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32 bit and above

Discussion in 'Recording' started by andshesbuyingastairway, Apr 26, 2007.

  1. maybe it's a remedial question, but why aren't there 32 bit and above PCM converters, why does it just jump from 24 to delta sigma
  2. RemyRAD

    RemyRAD Guest

    I may be incorrect here but I think I had a some kind of oversampled blah blah Delta Sigma converter in my old Panasonic DAT recorder? I've never quite heard what you just expressed before? You aren't referring to the Sony "DSD" recording system are you? Besides, that's not expressed as 32-bit. It's a single bit system. But then, you know I've been wrong.

    Although now I believe, with all of the 64-bit dual core processor integration and newer 64-bit operating systems, higher speeds, extended RAM capabilities, etc.I think we'll see 32-bit soon? Or maybe not? Since it would be moot with DSD (direct stream digital). I'd really love to have one of those systems, just can't afford it yet. DSD, sounds sooooooo much better to me than PCM. But that's just my opinion. And I have one of those same things that everybody else has, which I'm currently sitting upon.

    30 pounds lighter now!
    Ms. Remy Ann David
  3. dementedchord

    dementedchord Well-Known Member

    a 32 bit pcm convertoer would just be so much masturbation.... if viewed as discrete slices of a voltage the breakdown would 16bit -2 (16thpower)
    24 bit 2-(24th power) and of course 32- 2(32 power) i dont feel like doing the math... and realize that in order to get the dynamic range you would take each of those and devide by 6.... this is not to be confused though with use of a 32 or even 64 bit accumulator and storage there of.... welcome back liquid....
  4. Boswell

    Boswell Moderator Distinguished Member

    A few clarifications:

    PCM (pulse code modulation) is the name for the representation of an analog waveform as individual digital samples, uncompressed. It does not specify how the sample was converted from analog to digital, or how it will be converted back from digital to analog. Each PCM sample is an n-bit accurate representation of the analog signal at the sampling instant and does not depend on what comes before or after. In a band-limited signal, there are limits to the range of values that could precede or follow a given sample, but that is not a property of PCM. Note that PCM values can be represented as integers (fixed-point) or as floating-point numbers.

    Delta-sigma (also the slightly different sigma-delta) is a method of converting analog signals to digital representations. It contrasts with successive approximation (SAR) and other types of converter. Sigma-delta converters work by making high-speed single-bit decisions about whether an accumulating representation of the signal is greater or less than the signal itself, and adjusting the accumulating value appropriately. A number of digital filtering and noise-shaping techniques are applied to the accumulating value, and the end result is output at the word sampling rate as an n-bit PCM value. Note that accuracy (number of bits) is not in principle a function of the type of converter (SAR, D-S etc), but it is easier to construct converters that have a high dynamic range using D-S (or S-D) methods due to the noise-shaping techniques that can be employed.

    The limit on the number of useful bits of an audio ADC is largely determined by circuit implementation and the physics of noise. It's really difficult to get more than about 120dB (20 bits) of signal-to-noise ratio, but yet a 24-bit converter sounds demonstrably better than a 20-bit converter, not least because the dynamic range is increased. Similarly, 28-bit converters are becoming available, but whether any audible improvement over 24-bit would be due to the extra 4 bits or to the necessary increased care in the realisation of the converter improving its performance at the 24-bit level remains to be seen.
  5. so you're inclined to say that people will just as soon go delta-sigma DSD before they extend the range of PCM bit resolution?

    there are some arguements that say 32 bit (and such and such half the sampling rate of DSD) would yield more high fidelity results.

    sonic nature over ease of implementation
  6. sheet

    sheet Well-Known Member

    Look. We can't even get a full 24-bits out of the 24-bit converters now. They are like 20-bits really. Going to 32-bits really won't benefit much.

    Sampling for DSD is an expensive proposition. There are less than 5 manufacturers making converters for it.

    DSD is not a good multitrack/DAW solution, especially if you want to edit. You cannot really edit on it the way that you can say with PT or any other high performance DAW. You have to use it like you would an analog recorder.

    DSD/SACD is pretty much dead in the marketplace.
  7. what i am referring to is extending PCM in terms of bit quantization as opposed to going with DSD (which doesn't have to be associated with SACD.) not necessarily just 32, but 48, 64, and the numbers in between. what do you think about that?

    i'm familiar with the necessity for using DXD to edit, describe what you mean when you imply that its a hassle to edit with DSD.
  8. Boswell

    Boswell Moderator Distinguished Member

    As I said above, you can represent the digital samples in any wordlength you like. For example, if you were using an application compiled and linked for 64-bit Windows Vista, your samples would be held in a 64-bit word. It doesn't make the samples any more than 24-bits accurate, or whatever your converter was in the first place. What we have been trying to tell you is that the current state of the art is somewhere between 24 and 28 bits, and is unlikely to get better than that using present technology, for reasons of physics.
  9. sheet

    sheet Well-Known Member

    You should study what DSD/SACD is. DXD is not the topic.

    You cannot build a true 32-bit audio converter. Heat is the big killer, plus it is unnecessary. There are no true 24-bit audio converters that I know of.

    To full realize the benefit of DSD, you must capture at the appropriate sample rate, edit, mix and master. To do what others have done, capture at 24/96 or 192 and then master to it is silly. The majority of the industry thinks DSD/SACD is silly, because few can hear a difference. Sales verify that as well.

    Senoma and Pyrmamix are the only two DSD editors. Neither have the flexibility of a DAW like PT, Check out what they do and why. Also check out what it would cost to own a complete 24-track Sonoma system: $72,400!!!. All of that expense for a debatable difference in sound to engineers and the consumers. This is the reason it isn't selling. Most people can't hear a difference between 192k and 44.1. Why go higher into the 300s and eat more drive space? Crazy.
  10. interesting, so what kind of "physics" more specifically when you say "heat is the killer," prevent the creation of 32 bit and beyond PCM converters? in the advent of moore's law and technology in general always increasing are you willing to take your statement "you cannot build a true 32 bit audio converter" to the bank?

    what kind of flexibility does sonoma lack? i believe on that price that would be including a 3.4GHz computer (definently necessary for this application, but speed is always increasing anyways and for lower cost) and the converters. mytek 8X192's are DSD compatible (will it communicate with sonoma?). for one 8 track PCI card and the software it's 8,000 just like pro tools. but thanks for your comments,
  11. dementedchord

    dementedchord Well-Known Member

    ok.... turning it back on you what would it accomplish???? what need does it fill that isnt being addressed??? the demonstrable advantages????
  12. how is that turning it back on me? more like turning it back on the engineers who relentlessly study and attempt to create different outlets for digital. quit trying to make every post a mockery, some of us are in the pursuit of knowledge.

    go read some white papers if you want that question answered. or at least start a new topic.
  13. dementedchord

    dementedchord Well-Known Member

    exactlly.... a nonanswer... i'm not wishing ill for tony snow but you could be bushes next media guy... and i have and will continue to read the white papers etc... infact you can probably do a search and see where i've posted links to some... and the pursuit of knowledge does drive me.... i'm painfully aware of some areas i dont understand... and the price i pay for my enlightenment is the sharing of that which i do understand.... what do you do becides baiting everyone???? i only bait you and your ilk...
  14. i'm not baiting anyone, this isn't a competition i'm sorry you fail to realize that, move aside and let the people who want to discuss this do so.
  15. sheet

    sheet Well-Known Member

    You cannot compare the two. For one, plug ins for DSD are rare, and you cannot get the same plug ins offered by TDM plug providers. So, you have more plugin options with PT. Sonoma is not a full featured DAW. Neither is Pyramix. Again, if you look at what DSD is actually doing, you will see that editing in PT is not the same.

    I personally spoke with two major league Grammy power users who tracked in Sonoma, because it was free for them to try out. They ended up going back to Sequoia.

    DSD at 192 is pointless. Learn what DSD is about.

    As far as price, you can buy a PTHD3 that has 24 analog I/O, able to record, edit and mix hundreds of tracks, with MIDI features, etc, etc for mucho less than Sonoma. I looked hard at Pyramix before I bought PTHD. It was not even close performance and options wise.
  16. yeah i could have figured the plug-in's aren't going to be bundled. i understand what you are saying about if one is to use DSD, use it consistently; throughout input, edit, master

    i assume that when you say DSD is pointless at 192, you mean recording DSD then going into a PCM program like pro tools
  17. ghellquist

    ghellquist Member

    The limiting factor is indeed heat. I will take a roundabout and come back to why.

    In audio we take a bit of a shortcut sometimes and say that each bit of resolution represents 6 dB of signal/noise ratio. (It is a very good shortcut as they go, not the full story but close enough).

    The signal/noise ratio is basically the difference in strenght between the strongest signal we have and the noise floor. You can easily hear this on just about any electronic media, just turn the volume up till you just about hear the noise, then add a signal and turn the volume up on that one til your ears hurt.

    In a typical bedroom you might expect about 40dB sound pressure of noise in the room. Let us assume you crank your sound source up until it adds 3dB of noise to this. The sound source would then output 40 dB of sound and the resulting sound in your room will be 43 dB.

    Now if we take a perfect 16 bit signal, maximum signal/noise ratio is around 96dB (again, an approximation, I will not go into the details here). This would put a maximum strenght signal at 139 dB, well up among starting jet fighters and sure to hurt your hearing.

    What I wanted to say by this, is that 96dB of signal noise ratio is a very large ratio. In many applications, when used with care, it will work a long way towards transmitting your music. In fact, typically a CD gives you a little less than that depending on dithering, but for most domestic uses the ratio is not a problem.

    Now enter the so called 24 bit converters. Read up on the figures of signal to noise ratio. Few if any gives you more than 120dB as that figure. Why then? That should be only 20 bits of resolution.

    Yes, the truth is that most so-called 24 bit converters gives you less than 20 bits of true resolution. The reason is heat.

    Heat creates random motion of electrons. This random motion is the noise (as in the signal to noise ratio). In the real world it is difficult to create a full signal chain (analog circuits + AD converter) with an S/N better than 120 dB. It can of course be done, but it is very difficult to do on the very small real estate you have inside an integrated circuit. It can be done with discrete components, but is generally not done due to the great cost.

    One way to increase signal to noise ration is to cool the circuits. If you go to a radio astronomical observatory, you can probably find circuits with very great ratios, cooled by liquid nitrogen or such.

    But it all boils down to two things:
    one : there is no need. When you record a source, the signal to noise ration is hardly ever larger than 80 dB. Nowhere close to the 120dB your AD gives you and even further from the theoretical 144dB of a 24 bit converter.
    two : the price would be astronomical. (Today that is, what the future brings is another thing).

    Hope this rather unscientic explanation may help understanding why we do not see more than 24 bits as advertised today.

  18. dementedchord

    dementedchord Well-Known Member

    thanks gunnar.... perhaps that will put it to rest....
  19. you say that heat causes noise, yet we are following the rule that reducing the amount of bits gives you noise. from deduction that would mean that adding bits gives you less noise. while you are implying that there is increased heat (and therefore increased noise) throughout higher amount of bits.

    maybe you could clarify
  20. dementedchord

    dementedchord Well-Known Member

    nope your confusing different issues here i believe... the nature of the noise is that it comes from the heat of operation... current is used it manifests as heat loss and exhibits a noise.... incresing the bit depth it works harder and increases.... the difference in the values expressed increases so the dynamic range increases despite the fact that for all intents we cant use them as a practicle matter relitive to the increased noise floor... so it seems counterintuitive that we can use the dynamic range of the lesser bit depth because the noise floor is lower...

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