Why Ed Meitner Likes Super Audio - Technical Discussion

Discussion in 'Archived Threads 2001-2004' started by Lee Scoggins, Apr 10, 2002.

  1. Lee Scoggins

    Lee Scoggins Producer

    Aug 30, 2001
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    I thought the technically minded might appreciate some more discussion on the noise shaping in SACD players that DVDA fans are concerned about. From Positive Feedback...

    The DSD/SACD Revolution, Part II:

    PF Interviews Digital Designer Ed Meitner

    Mike Pappas

    (From Positive Feedback, Vol. 8, No. 2)

    Meitner: Yes. Also, the thing is that if you try to archive or re-archive analogue tapes and analogue masters…

    Pappas: Right.

    Meitner: … well, if you have it in the one bit DSD format, you can, (A) you have a pretty robust storage that way, (B) you can now convert it to any other format that may come about, PCM 96/24, whatever. So it’s a very versatile format to begin with. And don’t forget that every A to D converter that you see on the market today starts off as a DSD modulator. So then you have the DSD signal on the A to D that just goes to the PCM down sampler or decimator and gets turned into PCM, so the life of the audio in the digital world really starts off as a one-bit signal.

    Pappas: And so this is just a natural extension of that.

    Meitner: That’s right. Yes. So now if you have an audio chain, you have one bit coming out of the A to D and going into a storage medium, i.e., hard disk or the AIT. And your D to A converter again is relatively simple because you don’t have to do any conversion from PCM to bitstream. And most converters out today are some form of bitstream. So we just cut out the PCM part.

    Pappas: So in effect, DSD is simpler because you’re not doing the kind of conversions using a decimation filter to make it into PCM. You’re just eliminating all that extra stuff.

    Meitner: That’s right. Yes.

    Pappas: So what are the challenges of designing converters for this format?

    Meitner: Really, the way you drive the A to D converter, you have to be careful with your analogue circuitry there. In all this digital audio, analogue is the weak link. So it requires a good deal of attention to the classic analogue problems and analogue issues that have to be dealt with. And you know, other than that, it’s really no big magic.

    Pappas: When you say the analogue is the weak part of the chain, can you explain that a little further for me?

    Meitner: Well, you have an analogue input that goes into the A to D.

    Pappas: Right.

    Meitner: Somehow it needs to be processed. It has to have probably some gain or some gain adjustment. It has to be balanced in, single-ended in. And then the actual ports of the A to D need to be driven with a fairly low impedance. There is some reactive impedance from the A to D input, which means often op amps generally are not happy there. So you need analogue circuitry that can live and do the best to drive the A to D inputs. Once you have that done, of course, then you have the power supply issue and stuff like that.

    On the D to A side you have your digital part; in other words, the bitstream part that comes into the D to A and has to put out analogue. So the same issues are there. You have high frequency in the presence of op amps or other analogue circuitry. So you just have to be very careful. And I think my specialty is that interface part, how to deliver audio to the A to D converter, and how to take it from the D to A converter and present it to the outside world again. And then there are issues of jitter and clock distribution, relatively sensitive issues that over the years I’ve learned to deal with and make them as best as we can today.

    Pappas: Now one of the other issues with DSD is that you’ve got to do fifth-order noise shaping filters to shift the noise that is a result of the conversion process out of the audio band.

    Meitner: Yes.

    Pappas: Can you tell me what kind of challenge that presents from a design standpoint?

    Meitner: It’s what I said before. There is high-frequency noise present in the system. And the analogue parts have to deal with that, which means it has to be low distortion, low noise, and very high speed analogue circuitry that can live without generating any intermod products in the presence of the kind of noise the DSD signal presents.

    Pappas: The other thing is you’ve got a pretty wide input band, somewhere between DC and 100 kilocycles, which has also got to present some challenges.

    Meitner: Oh Yes. Yes. It again comes back to the same thing. The tradeoffs are low noise, low distortion, wide bandwidth and the ability to drive, for instance, reactive loads, like the A to D input for instance. So those are usually tradeoffs that are diametrically opposed.

    Pappas: The wider the bandwidth, generally the more noise you end up with.

    Meitner: That’s right. Yes.

    Pappas: And so when you try to squeeze 120 dB of dynamic range and still maintain a DC to 100 kHz signal, that’s got to be pretty tricky.

    Meitner: Mm-hm.

    Pappas: Does that translate into "expensive"?

    Meitner: Not necessarily.

    Pappas: So you feel that you can build these products for a competitive price compared to a PCM converter?

    Meitner: Oh, absolutely. Because in a PCM converter you have much of the same problems, except you don’t see the high frequency problems directly. But you might still see them as intermod, and if you don’t see them, you hear them as bad sounds. And, you know, what was very helpful for me is that for years I worked with a company called Amber Electro Design and we built distortion analyzers.

    Pappas: Very good distortion analyzers.

    Meitner: Yes, so again I did all the analogue work on that. So, low distortion, low noise, and wide bandwidth is not a strange thing.

    Pappas: The world is working on two different standards here. One end of the spectrum is pushing 96/24 PCM which basically is nothing more than hot-rodded 44.1…

    Meitner: Right.

    Pappas: … and then you guys are coming in from a completely different angle saying, "Forget PCM. There’s a better solution here."

    Meitner: Uh-huh.

    Pappas: I understand why I think it’s a better format. Tell me why you guys think it’s a better format.

    Meitner: To convert audio into PCM is a very alien thing, whereas if you look at the convert audio into one-bit format, it’s a very natural thing. In any form of conversion, you will lose something. You have to choose the format where you lose the least, which means the format that’s the friendliest to audio, which is definitely DSD over PCM.

    Just look at one problem with PCM. Imagine what happens at your zero crossing. You have all those bits flipping. You have, you know, noise shock in the system coming off the power supply if all of a sudden 23 bits change from all zeros to all ones. You have that at every zero crossing. And you need really good error correction, because if a sign bit gets screwed up in the process, all of a sudden instead of your signal being positive, it thinks it’s negative.

    Pappas: That wouldn’t be very good.

    Meitner: No. And the other thing is archiving our recordings, which to me is a very important part, because the old audio tapes are falling apart, and somebody needs to do something with them. If you convert them to PCM now, you’re nailed into that format. You can’t ever get out of it. Whereby, if you do it in the one-bit domain, in the DSD domain, you can convert it, like I said before, to any other format again.

    Pappas: Through the use of…

    Meitner: … through the use of a decimator and down sampler. So we can take the DSD and put it back into 44.1 or 96/24 or whatever comes about.

    Pappas: And not lose as much.

    Meitner: That’s right.

    Pappas: Because the guys who were doing 96/24 and trying to convert it to 44.1 are going to end up tossing something.

    Meitner: Well, yes, that’s a nightmare.

    Pappas: Well, not if you believe their propaganda.

    Meitner: Well, it’s a nightmare. I don’t believe the propaganda.

    Pappas: They’re talking about it like it’s no big deal. And for those of us who know that 96/24 does not equally divide into 44.1, know that that’s not going to —

    Meitner: Yes, but also don’t forget, those are the same people that have said, "Digital is perfect forever."

    Pappas: You guys have got to be taking a pretty big risk on this thing. Let’s…

    Meitner: I don’t think so.

    Pappas: Let’s move to the more practical side. The world currently seems to be moving towards just taking PCM and extending it. There’s a lot of hardware out there, not a lot of it’s particularly good, but there’s hardware. Tell me why I shouldn’t think you’re in a "come-from-behind" mode here.

    Meitner: What do you mean by "come from behind"?

    Pappas: Well, 96/24 in the pro audio business, there are a myriad of vendors providing at least digital audio work stations that run on 96/24.

    Meitner: Yes. Those are also the naysayers because they’ve got all this invested interest.

    Pappas: I believe that DSD is a much better way of doing this and I pretty much feel that us going to 96/24 is like a kid trying to stuff a bigger engine in a Chevette.

    Meitner: That’s right.

    Pappas: You know, you still have all the limitations of the drive train and chassis, it maybe it goes a little faster.

    Meitner: See, I always have to come back to the same thing, it’s that your A to D converter starts off its life — I should say the audio through an A to D converter starts off its life — as a one-bit, or similar to a one-bit signal, and then you muck it up.

    Pappas: And as you refer to that process, PCM is alien.

    Meitner: Yes, it has really nothing to do with audio. You have minimal resolution at zero crossing, whereby with DSD you have maximum resolution and on and on.

    Pappas: So in other words, DSD is pretty much the —

    Meitner: It’s the least conversion of a conversion.

    Pappas: And with PCM, you were mentioning that you know your biggest problems are when you get around the least significant bit, which is at zero crossing; with DSD, that’s where you have your maximum resolution.

    Meitner: That’s right.

    Pappas: That’s very interesting.

    Meitner: You have, in fact, a situation that is very akin to what we hear. We mostly hear velocity changes. Now velocity changes are at maximum at zero crossing of the sine wave. So this is where you have to be so careful. And if you look all through the high-end audio industry, it’s class A, class A, class A. And you know the old solid state zero crossing distortion amplifiers and stuff like that never worked.

    Pappas: Because the ear is more sensitive at that point.

    Meitner: Yes, "maximum velocity" means "maximum intensity," which means the point where we hear the most. So now look at a PCM signal at zero crossing, and all you’ve got at that moment where it crosses zero is you have zero-bit resolution. The only resolution you’ve got is dither.

    Say you have a 16-bit system where one bit is dithered. You now have your noise floor quantized at one bit, and that’s why, when you go to higher resolution tracks or higher resolution PCM converters, people think they are better — and they might be — but the really fundamental reason in my mind why they are better is because you quantize your noise with more bits.

    Pappas: And that exists through the zero crossing issue.

    Meitner: Well, anywhere near zero crossing. Because we know that as your signal level goes down, with every 6dB decrease you lose one bit.

    Pappas: Right.

    Meitner: That’s why when you do digital recordings, you nail this thing to zero dB as much as you can. So let’s look at a typical mike pre-amp that might have, say, an equivalent input noise of -131 or -130 dB, right?

    Pappas: Right.

    Meitner: So then we put, say 40 dB of gain, which is a typical gain, I guess, and we end up now with a noise at -90. So now we have a 16-bit converter. That converter is going to be now dithered from that noise with one or two bits. Right?

    Pappas: Right.

    Meitner: So the resolution of that noise at the bottom is one or two bits. Now, we take a 20-bit converter. All of a sudden we have the same situation, the same noise floor still, except that that noise floor is now quantized with five or six bits, and so on and so on. So I call that "the bits that are dancing."

    Pappas: Dancing bits.

    Meitner: Which is what gets you through the parts where you need resolution and you just don’t have the bits to do it.

    Pappas: And DSD gets around these problems.

    Meitner: Yes. And that’s what I call a friendlier conversion or a lesser conversion or minimal conversion, I should almost say.

    Pappas: Do the same issues about jitter apply to DSD as they would for PCM?

    Meitner: Of course.

    Pappas: Okay, there are no disadvantages or advantages in terms of jitter performance?

    Meitner: No. Jitter is still an important issue. The way I see it, the problem with jitter is that very few people — almost none out there — have an analyzer for it. I think we were about the only ones who, for the past five or six years — from the moment when Stereophile adopted our jitter analyzer — actually measure this stuff. And we measure it in such a way that you don’t only know how much jitter you’ve got, but you also know what your frequency components are in that jitter.

    Pappas: That’s the LIM device?

    Meitner: Yes.

    Pappas: Can you give us a little background on it? Because I think it’s important.

    Meitner: Basically, it is a face demodulator that you can look at any sample frequency within a digital system from 2FS to 768FS. In a 20 kHz bandwidth, it gives you your jitter components and your frequency components in the jitter, which is more important than just knowing what the actual value of the jitter is. Because if it’s just noise, it’s not so bad, but if you have components in there that are alien to audio, well, that’s a different story. What are these "components"? Well, typically, it’s artifacts coming from a CD drive, a deck; you have components in there that are jitter components that come from the focus adjustment, tracking adjustment, motor speed adjustment, and all sorts of other staff that have nothing to do with audio. So this device shows you what those components are.

    Pappas: Very interesting.

    Meitner: You know, when Stereophile adopted that, we agreed and promised that any of our competitors that would want that equipment, we would make it available at a very reasonable cost, which we did.

    Pappas: And the primary advantage of this is not just knowing the number, but knowing what the spectra of the jitter components is?

    Meitner: Precisely.

    Pappas: And that’s a lot more —

    Meitner: Well, then you can find it.

    Pappas: Otherwise you’re just hunting around.

    Meitner: That’s right.

    Pappas: So, let’s talk about the future. It would appear that there are some vendors who are getting on board with DSD in terms of being able to provide things like digital audio work stations and things like that. Is working in the DSD realm easier or tougher for things like building digital mixing desks —

    Meitner: It’s really no different.

    Pappas: Okay.

    Meitner: It’s just re-learned. That’s all. At the Sony demo they had a mixing console that was purely one bit, filters and all.

    Pappas: Everybody’s brain is so accustomed to having PCM represent an absolute value, and there are certainly a fair number of naysayers out there who say that you can’t do it.

    Meitner: Yes, well, there always is. The thing is, there is an obsession with absolutes, totally forgetting that human sensory inputs are not so much sensitive to absolutes as they are to deltas, and our hearing is really no different. So, you know, this is also one of those philosophical things where PCM is this absolute machine, and DSD is this relative machine. And it’s just better for us as humans. And I firmly believe that general health would be better if PCM would not exist.

    Pappas: And why, particularly, is that?

    Meitner: Because there is a subliminal irritation about PCM that may just affect the psyches of people in a bad way, and certainly distracts from the pleasure of listening to music. And if listening to music was considered as relaxation and was supposed to be a way to relieve stress, then PCM, like CD playback, certainly doesn’t do it as well as some of the old analogue stuff did.

    Pappas: So maybe that’s one of the reason the music industry sales have been down.

    Meitner: Possible. Aside from the fact that, right now, it doesn’t seem to be the same scene as I remember from the ‘60s and ‘70s. This is a hard thing to say, but I hear from a lot of people that, with an LP, you used to sit down, close your eyes, and sort of float away with the music, relax and unstress. And with CD, it’s just not the same thing anymore. So even though you might not hear the problems glaring at you immediately, I’m sure they wear.

    Pappas: Do you think the other thing might be the fact that when you ran an LP, it was generally about 22 minutes, and then it was time to get up and change it?

    Meitner: Well, that could be too. Now with the CD you have the remote control. You can change it at all times. But I find a lot of people don’t even get to 20 minutes.

    Pappas: They’re bailing out long before then.

    Meitner: Yes. And you know the funny thing is, on top of it, you know all the converter people who started off in the multi-bit scene, have all changed to single bit. Phillips with their bitstream; and look at the vendors of DAC chips and A to D chips and it’s all gone to single bit. You would be hard-pressed today to still find multi-bit converters because of the added problems of them not being enabled to do zero crossing and stuff like that properly.

    Pappas: This problem is going to get worse with 24-bit. Because trying to get linearity at those low levels out of converters is a real problem for PCM guys.

    Meitner: I think the 24-bit is a crock of lies. I don’t know how many marketing bits have snuck into that.

    Pappas: So what do you think they really actually deliver, in terms of linear bits?

    Meitner: Oh, in the digital domain, I’m sure they deliver the 24-bits. But when it comes to analogue, I’ve never seen an analogue circuit that can do that.

    Pappas: So it’s marketing bits, huh?

    Meitner: Well, yes. But again, you know, the nice thing about it is that at least now your noise quantization has more bits.

    Pappas: Which improves your performance at zero crossing.

    Meitner: Yes, yes. And also that fact that usually those converters are of delta sigma modulation type design.

    Pappas: Which is DSD is once you take out the decimation filters.

    Meitner: That’s right. Yes. And you know, with audio traditionally, less has always been better.

    Pappas: I don’t know, unless you’re an English console manufacturer.

    Meitner: Usually less is better. If you can make a shorter feedback path, a shorter this, a lesser that, it usually works better sonically.

    Pappas: Has the output of DSD got some sort of a standard format?

    Meitner: It’s S/DIF. Except that it’s three wires so you have your sync, which is the clock, and then you have left DSD and right DSD.

    Pappas: Very interesting.

    Meitner: So of course you have none of that squirrelly S/PDIF where you have to extract a clock from the data and the whole data stream, which is another jitter problem.

    Pappas: And you also don’t have to extrapolate left and right out of it, either.

    Meitner: No.

    Pappas: That’s very interesting.

    Meitner: It’s discrete, you know, clock left and right.

    Pappas: That makes a big difference in trying to get the reconstituted clock out of it.

    Meitner: All this audiophile work with their separates? Like a CD drive and then a converter? Well, the weak link really is the S/PDIF. If that would have gone to even a clock and left/right on one wire, that would have been a lot better, instead of this one wire where everything is on one wire and then you have all these problems.

    Pappas: Very interesting. So in other words, it might actually be better to buy a one-piece unit —

    Meitner: Oh Yes. Oh Yes.

    Pappas: Because you don’t have to make the conversion to S/PDIF…

    Meitner: That’s right.

    Pappas: … and then back again. You can keep it all …

    Meitner: I think every time there is a conversion, nobody could be as arrogant as to assume that there isn’t going to be some loss.

    Pappas: And so effectively, you’re making two conversions by going to a two-piece system.

    Meitner: That’s right. No free lunch.

    Pappas: Not even a reduced-price lunch.

    Meitner: No.

    Pappas: Well, if this format is going to be accepted vendors are going to have to come forward with all kinds of hardware.

    Meitner: Actually less hardware.

    Pappas: But you’re going to need a multi-track machine.

    Meitner: Yes.

    Pappas: You’re going to need some sort of a way to do this into a two-track environment. You’re going to need some way to put it in a computer and manipulate it. When I say "lots of hardware," that’s what I mean.

    Meitner: Okay. But don’t forget that Sony and Phillips are pretty powerful engines.

    Pappas: Oh, absolutely.

    Meitner: They are really working — did you go to that AES show?

    Pappas: No,

    Meitner: Because you would have been really surprised how much gear there was there already.

    Pappas: Really?

    Meitner: Yes. And the issues with the dual layer disc which is backwards compatible. You know that never happened before in this industry.

    Pappas: Yes, you had to throw out everything that you used before.

    Meitner: Exactly.

    Pappas: And that was I think one of the reasons why the CD was so successful was that retailers could dump their LP stocking requirements and they put more product in a given amount of floor space than with LPs and they didn’t have as big of a return problem.

    Meitner: Right.

    Pappas: And one of the things that you will be very hard-pressed to do is get retailers to double-stock inventory.

    Meitner: Uh-huh.

    Pappas: You’re just not going to see guys like Tower Records want to give up any floor space for another format. So I think the dual layer thing to me was a brilliant piece of work, because it gets you around the hole. The retailer doesn’t know any difference. The consumer doesn’t know any difference. It just depends on what machine he plugs it into.

    Meitner: That’s right, Yes.

    Pappas: And that to me is going to be the key for making a format successful is it’s got to be transparent to the customer.

    Meitner: Yes.

    Pappas: Because frankly, you know, most customers think the perfect sound forever is okay right at this point. Also most customers are listening on boom boxes and ghetto blasters.

    Meitner: Right.

    Pappas: But they’re the guys who buy the majority of the music and drive the market.

    Meitner: So it’s not a bad thing. You know, it’s not some evil big company that wants to railroad the new format. In fact, they very seriously thought about it and came up with this dual-layered thing where nobody really loses anything. It’s a win-win situation the way I see it.

    Pappas: Well, and it looks like they’ve done a good job at least getting people like you and the folks at Sonic Solutions involved in it so that people know it’s not a one-stop-shop.

    Meitner: Right.

    Pappas: Because the industry does not like that. They want to go to multiple vendors and pick and choose who they’re buying stuff from and where they’re getting it from. And a one-stop-shop is not a good thing for this business.

    Meitner: I think this whole development is politically very correct. It’s the people that are interested in better sound will get the benefit from it. 96/24 can run in parallel. 96/24 can even be the top layer.

    Pappas: Oh really?

    Meitner: Well, why not?

    Pappas: And if you shoot it all in DSD then you can convert to 96/24.

    Meitner: Yes, you’ve got an open format that you can use for any sort of conversion. So I really don’t understand why people have a pickle up their rear and slam it.

    Pappas: Well, perhaps because they didn’t think of it.

    Meitner: Well, maybe, yes. I don’t really want to dwell on the naysayers; I know what their thoughts are. For sound quality, here’s one simple test. I’m doing some transfers of vinyl LPs onto DSD. And, you know, in DSD this is a conversion with a minimal amount of damage to the original sound. If you consider playing back vinyl and liking all the good things about it, now we can have it in DSD format. We could possibly take some of the clicks and pops out of it and still have the general good flavor preserved. The same holds true for analogue tapes and any kind of conversion. So it’s really a very nice thing.

    Pappas: Well, the analogue tape situation is becoming real critical, because all that stuff is really starting to fall apart.

    Meitner: That’s right.

    Pappas: And the ‘60s stuff actually isn’t quite as bad as the ‘70s, when everybody switched to the high-output tapes; those puppies are really falling apart.

    Meitner: Yes.

    Pappas: And if you make a conversion into 96/24, then that’s it: you’re stuck. You’re not going anywhere else.

    Meitner: You should also get the Tom Petty disc that MoFi did. Compare that to the one that was done with the DSD system and then down sampled with the Sony Direct, SBM direct versus the original. It is a world of difference.

    Pappas: Really?

    Meitner: Yes, you definitely want to get that. Because that will show you that when it’s down converted properly, how much of the good flavor is preserved.

    Pappas: I like your term for it: "good flavor." To my way of thinking, the DSD is a significant change in thinking in terms of how digital works. We have been doing PCM, effectively, since the late ‘70s. And I think DSD presents a real shift in thinking. I have to assume that you guys are in that same kind of vein thinking that way.

    Meitner: It’s the undigital digital.

    [edited on Thursday to decrease length of post]
  2. John Kotches

    John Kotches Cinematographer

    Mar 14, 2000
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    If you didn't get permission to post this interview verbatim from Positive Feedback, you've just violated international copyright law, and probably the rules of the forum. In the future, you might want to post a link instead.
    With that out of the way, I've read this quite a number of times. I've even talked to Ed about this personally, and he knows I don't agree with everything he says. He tolerates me anyway [​IMG] There's no way I know more about the topic than he does, but that doesn't mean I'm not trying to learn more either.
    Here's an interesting analogy for you.
    Let's say you want to digitize a painting. Your color palette consists of either complete black, or complete white. Is the current pixel brighter or darker than the previous? If it's brighter it's 1. If it's darker it's 0. If it's the same, it will be the inverse of the previous sample, as you want to maintain an average. The goal is to capture the difference in brightness.
    On the other hand, another method scans the picture, and while it doesn't have as many pixels, the color palette is very large, 24 bits, just like our computer monitors these days. The goal is to accurately calculate the value of the color at any given position.
    The first example is DSD. The second is High-Res PCM. While you might say this isn't an apt analogy, I'll say... Why not? In the end we're talking about a voltage on a Charge Coupled Device at a particular point in time which is being sampled. The encoder knows not what it encodes, it just carries out the task of checking voltages vs. previous, or calculating with high precision it's value.
    This is no different than sampling the voltage off a microphone at a point in time.
    We haven't touched on editing and DSP issues yet. How many recordings have zero editing, and zero processing? My guess? Well under 1%. Since I don't work in a recording studio, I can't answer that -- what has your experience been?
    Just what is the bit depth of DSD? Is it 1-bit? Is it 8-bit Quasi-PCM? Is it 1.5 bit? I just don't know. Why? Because depending on who you ask, and where in the signal chain we're talking... that's the answer you'll get.
    Encoder is supposed to be 1-bit. But to do any signal processing or editing, you have to change over to an 8-bit Quasi-PCM, then go back to a 1-bit result for passing down the chain. At each junction, you have to noise shape, just as you have to re-dither PCM. This is far, far, far from that pristine chain that Sony and Philips tout in the marketing glossies.
    I'll spare everyone from posting the Vanderkooy and Lipshitz papers on the topic of DSD. Needless to say, they (and a number of other individuals) aren't thrilled by DSD.
  3. Lee Scoggins

    Lee Scoggins Producer

    Aug 30, 2001
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    I'm not sure your analogy gives full credit to the much higher sampling rate of the DSD process and I'm not sure it is fair to compare 1 bit to 24 bit as black-white versus a deep color palette.

    In any event, I have heard that zero-crossing is a big issue although the mastering chain is getting simpler as DSD implementation improves.

    Ironically, for all our technical discussion, it does appear from my friends in the New York studios that the analog inputs and output devices like Ed says are improving and making the most difference.

  4. John Kotches

    John Kotches Cinematographer

    Mar 14, 2000
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    In the end -- the comparison is to a voltage on the plate of a CCD. It isn't really comparing a "color".
    I've asked a couple of friends about the aptness of the analogy -- and they'll give me honest feedback sometime later today.
    I understand the fundamental principle here -- the ear (and eye) are tuned to changes, and DSD takes full advantage of that fact.
    I just don't think DSD is the best answer.
    And yes, analog itself is getting far, far better. In the end, we all benefit from the technology, because at your end (the studio) the recording technique itself (DSD or high-res PCM) is no longer the limiting factor. All the associated gear is.
    This helps push the associated gear to being better, which will eke out improvements in the recording technique (A/D convertors, etc. Ahh the beauty of the digital world [​IMG]
  5. Yohan Pamudji

    Yohan Pamudji Second Unit

    Apr 3, 2001
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    I'm not very knowledgable in the technical aspects of audio, but I'd like to jump into this discussion because it's really fascinating to me. Do your worst! [​IMG]
    Actually I need to get some questions out of the way before I can say anything close to intelligent. Here's where I show my ignorance: when an audio signal is digitized, whether it be DSD or PCM, what is being measured and stored? Amplitude at any given sample point? Are DSD and PCM different in this regard, that is, do their respective bitstreams represent/encode different aspects of audio? If these questions are too newbie in nature, feel free to point me to some good primers on the subject.
  6. John Kotches

    John Kotches Cinematographer

    Mar 14, 2000
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    PCM tries to sample the precise amplitude at a point in time, with relatively high precision. This is done with a series of approximations, which are honed down to a closer and closer value to the actual voltage at the point in time. Each approximation uses a step half the size of its predecessor. These are the various bits. This is anywhere from 16 to 24 bits as delivered to consumers.

    DSD is a form of Delta-Sigma Encoding. Remember calculus? Delta was a change, Sigma a summation. So Delta-Sigma encoding is a Change Summation. It is a single bit encoding format. With DSD, 1 is a full on + voltage, and 0 is a full on - voltage. Zero is a special case, and depending on the duration is either flagged with metadata, or oscillating 0 and 1s to get an average value of zero. What happens is that a point in time voltage is sampled, and the question is simply are you higher or lower than the previous sample? Higher = 1, lower = 0, and the same = opposite of previous values. I think of it as frequency modulating a square wave, and using a low-pass filter to remove the square wave. This leaves you with just the FM portion as the analog waveform.

    I spent more time on DSD, because it is a less familiar process to most people.

    Hope this helped.


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