine, functions, transform the video signal back into the analog domain. Connecting the dots produces an analog signal but it is really still in the digital domain, the pixel footprint remains impressed on the video. This explains why we lose resolution when samples straddle rather than coincide with details.
--------------------------------------------------------------------------------
A Few Rules:
1. Fourier's theorem states that any complex waveform is the sum of sinusoids (sine waves, the simplest kind of waveform). Any cycle of the subject waveform can be selected and considered to be one cycle of a periodic (repeating) waveform where the fundamental frequency is represented by a sine wave of that wavelength. If the selected cycle is not itself a sine wave, then the other sine waves making up the sum are all multiples of the fundamental frequency.
2. Nyquist's theorem states that, if we sample the complex waveform uniformly at a rate just a tad over twice the highest frequency component sine wave contained within, the conglomeration of samples thus obtained are sufficient information to reconstruct the waveform.
3. A mathematical function, when given a particular set of inputs, always produces the same result. Although more than one set of inputs can yield the same result, one set of inputs cannot cause a function to deliver one output some of the time and another output some other time (unless time is itself one of the inputs).
Nyquist's frequency is the frequency that no part of the source material may attain or exceed, which is half of the sampling frequency.
Consider the following situation:
Let's pretend that in the left diagram below assume we are sampling at very slightly over twice the frequency of the subject waveform, and in the fashion as shown. (For video pretend that black is "up" and white is "down".)
Suppose that we reconstruct the analog output by "connecting the dots" as in the center diagram. Here's a secret: the angular waveform in the center has a fundamental frequency equal to half the sampling frequency (not exactly the frequency of the original waveform), also it comprises many harmonics, or sine waves that are multiples of the fundamental, as well. (Restating rule 2 above, any repeating waveform that is itself not a plain sine wave has harmonics of its fundamental frequency.) In a real life system with analog circuits there is a bandwidth and for the highest fundamental frequencies the system is designed for or contrained to, their harmonics won't pass through and such high frequency signals are reduced to (de-facto low pass filtered into) pure sine waves represented by the waveform on the right.
Proper reconstruction of the original waveform can be expressed in English as "curve fitting". For this latest discussion, dots depicting the samples in the above diagram represent a frequency slightly greater than the original analog signal frequency as we explained earlier. The curve we want to fit and hit all the dots represents the finished waveform but we have an added constraint:
4: The finished reconstructed waveform must not possess any content (such as harmonics) greater than or equal to half the sampling frequency as would be revealed if that waveform were re-decomposed in accordance with Fourier's theorem.
If we did not have rule 4, then more than one curve could be found that went through all the samples and we would have violated rule 3.
Thus I suppose (I am not a Fourier and Nyquist expert) that the one and only curve that obeys rule 4 and connects all the dots is the original waveform.
The angular waveform in the middle, using what we explained earlier, clearly violates rule 4 due to the harmonic content. We can't see from the small diagrams above but if we use the rightmost diagram to represent half the sampling frequency itself, it as a finished reconstructed waveform also does not satisfy rule 4; the highest frequency allowed in the original material is an infinitesimal tad less.
--------------------------------------------------------------------------------
Lookahead
In reconstructing the original analog waveform (and also for manual curve fitting using French curve templates), the process can't take just one dot at a time but instead must size up several upcoming dots (looking ahead) at all times. For example we may work on samples 1-8, then work on samples 2-9, then 3-10, and so on. The closer any of the actual frequency content is to the sampling frequency, the more lookahead is needed (I don't know the formula for how much). Insufficient lookahead means that the reconstructed waveform might come out different from the original, for example the original signal on the left in the above diagram might still be reconstructed as the waveform on the right. The lesser the amount of lookahead, the greater the chance that the reconstruction logic runs into a situation where it cannot "connect the dots" and still conform with Rule 4. What usually happens is that the reconstruction logic inserts a discontinuity (a sudden bend that violates Rule 4) and then keeps on going.
--------------------------------------------------------------------------------
Output Oversampling
Some D/A conversion techniques use digital processing where several graduated steps are used to shape the output waveform between samples. Were it not for these graduated steps, that process could not do better than an ordinary "connect the dots" process that violates Rule 4. This process with graduated intermediate steps is referred to as output oversampling.
--------------------------------------------------------------------------------
Output Low Pass Filtering
Even though it may make an excellent attempt to conform with rule 4, the D/A conversion may still produce an output waveform with spurious high frequency content. Jaggies or stairstepping or squared off peaks or peaks of the right width but too sharp seen if the waveform is graphed are examples of such "noise". For example the process using output oversampling as described immediately above will still have tiny jaggies in the waveform emerging from the D/A converter. The process is still considered successful when an analog low pass filter follows and the output after that conforms with rule 4.
Of course the analog waveform is reconstructed from the samples using mathematical formulas, not pure trial and error.
--------------------------------------------------------------------------------
Noise, Accurate Sampling, and Accurate Filtering
To conform with Rule 2, the original material must not possess any harmonic content in excess of (or equal to) half the sampling frequency. It is necessary to low pass filter the input material to prevent violation of this rule. Should noise entering the system after this filtering represent higher frequency content, the samples can be corrupted and at the other end, the D/A conversion, the finished waveform that obeys Rules 3 and 4 above will not be correct.
It is common in audio applications for sample values to be accurate to 16 bits, or one part in 65536, or even just 8 bits or one part in 256. When the content is very close to half the sampling frequency, slight inaccuracies in the samples can result in major inaccuracies in the finished analog waveform. There could even be large peaks in the output that were not present in the original material.
The closer the content comes to half the sampling frequency, the more complicated the D/A conversion calculations must be. Practically speaking, this means more expensive circuitry. And the more correct the D/A conversion is, the more likely and more profound artifacts from noise and inaccurate sampling will be with respect to the highest frequency content.
Noise, sampling inaccuracy, and D/A conversion accuracy all interact to govern the quality of the output. By choosing a higher sampling frequency relative to the highest frequency in the content, inaccuracies in sampling and less than perfect D/A conversion become less important.
Now back to the world of video, and the bad news.
5. Digital to analog converters, like comb filters and line doublers, come in different levels of quality. Nowadays every DVD player (and every line doubler) needs a D/A converter to get the video back into analog form to send down a component video or S-video cable. The ultra simple D/A converters merely connect the dots i.e. look ahead zero pixels beyond the pixel being worked on..
6. The video source material may contain details finer than the pixel spacing. In mathematical, Fourier, and Nyquist terms, that means the material being sampled contains frequencies in excess of half the sampling frequency. When this happens, all bets are off regarding the ability to recreate the input from the samples. Incidentally, recording the video as analog at first and low pass filtering it will remove all details that are too fine to sample.
7. Video samples typically have an accuracy of no better than one part in 256. (The luminance or dark/light value for each pixel on DVD is an 8 bit number.) More lookahead is then needed to properly reconstruct the highest frequencies, and if the high frequency content is only brief (fine detail for a short horizontal distance across the screen) accurate reconstruction is not possible.
8. As part of the MPEG compression used to make the entire movie fit on the disk, the samples (one sample equals one pixel) may be manipulated or substituted with similar but not identical samples from another field or frame. Then turning them back into the analog waveform, even with perfect Fourier formulas, can no longer produce the original waveform. However if simple connect the dots D/A conversion were used, the difference between the incorrect results obtained from manipulated and compressed data and the incorrect results obtained from unmanipulated and uncompressed data is less profound.
--------------------------------------------------------------------------------
Another "Connect the Dots" Example
Let's imagine a picket fence subject where thirteen pixels corresponded to six pickets and six spaces between pickets. What we see is portions of the fence with pickets appearing slightly closer together than they really are (using the pixel spacing, not the picket spacing) and other portions of the fence blurred out where the pixels and pickets were out of phase. A correct D/A conversion would show the fence with all the pickets nice and crisp at their original pitch (the video signal is analog when it gets to the picture tube).
Here is a diagram showing details going in and out of phase with the pixels. Actually the nature of analog video is such that, near the resolution limit, the reproduced bars will never be as crisp as the top row in the diagram below. The reproduction at the extreme lower left and lower right is excellent. The purpose of the diagram is to show the profound loss of detail in some places due to pixel straddling.
--------------------------------------------------------------------------------
The good news: Connect the dots D/A conversion doesn't look too bad
Nobody counts the pickets in a fence when watching a movie.
If details were lost to pixel straddling in one video frame, chances are they will show up in the next frame in normal motion.
--------------------------------------------------------------------------------
Foldover Aliasing
What happens, if we may ask, if we sample material that does contain content greater than half the sampling frequency?
It so happens that if the highest frequency that is supposed to be present is less than X hertz (we sample at twice X hertz), if the source material happened to contain X+Y hertz, the samples obtained would be the same as if the source material contained X-Y hertz. For example we sample some audio at 10 KHz. (For telephones, we don't need frequencies greater than 5 KHz.). If per chance there was some 6 KHz content, we recreate the analog audio with 4 KHz (5 minus 1 KHz) content instead of the original 6 KHz (5 plus 1 KHz) in accordance with rule 4 above. For audio, this produces sour notes. So for audio, it is necessary to do low pass filtering of the original source material prior to A/D conversion.
The phenomenon of frequencies that are more than half the sampling frequency coming back as frequencies that are less than half the sampling frequency is referred to as foldover. It is a kind of aliasing in that the result after conversion back to analog looks like something else, namely what could have been another original source with such lower frequencies already present.
We won't explain thoroughly here the reason for wagon wheels seemingly rotating in reverse as seen in the movies which is also a situation where the sampling frequency is too low. But as a hint we will state that the sampling frequency is the frames per second rate and the frequency we are trying to reproduce is the rate at which spokes pass the "6 o'clock position" on the wheel.
--------------------------------------------------------------------------------
A De-Rating Factor for Digital Video
In this writer's opinion, we need to take three samples for each light and dark detail pair (line pair; waveform cycle) for still pictures such as from a computer scanner. Thus a photograph that supposedly has at most 200 dots per inch of resolution would need to be scanned at 300 DPI. In video the subject is usually in motion so details that are straddled in one video frame would likely be clear in the next video frame. The Kell factor represents the ratio between lines of resolution and pixels (spanning the same distance used for reference). It is subjective, and is said to be about 0.7 for still pictures and early (1950's) monochrome video, and about 0.85 to 0.9 for full motion video."
................:"Sampling and Nyquist's Theorem for Audio and Video
Updated 10/29/03
All line doublers, good comb filters, and DVD players process video as digital and must sooner or later convert it back into analog form.
This is an explanation of why video resolution and audio frequency response might not as great as Nyquist's theorem and sampling theory would otherwise suggest..
--------------------------------------------------------------------------------
In a Nutshell
All analog to digital conversion of audio and video is accomplished by sampling. Nyquist's theorem states that it is only necessary to take samples at a rate of slightly over two per cycle of the highest frequency component of the source analog signal.
That is only half of the story. The other half is that when the highest frequency content of the subject is close to half the sampling rate, the process of turning the samples (digital representation) back into a waveform or video scan line (analog representation) is complicated and sometimes not done right.
For video, we take samples by dividing up the picture into a grid of evenly spaced spots we call pixels. The highest frequency component stands for the finest details. A cycle consists of one dark detail followed by one light detail.
The fallacy in video or audio reproduction is that recovering the analog signal might be done simply by "connecting the dots or samples". I believe that the terminology used is "analog domain" for the original waveform and "digital domain" for the conglomeration of samples (pixels in the case of video). Proper conversion functions, I believe they are called sinc, not sine, functions, transform the video signal back into the analog domain. Connecting the dots produces an analog signal but it is really still in the digital domain, the pixel footprint remains impressed on the video. This explains why we lose resolution when samples straddle rather than coincide with details.
--------------------------------------------------------------------------------
A Few Rules:
1. Fourier's theorem states that any complex waveform is the sum of sinusoids (sine waves, the simplest kind of waveform). Any cycle of the subject waveform can be selected and considered to be one cycle of a periodic (repeating) waveform where the fundamental frequency is represented by a sine wave of that wavelength. If the selected cycle is not itself a sine wave, then the other sine waves making up the sum are all multiples of the fundamental frequency.
2. Nyquist's theorem states that, if we sample the complex waveform uniformly at a rate just a tad over twice the highest frequency component sine wave contained within, the conglomeration of samples thus obtained are sufficient information to reconstruct the waveform.
3. A mathematical function, when given a particular set of inputs, always produces the same result. Although more than one set of inputs can yield the same result, one set of inputs cannot cause a function to deliver one output some of the time and another output some other time (unless time is itself one of the inputs).
Nyquist's frequency is the frequency that no part of the source material may attain or exceed, which is half of the sampling frequency.
Consider the following situation:
Let's pretend that in the left diagram below assume we are sampling at very slightly over twice the frequency of the subject waveform, and in the fashion as shown. (For video pretend that black is "up" and white is "down".)
Suppose that we reconstruct the analog output by "connecting the dots" as in the center diagram. Here's a secret: the angular waveform in the center has a fundamental frequency equal to half the sampling frequency (not exactly the frequency of the original waveform), also it comprises many harmonics, or sine waves that are multiples of the fundamental, as well. (Restating rule 2 above, any repeating waveform that is itself not a plain sine wave has harmonics of its fundamental frequency.) In a real life system with analog circuits there is a bandwidth and for the highest fundamental frequencies the system is designed for or contrained to, their harmonics won't pass through and such high frequency signals are reduced to (de-facto low pass filtered into) pure sine waves represented by the waveform on the right.
Proper reconstruction of the original waveform can be expressed in English as "curve fitting". For this latest discussion, dots depicting the samples in the above diagram represent a frequency slightly greater than the original analog signal frequency as we explained earlier. The curve we want to fit and hit all the dots represents the finished waveform but we have an added constraint:
4: The finished reconstructed waveform must not possess any content (such as harmonics) greater than or equal to half the sampling frequency as would be revealed if that waveform were re-decomposed in accordance with Fourier's theorem.
If we did not have rule 4, then more than one curve could be found that went through all the samples and we would have violated rule 3.
Thus I suppose (I am not a Fourier and Nyquist expert) that the one and only curve that obeys rule 4 and connects all the dots is the original waveform.
The angular waveform in the middle, using what we explained earlier, clearly violates rule 4 due to the harmonic content. We can't see from the small diagrams above but if we use the rightmost diagram to represent half the sampling frequency itself, it as a finished reconstructed waveform also does not satisfy rule 4; the highest frequency allowed in the original material is an infinitesimal tad less.
--------------------------------------------------------------------------------
Lookahead
In reconstructing the original analog waveform (and also for manual curve fitting using French curve templates), the process can't take just one dot at a time but instead must size up several upcoming dots (looking ahead) at all times. For example we may work on samples 1-8, then work on samples 2-9, then 3-10, and so on. The closer any of the actual frequency content is to the sampling frequency, the more lookahead is needed (I don't know the formula for how much). Insufficient lookahead means that the reconstructed waveform might come out different from the original, for example the original signal on the left in the above diagram might still be reconstructed as the waveform on the right. The lesser the amount of lookahead, the greater the chance that the reconstruction logic runs into a situation where it cannot "connect the dots" and still conform with Rule 4. What usually happens is that the reconstruction logic inserts a discontinuity (a sudden bend that violates Rule 4) and then keeps on going.
--------------------------------------------------------------------------------
Output Oversampling
Some D/A conversion techniques use digital processing where several graduated steps are used to shape the output waveform between samples. Were it not for these graduated steps, that process could not do better than an ordinary "connect the dots" process that violates Rule 4. This process with graduated intermediate steps is referred to as output oversampling.
--------------------------------------------------------------------------------
Output Low Pass Filtering
Even though it may make an excellent attempt to conform with rule 4, the D/A conversion may still produce an output waveform with spurious high frequency content. Jaggies or stairstepping or squared off peaks or peaks of the right width but too sharp seen if the waveform is graphed are examples of such "noise". For example the process using output oversampling as described immediately above will still have tiny jaggies in the waveform emerging from the D/A converter. The process is still considered successful when an analog low pass filter follows and the output after that conforms with rule 4.
Of course the analog waveform is reconstructed from the samples using mathematical formulas, not pure trial and error.
--------------------------------------------------------------------------------
Noise, Accurate Sampling, and Accurate Filtering
To conform with Rule 2, the original material must not possess any harmonic content in excess of (or equal to) half the sampling frequency. It is necessary to low pass filter the input material to prevent violation of this rule. Should noise entering the system after this filtering represent higher frequency content, the samples can be corrupted and at the other end, the D/A conversion, the finished waveform that obeys Rules 3 and 4 above will not be correct.
It is common in audio applications for sample values to be accurate to 16 bits, or one part in 65536, or even just 8 bits or one part in 256. When the content is very close to half the sampling frequency, slight inaccuracies in the samples can result in major inaccuracies in the finished analog waveform. There could even be large peaks in the output that were not present in the original material.
The closer the content comes to half the sampling frequency, the more complicated the D/A conversion calculations must be. Practically speaking, this means more expensive circuitry. And the more correct the D/A conversion is, the more likely and more profound artifacts from noise and inaccurate sampling will be with respect to the highest frequency content.
Noise, sampling inaccuracy, and D/A conversion accuracy all interact to govern the quality of the output. By choosing a higher sampling frequency relative to the highest frequency in the content, inaccuracies in sampling and less than perfect D/A conversion become less important.
Now back to the world of video, and the bad news.
5. Digital to analog converters, like comb filters and line doublers, come in different levels of quality. Nowadays every DVD player (and every line doubler) needs a D/A converter to get the video back into analog form to send down a component video or S-video cable. The ultra simple D/A converters merely connect the dots i.e. look ahead zero pixels beyond the pixel being worked on..
6. The video source material may contain details finer than the pixel spacing. In mathematical, Fourier, and Nyquist terms, that means the material being sampled contains frequencies in excess of half the sampling frequency. When this happens, all bets are off regarding the ability to recreate the input from the samples. Incidentally, recording the video as analog at first and low pass filtering it will remove all details that are too fine to sample.
7. Video samples typically have an accuracy of no better than one part in 256. (The luminance or dark/light value for each pixel on DVD is an 8 bit number.) More lookahead is then needed to properly reconstruct the highest frequencies, and if the high frequency content is only brief (fine detail for a short horizontal distance across the screen) accurate reconstruction is not possible.
8. As part of the MPEG compression used to make the entire movie fit on the disk, the samples (one sample equals one pixel) may be manipulated or substituted with similar but not identical samples from another field or frame. Then turning them back into the analog waveform, even with perfect Fourier formulas, can no longer produce the original waveform. However if simple connect the dots D/A conversion were used, the difference between the incorrect results obtained from manipulated and compressed data and the incorrect results obtained from unmanipulated and uncompressed data is less profound.
--------------------------------------------------------------------------------
Another "Connect the Dots" Example
Let's imagine a picket fence subject where thirteen pixels corresponded to six pickets and six spaces between pickets. What we see is portions of the fence with pickets appearing slightly closer together than they really are (using the pixel spacing, not the picket spacing) and other portions of the fence blurred out where the pixels and pickets were out of phase. A correct D/A conversion would show the fence with all the pickets nice and crisp at their original pitch (the video signal is analog when it gets to the picture tube).
Here is a diagram showing details going in and out of phase with the pixels. Actually the nature of analog video is such that, near the resolution limit, the reproduced bars will never be as crisp as the top row in the diagram below. The reproduction at the extreme lower left and lower right is excellent. The purpose of the diagram is to show the profound loss of detail in some places due to pixel straddling.
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The good news: Connect the dots D/A conversion doesn't look too bad
Nobody counts the pickets in a fence when watching a movie.
If details were lost to pixel straddling in one video frame, chances are they will show up in the next frame in normal motion.
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Foldover Aliasing
What happens, if we may ask, if we sample material that does contain content greater than half the sampling frequency?
It so happens that if the highest frequency that is supposed to be present is less than X hertz (we sample at twice X hertz), if the source material happened to contain X+Y hertz, the samples obtained would be the same as if the source material contained X-Y hertz. For example we sample some audio at 10 KHz. (For telephones, we don't need frequencies greater than 5 KHz.). If per chance there was some 6 KHz content, we recreate the analog audio with 4 KHz (5 minus 1 KHz) content instead of the original 6 KHz (5 plus 1 KHz) in accordance with rule 4 above. For audio, this produces sour notes. So for audio, it is necessary to do low pass filtering of the original source material prior to A/D conversion.
The phenomenon of frequencies that are more than half the sampling frequency coming back as frequencies that are less than half the sampling frequency is referred to as foldover. It is a kind of aliasing in that the result after conversion back to analog looks like something else, namely what could have been another original source with such lower frequencies already present.
We won't explain thoroughly here the reason for wagon wheels seemingly rotating in reverse as seen in the movies which is also a situation where the sampling frequency is too low. But as a hint we will state that the sampling frequency is the frames per second rate and the frequency we are trying to reproduce is the rate at which spokes pass the "6 o'clock position" on the wheel.
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A De-Rating Factor for Digital Video
In this writer's opinion, we need to take three samples for each light and dark detail pair (line pair; waveform cycle) for still pictures such as from a computer scanner. Thus a photograph that supposedly has at most 200 dots per inch of resolution would need to be scanned at 300 DPI. In video the subject is usually in motion so details that are straddled in one video frame would likely be clear in the next video frame. The Kell factor represents the ratio between lines of resolution and pixels (spanning the same distance used for reference). It is subjective, and is said to be about 0.7 for still pictures and early (1950's) monochrome video, and about 0.85 to 0.9 for full motion video."
evm1024
New member
Macro, micro and below
The state of art for digital has come far.
Macro and micro artifacts can be classified as acute which is to say that for those who perceive them they are observable at a conscious level. They can be said to exist or not exist at that moment of listening.
We find the state of the digital art at a point where it sounds quite good. And yet there is a fly in the ointment. For some percentage of the population there exists a chronic problem with digital audio.
Chronic in used (as is acute) in the medical sense. These chronic artifacts are not detected as a perception (I hear it now, now I don't) but show up over some period of listening. Something is there (or not there) that does not sound quite right but you can not put your finger on it. We here it spoke of off handily as "Digital Sucks" Which is a gross overstatement and tends to result in fights. It shows up in the comments of people who hear their band recorded to tape for the first time. And it shows up here in Analog Only by those who have discovered tape decks....
We can speculate as to what the cause of this is. We must realize that it is not an acute thing. You are not going to see it in your distortion meter or in an FFT plot.
Some on the digital side have stated that it does not exist or that it is a rejection of the clarity and preciseness of a digital recording. I think we can dismiss that thought out of hand. Others have stated that artifacts that are 50 or 60 dB down can be ignored. To them I suggest that they clap their hands in their studio and listen to the 50 and 60 dB down echoes that tell you everything your ears can tell you about the room you are in.
Regards --Ethan
The state of art for digital has come far.
Macro and micro artifacts can be classified as acute which is to say that for those who perceive them they are observable at a conscious level. They can be said to exist or not exist at that moment of listening.
We find the state of the digital art at a point where it sounds quite good. And yet there is a fly in the ointment. For some percentage of the population there exists a chronic problem with digital audio.
Chronic in used (as is acute) in the medical sense. These chronic artifacts are not detected as a perception (I hear it now, now I don't) but show up over some period of listening. Something is there (or not there) that does not sound quite right but you can not put your finger on it. We here it spoke of off handily as "Digital Sucks" Which is a gross overstatement and tends to result in fights. It shows up in the comments of people who hear their band recorded to tape for the first time. And it shows up here in Analog Only by those who have discovered tape decks....
We can speculate as to what the cause of this is. We must realize that it is not an acute thing. You are not going to see it in your distortion meter or in an FFT plot.
Some on the digital side have stated that it does not exist or that it is a rejection of the clarity and preciseness of a digital recording. I think we can dismiss that thought out of hand. Others have stated that artifacts that are 50 or 60 dB down can be ignored. To them I suggest that they clap their hands in their studio and listen to the 50 and 60 dB down echoes that tell you everything your ears can tell you about the room you are in.
Regards --Ethan
ausrock
Well-known member
Earlier today I was talking on the phone (for the first time) to a guy some 800 miles away who I'm acquainted with from another forum and we briefly touched upon the "analogue/digital" subject and as someone who has also regressed to tape, he said exactly what my thoughts are..........there is an audible difference between the two mediums, it is something that is almost indefinable BUT it is there and it does favour the older technology.
cjacek
Analogue Enthusiast
My original statement:
Your reply:
Note I forgot "to" in the above statement.
...but to get back to your question / issue about my post.... After having reread it, I probably should have clarified my use of the term 'incomplete'.... All I meant is that the content is sampled vs linear, as would be with analog audio, which I think speaks for itself. That's all. I don't know how to explain it better. Actually lemme try this... With digital, you get a series of snapshots [like on a camera] but with analog, it's one big solid line of capture. So while the frequencies are there for both formats, the analog doesn't throw anything away in its effective frequency range. But that's not the only [nor more important] issue here [speaking of frequency response and such] but rather that the 'content' is sampled vs linear and that makes a huge difference.
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Your reply:
Note I forgot "to" in the above statement.
...but to get back to your question / issue about my post.... After having reread it, I probably should have clarified my use of the term 'incomplete'.... All I meant is that the content is sampled vs linear, as would be with analog audio, which I think speaks for itself. That's all. I don't know how to explain it better. Actually lemme try this... With digital, you get a series of snapshots [like on a camera] but with analog, it's one big solid line of capture. So while the frequencies are there for both formats, the analog doesn't throw anything away in its effective frequency range. But that's not the only [nor more important] issue here [speaking of frequency response and such] but rather that the 'content' is sampled vs linear and that makes a huge difference.
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