Home |
Search |
Today's Posts |
#121
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Roger W. Norman wrote:
"Kurt Albershardt" wrote in message ... Also: due to the way computers store information, 18 bit PCM takes the same amount of disk space as does 24 bit PCM (and often the same amount as does 32 bit floating point PCM data.) Well, my point wasn't that using 24 bit has anything wrong with it, nor, with the cost of storage today, is there a problem with using 24 bit for any project. In fact, the processing aspects of digital certainly require that you use 24 bit, and save it as 32 bit floating point if you can. I was just trying to make it clear that there are technical aspects of 24 bit converters that wouldn't necessarily make them better than 16 bit in given circumstances. And I was merely pointing out one of the reasons we have so many 24-bit converters on the market when their real performance is more like 19-22 bits. |
#122
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Justin Ulysses Morse wrote in
m: Carey Carlan wrote: Carey wrote: Is your argument that this higher precision is inaudible? and your answer is "Yes". That's all I needed to know. What? Where do you get that? I've written a novel or two explaining why it's theoretically audible, even though in practice it will be covered up by noise most of the time. To say my answer is "yes" is to miss the point which I have over-articulated here. I didn't miss the point. I read your mini-novel. To summarize down to one phrase, your thesis is that I have very little chance of hearing it. That translates to 'no' in my book. |
#123
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Carey Carlan wrote:
I didn't miss the point. I read your mini-novel. To summarize down to one phrase, your thesis is that I have very little chance of hearing it. That translates to 'no' in my book. Okay. Upon further consideration, "it's audible" isn't really what I was getting at anyway. Not that I would know. My intuitive understanding of the basic theory surpassed my listening experience. Not that I know all that much about the theory either. Just the basics. ulysses |
#124
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , "Arny Krueger"
wrote: makes pretty good use of it. However a converter with a relatively coarse step size is still going to be noisier than one that has a smaller step size. This is the simplest way to state the advantage of greater bit depths - less noisy. Dither is essential in a quantizer in audio and can be thought of as levelling the playing field (to borrow another poster's phrase) above the nosie floor of the system. However, to pick nits, it is true that the error is not completely "cured" (again, to borrow a poster's term). In practical terms, however, it's just turned into a benign noise floor. The resolution of the audio in the available dynamic range in terms of voltages being reproduced by the system is continuously infinitely variable and not stepped with distortion and aliasing. So yes, the posters who have been asking the questions appear to basically have a grasp on it now, though there are always little caveats and distinctions. However, don't let the little technical nit picking confuse the practical understanding you have gained. Digest the conceptual big picture for a moment, and then you can suffer through the distinction of why the extra steps and their gained precision only lower the noise in a practical sense, even though the numbers themselves are certainly more precise. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#125
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , "Tommi"
wrote: If it is so, then it..umm..is so. So this is what happens in the real world, but in theory 24 bits represents the original signal more accurately than 16 bits. "24 bits is more accurate" is actually true as a general statement, but what that means specifically, and how that manifests itself in an audio system has been the source of confusion. I hope that's the distinction that has gotten accross in this discussion. Thanks to everybody for participating. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#126
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Thanks to everybody for participating.
I didn't participate by posting but I enjoyed reading the threads on this subject. It's good to keep learning. --------------------------------------- "I know enough to know I don't know enough" |
#127
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
I would also like to express my appreciation. I can't guarantee that I
understand everything that was posted here at this point, but I plan on rereading it at my leisure. My point is that hanging here is like getting a graduate level course in engineering for free. I can't tell you how much I appreciate everyone who takes the time to eloquently share their knowledge. |
#128
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Chris Hornbeck
writes: The process of quantizing has fundamental errors estimating the smallest bit, errors which track the signal itself. Is this in any way analogous to how a compressor with attack and release set fast enough can track the waveform of a bass signal and cause audible "distortion"? Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#129
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Jay -
atldigi writes: "24 bits is more accurate" is actually true as a general statement, but what that means specifically, and how that manifests itself in an audio system has been the source of confusion. I hope that's the distinction that has gotten accross in this discussion. It has. Thank you guys for being patient and probably repeating yourselves ad nauseum. Its sinking in. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#130
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , "Arny Krueger"
writes: Dither clearly doesn't correct quantization error. Dither does change quantization error into something that is sonically more benign. Allow me to hazard a guess as to how dither works: First of all, do I understand correctly that dither is used whether or not a file is being altered in some way or not? In other words, even in a first generation digital recording dither is already in use right? If the answer to that question is no then I am really lost. Anyway, if we have two adjacent samples of different voltages which are both being rounded, does the dither sort of randomly insert other near voltages around the two samples and in doing so smooth the differences between the two samples' voltages? e.g: if we had one sample with a value of 1.736400 volts and following sample with 1.736500 volts might the dither insert a voltage of 1.736450 (a voltage somewhere between the other two)? I realize those are not 16 bit numbers but I dont think it matters for this question. Of course it has now occurred to me that I dont know where this dither signal would be placed since there is no intermediate sample in which to put it. Eeewww buoy! Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#131
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Garthrr wrote:
In article , "Arny Krueger" writes: Dither clearly doesn't correct quantization error. Dither does change quantization error into something that is sonically more benign. Allow me to hazard a guess as to how dither works..... This topic has come up quite a few times here over the past few years - a Google search will provide you with a wealth of material. For example, there was a very informative thread on this topic that ran through the first half of December two years ago. Here's one article that I find to be very helpful in explaining exactly how dither works: On Dither by Mithat Konar Of the all the things that are commonly misunderstood about digital audio, one of the most common is dither. Dither is the deliberate introduction of a random signal (i.e., noise) prior to quantization in an analog-to-digital converter or prior to reducing the wordlength (e.g., 24 to 16 bits) in a DSP algorithm. Dither comes in various forms and in varying levels levels of complexity, but all forms have in common the introduction of a random signal into the signal chain. It might seem like a stupid idea to add noise to a system, but when dither is done correctly, it completely eliminates the low-level distortion found in quantized systems while increasing the resolution of the system to a level well below the noise floor. (In theory, the resolution becomes infinite). In these respects, dither makes digital systems behave exactly like analog ones. The primary consequence of adding dither is a slight decrease in signal-to-noise ratio -- typically a few decibels. However, advanced schemes add the noise in such a way that its audibility is minimized. The end result with 16-bit, 44.1 kHz systems is a digital signal with all the benefits of dither (elimination low-level distortion and an increase in resolution) but virtually no perceived increase in noise. It is often stated that dither simply masks, not eliminates, the low-level errors found in digital systems, that the errors are still present, but they are simply buried in hiss. This is very much not the case. The following thought experiments may help to put these issues into intuitive perspective. (1) Imagine you have a bipolar, DC coupled analog-to-digital converter with a least significant bit (LSB) threshold of 1 volt. (Just go with it -- this is a thought experiment!) Furthermore, assume the ADC stairstep function is aligned such that it is symmetric about zero. In other words, an ADC input over the 1-volt interval [-0.5, 0.5] produces an ADC output of 0, an input over the range [0.5, 1.5] produces an output of 1, and so on. (2) Without applying any dither, apply a 0.25 volt signal to the input of the ADC. The output of the ADC will be a string of zeros. In fact, any signal between -0.5 and 0.5 volt will result in an ADC output of zero. Any information below the LSB threshold is completely lost. (3) Remove the 0.25 volt signal and apply dither to the input of the ADC in the form of a completely random signal (i.e., noise) centered around 0 volts and large enough to just barely twiddle the LSBs of the ADC. The output of the ADC will be a stream of very small random numbers. However, the AVERAGE of all these values will be zero. (4) Now let's apply our 0.25 volt signal again (with the dither on). The output stream will again look like a stream of very small random numbers, but guess what? The AVERAGE of all those numbers will now be...you guessed it, 0.25. We have thus retained the information that was previously lost (even though it's buried in "noise"). In other words, our system's resolution has improved. The conversion is still essentially random, but the presence of the 0.25 volt signal biases the randomness. The mathematician would say that the characterization of the system with dither is transformed from a completely deterministic one to a statistical one. (5) With the dither on, we can now change the input signal over a continuous range and the average of the ADC output will track it perfectly. An input signal of 0.373476 volts will have an average ADC output of 0.373476. The same will hold true of inputs going over the +/- 0.5 volt LSB threshold: e.g., an input of 3.22278 will have an average ADC output of 3.22278. So not only has the dither enhanced the resolution of the system, but it has also eliminated quantization effects! (6) The results in (4) and (5) will not happen by adding noise after the A/D conversion. Go back to our first experiment and add the random signal to the output of the ADC. As long as the input signal is between -0.5 and 0.5 volt, the ADC's average output will still be zero. Between 0.5 and 1.5 volts the average will jump to 1. There is no resolution enhancement and the quantization effects remain. So, you can see that dither's resolution enhancement and error elimination are truly physical/mathematical in nature and not a masking trick. You should also keep in mind that human beings are able to hear signals in the presence of noise of greater energy than the signal, i.e., with negative signal-to-noise ratios. Therefore, even though a given signal is below the noise-floor of the system, and therefore you might think it irrelevant, it in fact may not be. Depending on what else is going on around that signal, it can be audible at several decibels below the noise-floor. Therefore, both benefits of dither -- eliminating low-level quantization errors and increasing the system's resolution -- are truly beneficial, perceivable effects. Source URL: http://www.birotechnology.com/articles/dither.html |
#132
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
|
#133
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
|
#134
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Arny Krueger" wrote in message
... I just didn't want any lurkers to think that only 24 bit converters are a solution to any recording problems they might have. Again, it's the use of the tools rather than the tools themselves. I think that this is one of the most important messages that a group like this has to present to newbies. Maybe we should preface every post with it's not the fool's tool, it's the tool's fool. g -- Roger W. Norman SirMusic Studio Purchase your copy of the Fifth of RAP CD set at www.recaudiopro.net. See how far $20 really goes. "Roger W. Norman" wrote in message "Kurt Albershardt" wrote in message ... Also: due to the way computers store information, 18 bit PCM takes the same amount of disk space as does 24 bit PCM (and often the same amount as does 32 bit floating point PCM data.) Well, my point wasn't that using 24 bit has anything wrong with it, nor, with the cost of storage today, is there a problem with using 24 bit for any project. In fact, the processing aspects of digital certainly require that you use 24 bit, and save it as 32 bit floating point if you can. I was just trying to make it clear that there are technical aspects of 24 bit converters that wouldn't necessarily make them better than 16 bit in given circumstances. Pretty much the concept of choosing one's tools to fit the job. Agreed. One thing to realize is that most if not all of the major chip makers have unflinchingly released 24/96 and 24/192 converters with worse measured dynamic range than some of their earlier 16/44 converters. There's a part of the market that is all about numbers. When I go out and do location recordings, I still use Tascam DA38s and I've not had one client that wasn't happy with the recordings. They might not have liked the performances, but that's a different story. My last submission to A Fifth of RAP was recorded onto DA38, and it sounds pretty good to me (obviously biased) with plenty of dynamics. The room lacked something and 24 bits wouldn't necessarily have bought me anything more. The same with Scott Dorsey's extremely dynamic recording. And again, Tonebarge absolute knocks one out of the park with his Mackie/Adat combination (although he uses different pres for tracking). |
#136
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Dither by Mithat Konar (6) The results in (4) and (5) will not happen by adding noise after the A/D conversion. Go back to our first experiment and add the random signal to the output of the ADC. As long as the input signal is between -0.5 and 0.5 volt, the ADC's average output will still be zero. Between 0.5 and 1.5 volts the average will jump to 1. There is no resolution enhancement and the quantization effects remain. I'm probably just reading this wrong, but on it's face, this is incorrect. Digital dither is used all the time. Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#137
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
(Garthrr) wrote in message
Dither clearly doesn't correct quantization error. Dither does change quantization error into something that is sonically more benign. Of course it has now occurred to me that I dont know where this dither signal would be placed since there is no intermediate sample in which to put it. Eeewww buoy! Garth~ Lip****z and Vanderkooey from Toranto university have a lot of articals on the dither subject. As academics they use a lot of math, which may be difficult reading for some. I wrote an artical for folks that do not want the math but wish to get some common sense understanding. Included are plots, both time and frequncy domain. It is under the suport section of www.lavryengineering.com and the artical name is "Do you need 20 Bits". This was written in 1997. Dither was and still is importent for say 16 bits work. Of course, if 24 bit format ever becomes a serious release standard, dither becomes that "gold comb to a bold head"... Dither is cool, but it is a bit of a tradeoff between dynamic range and distortions. Noise shaped dither provides some controll over where the noise gets added (obviously away from the hearing sensitive region of say "mid KHZ"). This is is much more advanced. If you want to see that, go to the same place (suport) and check Acostic Bit Correction. Again, if 24 bits becomes a standard, it all becomes just history... BR Dan Lavry |
#138
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Chris Hornbeck wrote:
On Dither by Mithat Konar (6) The results in (4) and (5) will not happen by adding noise after the A/D conversion. Go back to our first experiment and add the random signal to the output of the ADC. As long as the input signal is between -0.5 and 0.5 volt, the ADC's average output will still be zero. Between 0.5 and 1.5 volts the average will jump to 1. There is no resolution enhancement and the quantization effects remain. I'm probably just reading this wrong, but on it's face, this is incorrect. Digital dither is used all the time. Dither is used when reducing the resolution of the signal. It is effective only when it is applied before the resolution is reduced, not after. His example uses an imaginary ADC as its basis, but it would be equally valid if he were talking about reducing word length in the digital domain. So, yes, I think you _are_ reading this wrong. His only point is that dither is NOT effective if it is applied after the resolution has been reduced (ie, after truncation). The process of digitizing an analog signal is a process in which resolution is reduced. So dither is required there, just as it is when reducing the bit depth of a digital signal. The basic principle is exactly the same in either case. |
#139
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Mike Rivers wrote:
I haven't given this any thought, so I'll just throw out the question to ponder. Is it possible that time resolution between components of a complex wave could be better with a higher sampler rate? Could the phase relationship between the fundamental and, say, third harmonic of a distorted guitar, be more accurately preserved at a higher sample rate? If my understanding is correct, and it's based mostly on what I read here from I think Arny (but it makes perfect sense to me), then the answer to your question is that yes, a higher sample rate (or a higher bit rate for that matter) would improve that time resolution; but that the resolution is already WAY higher than necessary. People assume that the time-domain accuracy is one sample period; but it's really one sample period divided by the quantization range. So the time domain accuracy for CD audio would be 1/(44100*2^16) or 346 picoseconds. That's about a decimal place or two off of what I vaguely remember Arny saying, so I've probably messed it up. But you get the idea? I probably shouldn't be paraphrasing from memory, so go back and read Arny's post from yesterday in one of these threads. ulysses |
#140
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Chris Hornbeck wrote:
On Dither by Mithat Konar (6) The results in (4) and (5) will not happen by adding noise after the A/D conversion. Go back to our first experiment and add the random signal to the output of the ADC. As long as the input signal is between -0.5 and 0.5 volt, the ADC's average output will still be zero. Between 0.5 and 1.5 volts the average will jump to 1. There is no resolution enhancement and the quantization effects remain. I'm probably just reading this wrong, but on it's face, this is incorrect. Digital dither is used all the time. I know Mithat, and he knows what he's talking about. If he didn't, he wouldn't have written it. He's not saying that dither can't be applied in the digital domain. He's saying it needs to be applied prior to the quantization (or re-quantization) that's being dithered. In an ADC, this could occur in the analog domain, or in the case of an oversampling converter, it could be done digitally before the signal is quantized to 16 or however many bits. In the case of truncating 24 bit data into 16 bits, he's saying you have to add the dither BEFORE you lop off those bottom 8 bits. The dither and the LSBs have to be summed before truncation. ulysses |
#141
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Fri, 21 Nov 2003 13:42:15 -0600, Justin Ulysses Morse
wrote: I know Mithat, and he knows what he's talking about. If he didn't, he wouldn't have written it. He's not saying that dither can't be applied in the digital domain. He's saying it needs to be applied prior to the quantization (or re-quantization) that's being dithered. In an ADC, this could occur in the analog domain, or in the case of an oversampling converter, it could be done digitally before the signal is quantized to 16 or however many bits. In the case of truncating 24 bit data into 16 bits, he's saying you have to add the dither BEFORE you lop off those bottom 8 bits. The dither and the LSBs have to be summed before truncation. Hi Ulysses, Thanks for your comments. I really don't know how to resolve this. What you say makes perfect sense, but Watkinson says otherwise. Maybe someone else could help out?..... Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#142
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Chris Hornbeck wrote:
On Fri, 21 Nov 2003 13:42:15 -0600, Justin Ulysses Morse wrote: I know Mithat, and he knows what he's talking about. If he didn't, he wouldn't have written it. He's not saying that dither can't be applied in the digital domain. He's saying it needs to be applied prior to the quantization (or re-quantization) that's being dithered. In an ADC, this could occur in the analog domain, or in the case of an oversampling converter, it could be done digitally before the signal is quantized to 16 or however many bits. In the case of truncating 24 bit data into 16 bits, he's saying you have to add the dither BEFORE you lop off those bottom 8 bits. The dither and the LSBs have to be summed before truncation. Hi Ulysses, Thanks for your comments. I really don't know how to resolve this. What you say makes perfect sense, but Watkinson says otherwise. Maybe someone else could help out?..... OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote above was pretty well established and non-controversial, and furthermore that you (Chris) were already in agreement with it (based upon your post back to me just a few minutes earlier). So what exactly is contradictory here? What does Watkinson say that seems to counter the above? |
#143
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Fri, 21 Nov 2003 20:40:02 GMT, Jim Gilliland
wrote: OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote above was pretty well established and non-controversial, and furthermore that you (Chris) were already in agreement with it (based upon your post back to me just a few minutes earlier). So what exactly is contradictory here? What does Watkinson say that seems to counter the above? Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry, don't have The Art of Digital Audio handy): " the principles of analog and digital dither are identical; the processes simply take place in different domains using two's complement numbers which are rounded or voltages which are quantized as appropriate. In fact quantization of an analog dithered waveform is identical to the hypothetical case of rounding after bipolar digital dither where the number of bits to be removed is infinite, and remains identical for practical purposes when as few as 8 bits are to be removed. Analog dither may actually be generated from bipolar digital dither (which is no more than random numbers with certain properties) using a DAC." Hopefully, I'll be shown to have misunderstood the above, and can go back to my comfortable but fuzzy understanding of things before this blasted thread started. Ignorance was bliss, and I'm too old to study. Thanks! Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#144
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: Mike Rivers wrote: I haven't given this any thought, so I'll just throw out the question to ponder. Is it possible that time resolution between components of a complex wave could be better with a higher sampler rate? Could the phase relationship between the fundamental and, say, third harmonic of a distorted guitar, be more accurately preserved at a higher sample rate? If my understanding is correct, and it's based mostly on what I read here from I think Arny (but it makes perfect sense to me), then the answer to your question is that yes, a higher sample rate (or a higher bit rate for that matter) would improve that time resolution; but that the resolution is already WAY higher than necessary. People assume that the time-domain accuracy is one sample period; but it's really one sample period divided by the quantization range. So the time domain accuracy for CD audio would be 1/(44100*2^16) or 346 picoseconds. That's about a decimal place or two off of what I vaguely remember Arny saying, so I've probably messed it up. But you get the idea? I probably shouldn't be paraphrasing from memory, so go back and read Arny's post from yesterday in one of these threads. ulysses And you wouldn't believe it, but dither actually affects the time domain as well. No kidding. Little known, but true - it's similarly helpful in the time domain as it is for amplitude. Most of the science geeks say that time resolution also becomes essentially infinite with proper dither. So Mike's concern is probably something that doesn't need to be worried about as some say the limit doesn't exist in a well designed system, and even the competing camp says it's so small as to not make a difference. In this case it doesn't matter who's right since it just plain doesn't seem to matter. There's a few stragglers left over that say it's possible that imaging can be affected when you have multichannel in play, but even the significance of this deserves some skepticism as it is far from proven or agreed upon, and the "infinite time resolution in a properly implimented system" guys actually seem to have the science to back it up. My vote is with them. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#145
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: He's not saying that dither can't be applied in the digital domain. He's saying it needs to be applied prior to the quantization (or re-quantization) that's being dithered. In an ADC, this could occur in the analog domain, or in the case of an oversampling converter, it could be done digitally before the signal is quantized to 16 or however many bits. In the case of truncating 24 bit data into 16 bits, he's saying you have to add the dither BEFORE you lop off those bottom 8 bits. The dither and the LSBs have to be summed before truncation. Yes. If you do it after, the distortion is there, and possibly (probably) aliasing from it, and there's no way to get rid of it once it's there. However the designer wants to do it for the aplication at hand (digital or analog), it needs to be done before the quantization or requantization. In article , Chris Hornbeck wrote: Hi Ulysses, Thanks for your comments. I really don't know how to resolve this. What you say makes perfect sense, but Watkinson says otherwise. Maybe someone else could help out?..... I think this has already been reolveed in other posts in the thread, so I'll just add a small Watkinson quote: "The introduction of dither PRIOR to a conventional quantizer inevitibly causes a slight reduction in the signal-to-noise ratio attainable, but this reduction is a small price to pay for the elimination of non-linearities." So obviously Watkinson does say it must happen before quantization. Your Watkinson quote was talking about something a little different. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#146
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Chris Hornbeck wrote:
On Fri, 21 Nov 2003 20:40:02 GMT, Jim Gilliland wrote: OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote above was pretty well established and non-controversial, and furthermore that you (Chris) were already in agreement with it (based upon your post back to me just a few minutes earlier). So what exactly is contradictory here? What does Watkinson say that seems to counter the above? Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry, don't have The Art of Digital Audio handy): " the principles of analog and digital dither are identical; the processes simply take place in different domains using two's complement numbers which are rounded or voltages which are quantized as appropriate. In fact quantization of an analog dithered waveform is identical to the hypothetical case of rounding after bipolar digital dither where the number of bits to be removed is infinite, and remains identical for practical purposes when as few as 8 bits are to be removed. Analog dither may actually be generated from bipolar digital dither (which is no more than random numbers with certain properties) using a DAC." Hopefully, I'll be shown to have misunderstood the above, and can go back to my comfortable but fuzzy understanding of things before this blasted thread started. Ignorance was bliss, and I'm too old to study. OK, all of your quote makes sense to me. What I don't see, though, is how anything in Ulysses's quote contradicts it. Here is what you quoted from Ulysses earlier - can you help me see where the conflict is? I apologize if I'm being dense or stupid here. I know Mithat, and he knows what he's talking about. If he didn't, he wouldn't have written it. He's not saying that dither can't be applied in the digital domain. He's saying it needs to be applied prior to the quantization (or re-quantization) that's being dithered. In an ADC, this could occur in the analog domain, or in the case of an oversampling converter, it could be done digitally before the signal is quantized to 16 or however many bits. In the case of truncating 24 bit data into 16 bits, he's saying you have to add the dither BEFORE you lop off those bottom 8 bits. The dither and the LSBs have to be summed before truncation. |
#147
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
He's talking about the input (A/D), basically saying that dither must be
applied to the input of (ie, before) a conversion. You're thinking of bit depth reduction or output (D/A) dithering, where the same is true so it's done in the digital domain; before the conversion. Similarly, adding analog noise after a D/A conversion would not have the desired effect. Chris Hornbeck wrote: On Dither by Mithat Konar (6) The results in (4) and (5) will not happen by adding noise after the A/D conversion. Go back to our first experiment and add the random signal to the output of the ADC. As long as the input signal is between -0.5 and 0.5 volt, the ADC's average output will still be zero. Between 0.5 and 1.5 volts the average will jump to 1. There is no resolution enhancement and the quantization effects remain. I'm probably just reading this wrong, but on it's face, this is incorrect. Digital dither is used all the time. Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#148
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Jay - atldigi wrote:
There's a few stragglers left over that say it's possible that imaging can be affected when you have multichannel in play.... I bet they read Stereophile. g |
#149
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Fri, 21 Nov 2003 22:26:03 GMT, Jim Gilliland
wrote: OK, all of your quote makes sense to me. What I don't see, though, is how anything in Ulysses's quote contradicts it. Here is what you quoted from Ulysses earlier - can you help me see where the conflict is? I apologize if I'm being dense or stupid here. It's certainly not you being dense here. I suspect that I'm reading the word "identical", twice, and drawing unwarranted conclusions that it means interchangable. Since everybody but me is marching out of step, my conclusion that an ADC could be dithered digitally must be wrong. Thanks for your help; also Ulysses, Jay Frigoletto and S O'Neill. Great thread, ladies and germs, Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#150
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Chris Hornbeck wrote:
Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry, don't have The Art of Digital Audio handy): " the principles of analog and digital dither are identical; the processes simply take place in different domains using two's complement numbers which are rounded or voltages which are quantized as appropriate. In fact quantization of an analog dithered waveform is identical to the hypothetical Here's the problem. Assuming an infinite number of bits to be removed, you're right; dither is necessary to truncate to N bits, like 24 or 16. But they're saying that Analog is equivalent to infinite bit depth, which is clearly impossible; an A/D convertor gets you some finite number of bits. After it's done that, dither below its resolution buys nothing. It just stays below that resolution, and rather than adding information, it only adds noise. case of rounding after bipolar digital dither where the number of bits to be removed is infinite, and remains identical for practical purposes when as few as 8 bits are to be removed. Analog dither may actually be generated from bipolar digital dither (which is no more than random numbers with certain properties) using a DAC." Hopefully, I'll be shown to have misunderstood the above, and can go back to my comfortable but fuzzy understanding of things before this blasted thread started. Ignorance was bliss, and I'm too old to study. Thanks! Chris Hornbeck "That is my Theory, and what it is too." Anne Elk |
#151
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Jim Gilliland writes:
Here's one article that I find to be very helpful in explaining exactly how dither works: Thanks Jim! Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#152
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
|
#153
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In general, with audio it is better to sample at high depth and resolution
and then downgrade afterwards than it is to stay true to the final output. Jon I followed all the postings, and there are a lot of fine comments out there. I guess I'll add my 2 bits: First, going back to some of the earlier dither comments: The concept of digital dither was covered pretty well - word length reduction. I would probably magnify the fact that reduction with no dither causes 2 bad things. The first is harmonic distortions, the second is noise modulation. Think for a second of a tiny sine wave, less than 1 quantization peak to peak, residing between 2 quantization levels. It yields nothing. That is and incorect result. Now shift it so it "seats" on a quantization level, going over and under and over and under... by only one bit. That gives you a 1 bit amolitude square wave... This is not a good signal either... This is just a starting point to think about. Again you can see pictures on www.lavryengineering.com (under suport, "Do you need 20 Bits" artical. But I was more interested in analog dither. Years ago, I was experimenting with in. Some of it was around Nyquist frequencies, some was "subtuctive dither" (remove much in the digital domain what you injected in the analog domain). Why analog dither? Well, nothing happens if you feed a signal into a circuit and it does not cvause some digital transitions... Say your circuit generates a different code every 1 millivolt, but the signal is small and is always between 5.1 and 5.7mV... No action. So you need to "force it" to go over and under some threshold (transition) levels. So we add noise. It does get pretty complicated in a hurry. First knee jerk reaction is to add all the noise in high frequncey. It is not that simple, because after truncation, a lot of the high frequncey reapears in the audio range. But you "have to do somthing"- add some random noise, thus trade off some noise (dynamic range). For word reduction it is done to fix the distortions and noise modulation problems. On the analog side, it was about "getting anything to move" (and more than that). So the question beacme, was is the best tradeoff? Least energy added for fixing the problem? That was when triangle dither (2 least significant bits peak to peak) solution came about. But than there was a period where analog dither died (at least for me). The quantization levels got so close to each other, that the circuit noise was way bigger than I wished for. Take say a 10V range and try for 20 bits - 10uV steps (10 microvolts). Getting the signal in there with such low noie (10 uV peak to peak) is not a walk in the park... You "call" the noise "too much dither" and move farward... But than analog dither came back - BIG TIMES! It started with 1 bit (like DSD) and is now an everyday thing wth the newer multibit ADC's. Those have just very few bits, thus the quantization levels are very far apart. No longer in the microvolts, but in the volts (or near it). This are all noise shaping converters. You feed back a filtered quantized output and subtuct from the input.. This will take too long to explain, but such a raw structure will have all sorts of problems thus it requires dither, and often very large amplitude. The dither randomness tend to "break up" what would otherwise be consistant repetative patterns (called limit cycles), not exactly what you want in the audio... Regarding the comments on dynamic range and bits. Yes it was well said, and about 6dB per bit is right. So 24 bits is about 144dB, and we can all realize it is also a bit of crock. My AD122 MKII has the largest dynamic range (127dB unweighted) so it is 21+ bits. My first generation AD122 had 122dB (20 bits) and I called it a 20 bit converter. Than all those 100-110dB devices apeared on the market with "24 bits" on the panel and the sales guys insisted that I call mine a 24 bits. The bits are there, but the last few just bounce around mindlesly - no realtion to the audio. I have a cheap circuit for andom number generation. If I add 100 bits of noie at the thend (least significant bits) will it be a 124 bits machine? Lets go for it I could probably design a 24 bit AD, and it will be an unbelivable energy hog. Remember that we are on a log scale. The differance between 20 and 24 is not "just 4 more". It is a factor of 16 more. Sort of like a 6 earthquake vs an 8 earthquake. Big differance. Do you need 24 bits AD? Probably not, short of some of the headroom comments. What is the best Mic preamp out there? Say -130dBu? How much gain is it set for? Say I use 30dB gain, than the noise floor is at -100dBu and a peak to peak siganl out of the preampp is 24Bu driving an AD. So we have 100+24=124dB dynamic range. My AD122 MKII gives you 3dB margin. But say you need 40dB mic pre gain. Now you can use the 114dB dynamic range device... I am not a recording engineer, but I think that 127dB is already only for close mics that put out serious signal followed by a great mic pre... While I rather have folks specify dynamic range (not bits), the 24 bit number is pretty good in the sense that it is a multiple of 8 bits (thus 3 bytes). It is a good number for computers and hardware (multiple of 8). I just wish 24 bits did not get used by industry salesman as measure of quality. Last comment for now: Whatever I said regarding 24 bits and what is needed is ONLY for AD's and DA's. Let me call them CONVERSION BITS. There are other type of bits. Let me call them PROCESSING BITS. If 24 conversion bits may be an overkill, 24 processing bits is way too little. Folks need to realize it. A guy comes with some digital EQ and tells you that it has 56 bits, or 32 floating point DSP, and he does have a point to make. Just do not later go and look for a 56 bits AD... Different issue. I think it was already explained, but here we go: Say I want to avarage 50 cent (money) and 51 cent. It is 50.5 cents. I can only deal with cents so we either call it 50 or 51. We have 1/2 cent inacuracy. In a bianry wold, I lost a bit of acuracy. So lets agree to have a new coin - 1/2 cent. Now we avarage 50, 50 and 51. It yields 50.3333333... Well ae can call it 50 or 50.5.... The general statment is that when you add more and more computations, you reduce more and more of the accuracy. So yes, we typicaly need a lot of computational bits. To all, pardon me if soem of the above is a repeat of what you said, I tried to fill some gaps, and probably covered things that needed no help. I am new to audio NG. BR Dan Lavry |
#154
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
|
#156
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Chris Hornbeck" wrote in message
On Thu, 20 Nov 2003 16:04:43 GMT, Carey Carlan wrote: Dithering smooths out the differences between the 65,535 steps, making them as smooth as the 16 million steps. Dither is just noise, but noise has a special property in this case. Although it can't smooth out differences, it can remove errors. Dither doesn't remove the errors, it just makes them more palatable to the ear. Undithered quantization error is among the nastier, more unpredictable forms of distortion around. From http://www.pcabx.com/technical/bits44/index.htm (which has freely downloadable .wav files that are practical examples of how dither changes things) Artifacts of improperly dithered quantization: (1) Background noise level, from dither or quantization noise. Is it steady or does it follow the intensity or tonality of the test tone? (2) Intermittent spurious little whistles or "birdies" due to lack of dither and the steady frequency change in the tone. (3) Loss of low level signal due to lack of dither. (4) Raspy distorted sound due to lack of dither. OTOH, properly dithered quantization error sounds like background noise. We even have some control over what the background noise sounds like. As the signal goes down in amplitude, dither keeps it audible and recognizable even when it is smaller than one quantization step. |
#157
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Justin Ulysses Morse" wrote in message
m Mike Rivers wrote: I haven't given this any thought, so I'll just throw out the question to ponder. Is it possible that time resolution between components of a complex wave could be better with a higher sampler rate? Could the phase relationship between the fundamental and, say, third harmonic of a distorted guitar, be more accurately preserved at a higher sample rate? If my understanding is correct, and it's based mostly on what I read here from I think Arny (but it makes perfect sense to me), then the answer to your question is that yes, a higher sample rate (or a higher bit rate for that matter) would improve that time resolution; but that the resolution is already WAY higher than necessary. People assume that the time-domain accuracy is one sample period; but it's really one sample period divided by the quantization range. So the time domain accuracy for CD audio would be 1/(44100*2^16) or 346 picoseconds. That's about a decimal place or two off of what I vaguely remember Arny saying, so I've probably messed it up. But you get the idea? I probably shouldn't be paraphrasing from memory, so go back and read Arny's post from yesterday in one of these threads. Bascially, you've the concept right. Ditto what Jay said about dither, too. It inifinitizes things in both the amplitude and time domain. |
#158
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
|
#159
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , "Arny Krueger"
wrote: "Chris Hornbeck" wrote: steps. Dither is just noise, but noise has a special property in this case. Although it can't smooth out differences, it can remove errors. Dither doesn't remove the errors, it just makes them more palatable to the ear. It can remove distortion but the error still exists as broadband noise. I think Chris probably knew this but chose the wrong word. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#160
|
|||
|
|||
16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Jay - atldigi" wrote in message
In article , "Arny Krueger" wrote: "Chris Hornbeck" wrote: steps. Dither is just noise, but noise has a special property in this case. Although it can't smooth out differences, it can remove errors. Dither doesn't remove the errors, it just makes them more palatable to the ear. It can remove distortion but the error still exists as broadband noise. Well, we call it noise but in fact it's 100% deterministic given that we created the randomizing signal so we should know what it is. I think Chris probably knew this but chose the wrong word. Could be. But the point needs to be clearly made. IMO, there's a lot of etymological weirdness in this area. Quantization error is often called quantization noise. Spectral shaping of quantization error is commonly called "noise shaping". Quantization error is noisy, but it's noisy in the sense that loud neighbors are *noisy*. It's not noise in the sense of random noise, because quantization error is 100% predictable. I became impressed with how deterministic quantization error can be when I started looking at measurements of a series of samples of the same sound cards and saw how consistent they are. IME pure analog gear is not this consistent. The consistency comes from the fact that the quantizers in the converters were the major source of noise, and the quantizers were all digital and therefore extremely similar. |
Reply |
Thread Tools | |
Display Modes | |
|
|
Similar Threads | ||||
Thread | Forum | |||
Explain me this | Audio Opinions | |||
TS/TRS balanced/unbalanced can someone explain | General | |||
Can you explain this 50Hz hum?? | Pro Audio | |||
Reverb & EQ and "damping" etc .. please explain .. | Pro Audio |