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Thread: Final words about increase in bit depth/sample rate?

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    Final words about increase in bit depth/sample rate?

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    There's a lot of talk these days about the benefits of 24 bit/96 kHz recording over the old CD standard. Often there is a lack of technical knowledge and instead you are fed with final words like "There's a significant audible difference..." or "There's no audible difference whatsoever..."

    Claimed benefits of recording at a greater bit depth are:
    - Better dynamics/headroom - you can afford to lose a few bits here and there due to low levels and still have at least 16 used bits in all tracks when it's time for mixdown.
    - Plugins who can deal internally with higher bit rates sounds better when fed with 24 bits than 16.
    - A multitude of audio tracks mixed down to a final stereo track would present a higher demand in digital representation than just a few tracks. The 16 bit/44 kHz format is not sufficient to hold all this data with accuracy.

    I feel comfortable with the first and second claim, but the third one I find questionable. Is it really true that it requires more digital "resources" to represent the sound of a full symphony orchestra than it requires to represent a simple sine wave? And if so, is the CD standard format not sufficient to deal with accurate representation of very complicated wave forms (like the symphony orchestra or a multitude of mixed down audio tracks)? Is there a real and for every added track increasing degradation in sound when mixing down several 16/44.1 tracks to a single 16/44.1 master? Or is this just plain nonsense? If this is true it also implies that you would benefit from mixing down a multitude of 16/44.1 tracks to a 24/96 (or at least a 24/44.1) master which would more accurately be able to hold all data from the tracks added together, provided all 24 bits are used and not just the lowest 16 bits.

    To elaborate here, mixing down a bunch of 16 bit tracks within the same system (i.e. Cubase Export Audio) to a final 24 bit mix with everything at unity (master set not to go above 0 dB) would result in a 24 bit file with only the bottom 16 bits used. But if you calculate the added headroom of these 8 unused bits, you should be able to increase the levels on the master faders n dB above 0 dB (calibrated for 16 bits) and thus get a 24 bit mixdown with all bits used. Mixing down from 16 bits doesn't equal mixing down 1x16 bits but Nx16 bits, where N is the number of tracks. When mixing down to 24 bits you can allow more of these Nx16 bits to be transfered to the mix, to put it simply. In real life the equation gets a bit more complicated if we consider that some tracks are stereo and some are mono and the final mix is stereo. Then Nx16 then corresponds to the total number of mono tracks (where each stereo track is regarded as two mono tracks). The resulting mix would have 2x24 bits capacity.

    My guess is that audio quality is preserved no matter how many tracks are mixed down, and this is due to the fact that every track is mixed down with only a fraction of its original amplitude. Everyone knows that the more tracks you have in a mix, the more you have to back off on the faders in order not to get a total signal above 0 dB and introduce digital clipping (provided the individual tracks are recorded at or near 0 dB level). The amplitudes for all tracks are added together and must be reduced to fit the headroom in the final mix. Less amplitude/volume means less need of number of bits in the representation, thus no significant degradation in sound.

    The benefits from an increase in sample rate is more questionable. According to Nyquist 44.1 kHz can accurately reproduce frequencies up to 22.05 kHz, which is above most peoples (and certainly most ear abused musicians) hearing limit. Sufficient oversampling and good AD/DA converters is of course a must. What would be the benefits of increasing the sample rate to 96 kHz? Do you get a flatter frequency response and less distortion at the highest audible frequencies? Is the Nyquist theorem just theory and is 44.1 kHz sample rate when it comes down to dust really not enough to accurately reproduce all audible frequencies? Where does for example oversampling come into the picture or statistical quantification errors, the latter suggesting a greater accuracy for higher sampling frequencies than 44.1 kHz?

    These are, I think, important questions for anyone involved in digital recording. Any facts beyond "I hear the difference" are welcome. Hints of good literature (books, web sites) that deal with these questions would also be appreciated.

    /Mats D, Sweden

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    I agree with you about those 2 facts of better-than-CD quality. In general CD quality is BETTER than most people's hearing and the environment they listen in (also many people wouldn't realize that a 16-bit 44.1 kHz recording of a good vinyl album wasn't direct to digital unless they could hear the rumble, scratches, and wow). It is only the music/equipment industry that keeps pushing us to want for more so that they can make more sales on existing intellectual property (e.g. 5.1 surround sound for anything other than movies). As far as I'm concerned, the only improvement they could usefully make to a stereo CD is to make it unscratchable.

    I think one factor that is frequently ignored is that people's ears react differently to edges vs. continuous high frequencies. That is to say, ears can hear the momentary burst of high frequencies from a snare drum far better than they could hear a sine wave at 22 kHz. I don't know how that factor relates to this discussion however.

    My final comment is that there will always be those audiophiles that spend more time listening to the gaps between the songs than they do listening to the music. I think that's because they enjoy the technology or the snobbery MORE than the music.

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    What i am most interested in is if mixing down to a greater bit depth than the one used for the individual tracks has any benefits.

    An example: My song is made up of four 16 bit mono tracks, recorded hot (just below 0 dB). The total dynamic content is 4x16=64 bits. When mixing down to 16 bit stereo format, I need to cut the dynamic range down to 2x16=32 bits by reducing the volume of each track. But if I mix down to 24 bit stereo format, I would have 2x24=48 bits capacity and would not need to reduce the total dynamic content of the four mono tracks as much, and that would be a real benefit. But I'm not sure this is possible in the digital domain. Perhaps you always end up with a 24 bit stereo mix where only the bottom 16 bits are used and the eight top bits are empty, which indicates that it would be useless to mixdown to a greater bit depth than the original format. Perhaps you need some software or hardware device in order to resolve n number of 16 bit tracks to make full use of the 24 bit format.

    /Mats D

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    I didn't have the stamina to read through your entire post. But this is how it is:

    A 16-bit wave file is made up from sixteen-bit numbers. A sixteen-bit number is a number between 0 and 65535 (in base ten ). When you mix down a multitrack project to a stereo file what happens is - not exactly but lets say it anyway for the sake of simplicity - that the numbers for each of the tracks are added together. So mixing down 48 sixteen-bit tracks could give peaks of 48*65.535=3.145.728. And to represent that number you need log2(3.145.728) = 22 bits. And i hope that was what you were asking for, or I'll look kind of stupid.

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    Re: Final words about increase in bit depth/sample rate?

    Originally posted by MatsD
    with only the bottom 16 bits used....


    According to Nyquist 44.1 kHz can accurately reproduce frequencies up to 22.05 kHz, which is above most peoples (and certainly most ear abused musicians) hearing limit. Sufficient oversampling and good AD/DA converters is of course a must. What would be the benefits of increasing the sample rate to 96 kHz?....

    Usually its the first 16 bits that are kept and the last 8 are randomly truncated. Truncation is Bad!

    The Nyquist Theorm is great for higher frequencies, but suffers when the lower frequencies are sampled. The number of times you quantize a wave length is directly related to the frequency. So the a 20hz signal is so long that its nearly impossible to get a good sample when compared to a 20khz signal. There is alot of ambience at the higher frequencies that the converters just are not designed to sample. There is alot more to the issue of the increase of the 24/96 movement, but when you compare it to 16/44.1, its clear there is more definition of everything. Plus everytime a process is done digitally, the recalculations change the actual signal to a different frquency.

    Peace,
    Dennis

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    OK, so there is a gradual degradation from high to low frequences regardless of sample frequency but less obvious the higher frequency you use, due to the fact that lower frequencies equals longer wave lengths. OK, I buy that. You will primarily get a better representation of the lower frequencies by using 96 kHz instead of 44.1 kHz, right?

    I'm still confused about the bit issue though. Why would it not be possible within the digital domain to add two 16 bit files to one 24 bit file? Mathematically this presents no problem as far as I can see. If you can add two 16 bit files together and then reduce the dynamic range to fit within the dynamic range of a new 16 bit file, why can't you add two 16 bit files together and then reduce the dynamic range (somewhat less) to fit within and fully occupy the greater dynamic range of a 24 bit file?

    /Mats D

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    None of what you two guys say in the last two posts is correct, or even make any sence. But I have to get some sleep before I cach a train in four and a half hours, so I'll wait till tomorrow setting you straight. And hopefully someone else will have explained it by then...

    Oops, Sonusman alredy did it. I really have to take a speed typing class...

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    Atomictoyz wrote: "The Nyquist Theorm is great for higher frequencies, but suffers when the lower frequencies are sampled. The number of times you quantize a wave length is directly related to the frequency. So the a 20hz signal is so long that its nearly impossible to get a good sample when compared to a 20khz signal."

    Regrettably, this is false. Nyquist stated the upper limit of the passband with respect to the sampling frequency, but not the lower limit.

    Shannon's Sampled Data Theorem is the true basis for modern digital audio- Nyquist just kind of embroidered around the edges a little, years and years later. (;-) The fact is that Shannon's sampled data systems have no lower frequency limit, as such: they are theoretically accurate to +- 1/2lsb *right down to DC*.

    In practice (as we use them in digital audio), there are usually highpass properties designed into the actual converter support circuitry. But if the converter itself were to be wrapped in DC-coupled analog circuitry for one reason or another, you're good right down to DC. Example: the digital multimeter in your toolbox.

    You usually don't _want_ to be DC coupled for audio, though- there's no sense having your precious few bits used up in preserving DC offsets from previous stages, when you are trying to somehow preserve the ten-trillion-to-one dynamic range of our auditory apparatus! In practice, digital is too damned _good_ at preserving LF information: most knowledgeable people will actually roll off their signals below 20Hz or so just to keep the infrasonic stuff _out_ of their mixes, because digital sampling will preserve it very nicely. But real-world woofers can't do a thing with it...

    Anyway, Nyquist is silent on the lower limit of the passband in a sampled data system for the simple reason that the lower limit *is* DC, with no reduced-resolution effects whatsoever down there.

    Sonusman has offered excellent advice here: the essays at http://www.digido.com should be required reading for everybody *before* the religious arguments start. This is hard stuff, but it is _science_, not art: it can be understood, you really don't need to be an EE to understand it, and the pseudoscience and misunderstandings really should be put to rest.

    They never will be, probably, because a lot of this stuff has taken on the mantle of religion. And regrettably, religious-style issues are used by many people as a convenient excuse not to probe futher. But the truth is out there, and digido is an excellent place to start in achiving a _true_ understanding of resolution issues, distortion residuals, and the effects of proper dither compared to the effects of simple truncation. And learn not only that it can be heard (hell, even I can hear it, and everybody knows I'm as deaf as a post!), but _why_ it can be heard...

    It's absolutely worth everybody's while to get up to speed on that stuff- and _then_, let's talk.

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    I was not implying that 24 bits would make anything sound louder, in fact I was not stating the essence of the benefits at all. But if I am to do so, I meant that there ought to be a way of mixing down the sum of the dynamic ranges of several 16 bit tracks to fully occupy the dynamic range of a 24 bit file. What I am talking about is a better dynamic resolution, not a boost in volume. Perhaps there's a fundmental reasoning error in my suggestion, but the technical stuff from you two guys didn't make me any wiser and seems to a great extent to deal with other issues.

    The practical questions remains - can there in any way be any dynamic benefits of mixing down a bunch of 16 bit tracks to a 24 bit "master"? Is the nature of the digital domain such that I, no matter what, always will be ending up with a 24 bit file where 8 bits are unused? Is it impossible to reduce the sum of several 16 bit dynamic ranges to the full resolution of a 24 bit mixdown recording?

    /Mats D

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    sigh................

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