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#42
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Mixing and tracking at 96k verses 44k??
On Mon, 27 Oct 2008 23:26:29 GMT, Blind Hog
wrote: My first action in the editing software is to expand to 24 bits. My last action before burning a CD is to convert back to 16 bits. Both of which are surely inevitable (and transparant to the operator) in any decent multitrack editor? Except that the internal format is more likely to be 32-bit. |
#43
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
"geoff" wrote in
: What recording device are you using ? Obviously available media/memory is a restriction with it for you..... I use an Edirol R-4 with (I think) a 30 gb hard disk. 24 bits would allow less fretting over level-setting optimisation, and obviate the necessity for extra processing should ypou feel the need to convert. But 16 is enough and if I'm more than 30 dB off level, I'm hearing a lot of complaint from my preamps. I can't imagine that you would be getting any benefit from converting to 24 bits before any processing. But that depends, I guess, on how primitive your editing app is. The processing 'bitdepth' does not necessarily have to have a fixed relationship with source media file bit-dept. I use Audition. I can hear a difference when I add effects like noise reduction and reverb that trail to zero. I don't know of any software that will perform a process (regardless of the bit depth) that can cleanly perform to the last bit or two of resolution. Better to do that work in more bits, perform all the operations, then dither to 16 and cover every math error. |
#44
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
Blind Hog wrote:
I can hear a difference when I add effects like noise reduction and reverb that trail to zero. I don't know of any software that will perform a process (regardless of the bit depth) that can cleanly perform to the last bit or two of resolution. Better to do that work in more bits, perform all the operations, then dither to 16 and cover every math error. Well there are reverb plugins and reverb plugins, and not all are the same, or use the same principles. Some may be vastly superior (or not) to what you are using, and may (or may not) do those last 2 bits that you are worried about. And by the way, changing from 16 to 24 bits is NOT just adding extra zeros at the bottom end. geoff |
#45
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
Robert Orban wrote:
In article , says... There is no nonlinear distortion in a digital filter unless you screw up and overdrive it. Digital filters generally have far more dynamic range than analog filters, so it is unlikely that it will clip unless someone really screws up. Virtually all digital filters have some sort of nonlinear distortion. The IIRs typically used in equalizers are subject to limit cycles and truncation noise due to finite length arithmetic. There are various techniques (like error feedback) for minimizing these issues, but it is very rare to find a digital filter in the real world that in fact adds dither before every truncation, which is what would be required to linearize the filter. It's becoming increasingly popular to do this in DAW software, though, and the increasing move to floating point implementations on DAW systems change the precision issues a little bit (although they don't eliminate them). These problems all increase as the poles of the IIR filter get closer to the unit circle in the z plane. This is a strong argument for using as low a sample rate as you can get away with because doing so minimizes the noise in any filters. In audio, this includes EQ as well as crossovers in multiband compressors. In FIR filters, there is a similar advantage to minimizing the sample rate because it reduces the length of the filter required to produce a given target response. If you are doing nonlinear processing, like compression and limiting, then you can get aliasing in the sidechains and also when the audio is multiplied by the sidechain signal. So here the advice is different -- use a sampling rate that is high enough to render any such aliasing inaudible. These are both extremely true and they are also both good arguments for doing the actual processing with floating point data. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#46
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
(Scott Dorsey) writes:
[...] Robert Orban wrote: If you are doing nonlinear processing, like compression and limiting, then you can get aliasing in the sidechains and also when the audio is multiplied by the sidechain signal. So here the advice is different -- use a sampling rate that is high enough to render any such aliasing inaudible. These are both extremely true and they are also both good arguments for doing the actual processing with floating point data. Huh? Even infinite-precision arithmetic won't help with these. The problem is one of aliasing, not resolution. -- % Randy Yates % "Rollin' and riding and slippin' and %% Fuquay-Varina, NC % sliding, it's magic." %%% 919-577-9882 % %%%% % 'Living' Thing', *A New World Record*, ELO http://www.digitalsignallabs.com |
#47
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
Randy Yates wrote:
(Scott Dorsey) writes: [...] Robert Orban wrote: If you are doing nonlinear processing, like compression and limiting, then you can get aliasing in the sidechains and also when the audio is multiplied by the sidechain signal. So here the advice is different -- use a sampling rate that is high enough to render any such aliasing inaudible. These are both extremely true and they are also both good arguments for doing the actual processing with floating point data. Huh? Even infinite-precision arithmetic won't help with these. The problem is one of aliasing, not resolution. Ahh, seperate issue, then! I thought he was referring to a mathematical precision issue causing aliasing, not the harmonic distortion inherent in any nonlinear system. In the analogue world, though, a lot of people _like_ the distortion products caused by compression and limiting. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#48
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
(Scott Dorsey) writes:
Randy Yates wrote: (Scott Dorsey) writes: [...] Robert Orban wrote: If you are doing nonlinear processing, like compression and limiting, then you can get aliasing in the sidechains and also when the audio is multiplied by the sidechain signal. So here the advice is different -- use a sampling rate that is high enough to render any such aliasing inaudible. These are both extremely true and they are also both good arguments for doing the actual processing with floating point data. Huh? Even infinite-precision arithmetic won't help with these. The problem is one of aliasing, not resolution. Ahh, seperate issue, then! I thought he was referring to a mathematical precision issue causing aliasing, not the harmonic distortion inherent in any nonlinear system. In the analogue world, though, a lot of people _like_ the distortion products caused by compression and limiting. Yes, but it is not those distortion products that Robert was attempting to mitigate. If you want to implement compression or limiting in the digital domain, it helps to first upsample the signal before applying the nonlinearity, then decimate back to the original sample rate afterwards. In this way, fewer of the harmonics that result from the nonlinearity are aliased back. -- % Randy Yates % "She has an IQ of 1001, she has a jumpsuit %% Fuquay-Varina, NC % on, and she's also a telephone." %%% 919-577-9882 % %%%% % 'Yours Truly, 2095', *Time*, ELO http://www.digitalsignallabs.com |
#49
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
Laurence Payne wrote in
: On Mon, 27 Oct 2008 23:26:29 GMT, Blind Hog wrote: My first action in the editing software is to expand to 24 bits. My last action before burning a CD is to convert back to 16 bits. Both of which are surely inevitable (and transparant to the operator) in any decent multitrack editor? Except that the internal format is more likely to be 32-bit. Inevitable? If I'm working in 16 bits the data is in 16 bits. Surely it upsamples to perform and operation and downsamples again afterward, but I don't want the downsample to happen but once--just before I burn the CD. |
#50
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
"geoff" wrote in
: Blind Hog wrote: I can hear a difference when I add effects like noise reduction and reverb that trail to zero. I don't know of any software that will perform a process (regardless of the bit depth) that can cleanly perform to the last bit or two of resolution. Better to do that work in more bits, perform all the operations, then dither to 16 and cover every math error. Well there are reverb plugins and reverb plugins, and not all are the same, or use the same principles. Some may be vastly superior (or not) to what you are using, and may (or may not) do those last 2 bits that you are worried about. I'm not worried about them because the bottom 8 bits disappear entirely just before I downsample to 16 bits. And by the way, changing from 16 to 24 bits is NOT just adding extra zeros at the bottom end. In what way is it not? |
#51
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
On Wed, 29 Oct 2008 23:08:23 GMT, Blind Hog
wrote: My first action in the editing software is to expand to 24 bits. My last action before burning a CD is to convert back to 16 bits. Both of which are surely inevitable (and transparant to the operator) in any decent multitrack editor? Except that the internal format is more likely to be 32-bit. Inevitable? Yup. They mix in 32-bits - you don't get any choice about it. A 16-bit input isn't "expanded" exactly, it's just put in an environment where there's LOADS of headroom. The only place you have to worry about keeping under nominal zero is the output bus. But then you aren't expanding either - just padding out with 4 more empty bits. If your editor is one that mixes in a high bit-depth environment there's no point in you bothering. |
#52
Posted to rec.audio.pro
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Mixing and tracking at 96k verses 44k??
Blind Hog wrote:
"geoff" wrote in : Blind Hog wrote: I can hear a difference when I add effects like noise reduction and reverb that trail to zero. I don't know of any software that will perform a process (regardless of the bit depth) that can cleanly perform to the last bit or two of resolution. Better to do that work in more bits, perform all the operations, then dither to 16 and cover every math error. Well there are reverb plugins and reverb plugins, and not all are the same, or use the same principles. Some may be vastly superior (or not) to what you are using, and may (or may not) do those last 2 bits that you are worried about. I'm not worried about them because the bottom 8 bits disappear entirely just before I downsample to 16 bits. Save 'em off seperately. They're nothing but noise.... And by the way, changing from 16 to 24 bits is NOT just adding extra zeros at the bottom end. In what way is it not? -- Les Cargill |
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