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For editing, you want to use the highest possible starting bandwidth (sampling rate * bit width) because every intermediate edit and calculation leads to round-

For editing, you want to use the highest possible starting bandwidth (sampling rate * bit width) because every intermediate edit and calculation leads to round-off and other errors. You also need them to be WAVs or some other raw format since you cause minor errors every time you compress + decompress something, not to mention the performance loss. It's the same rationale for doing calculations in full 80-bit or 64-bit FP even if you don't need the full bits of precision -- if you start your calculations at your target precision, your result will have insufficient precision due to truncation, roundoff, overflow, underflow, clipping, etc. in intermediate calculations. In theory, if you knew what your editing pipeline is, one could mathematically derive the starting headroom required to end up with acceptable error, but in practice that's probably a very risky proposition because 1) you don't know every intermediate calculation going on in every piece of software and 2) errors grow in highly non-linear ways, so even a small change in your pipeline might cause a large change in required headroom [1]. [1] https://en.wikipedia.org/wiki/Numerical_analysis#Generation_and_propagation_of_errors https://en.wikipedia.org/wiki/Numerical_analysis#Generation_...

pmoriarty 10y ago
Would using fixed-point arithmetic[1] help with these issues at all? [1] - https://en.wikipedia.org/wiki/Fixed-point_arithmetic https://en.wikipedia.org/wiki/Fixed-point_arithmetic
- smallnamespace 10y ago
  Fixed-point arithmetic essentially gives you more resolution at the cost of not having a dynamic range, so in practice you reduce some round-off errors, but overflow errors are much more likely. Overflow ends up as saturation in the context of audio editing and usually sounds like really bad clipping. In practice, fixed-point or floating-point is better depends on the operation you plan to use and whether you have a good idea whether you'll be able to stay in a fixed range ahead of time [1]. [1] The loss of 'footroom bits' corresponds to the extra resolution lost whenever the FP mantissa increases - http://www.tvtechnology.com/audio/0014/fixedpoint-vs-floatingpoint-numbers-in-audio-processing/266375 http://www.tvtechnology.com/audio/0014/fixedpoint-vs-floatin...
semi-extrinsic 10y ago
Yes, that's exactly what I was trying to ask. Does OP really need 24 bits because they are producing music professionally? In that case, $100 on the i7->Xeon upgrade is a cheap no-brainer. If they are not producing audio professionally, I'm saying they probably can't tell the difference between 16bit and 24bit audio. Also, you don't cause errors when you compress and decompress using a lossless compression format (as the name sort of implies).
- clarry 10y ago
  > If they are not producing audio professionally, I'm saying they probably can't tell the difference between 16bit and 24bit audio. That's not how it works. You can totally hear clipping or noise even as an amateur. Just as an amateur photographer first thing first learns to look out for overlit skies and hilights that aren't really recoverable at all unless they perhaps have access to the camera's RAW files. But unlike in photography, in sound work you typically need to process as well as sum multiple recordings, all while not blowing out the peaks that might just happen to line up. If anything, dealing with the large dynamic range of 24-bit is easier for the amateur. An experienced pro would probably have a good bunch of tricks in his bag should he really have to mix music in 16 bits, enough to produce something that doesn't sound horrible. The amateur would be more likely to struggle.
  - semi-extrinsic 10y ago
    I understand in OPs case, someone else (being a professional) has already recorded and mastered the sound of a real live pipe organ. That person of course benefits from working at 24bit, but once he's done mastering, OP is "just" playing back the mastered audio and should be fine with 16bits, I think.
    - clarry 10y ago
      But OP is not just playing back a single mastered recording, it's thousands of simultaneous samples that get mixed together live, presumably. In that case the professional who recorded all these samples can do nothing to prevent the peaks from lining up during playback.

pmoriarty 10y ago

Would using fixed-point arithmetic[1] help with these issues at all? [1] - https://en.wikipedia.org/wiki/Fixed-point_arithmetic https://en.wikipedia.org/wiki/Fixed-point_arithmetic

smallnamespace 10y ago
Fixed-point arithmetic essentially gives you more resolution at the cost of not having a dynamic range, so in practice you reduce some round-off errors, but overflow errors are much more likely. Overflow ends up as saturation in the context of audio editing and usually sounds like really bad clipping. In practice, fixed-point or floating-point is better depends on the operation you plan to use and whether you have a good idea whether you'll be able to stay in a fixed range ahead of time [1]. [1] The loss of 'footroom bits' corresponds to the extra resolution lost whenever the FP mantissa increases - http://www.tvtechnology.com/audio/0014/fixedpoint-vs-floatingpoint-numbers-in-audio-processing/266375 http://www.tvtechnology.com/audio/0014/fixedpoint-vs-floatin...

semi-extrinsic 10y ago

Yes, that's exactly what I was trying to ask. Does OP really need 24 bits because they are producing music professionally? In that case, $100 on the i7->Xeon upgrade is a cheap no-brainer. If they are not producing audio professionally, I'm saying they probably can't tell the difference between 16bit and 24bit audio. Also, you don't cause errors when you compress and decompress using a lossless compression format (as the name sort of implies).

clarry 10y ago
> If they are not producing audio professionally, I'm saying they probably can't tell the difference between 16bit and 24bit audio. That's not how it works. You can totally hear clipping or noise even as an amateur. Just as an amateur photographer first thing first learns to look out for overlit skies and hilights that aren't really recoverable at all unless they perhaps have access to the camera's RAW files. But unlike in photography, in sound work you typically need to process as well as sum multiple recordings, all while not blowing out the peaks that might just happen to line up. If anything, dealing with the large dynamic range of 24-bit is easier for the amateur. An experienced pro would probably have a good bunch of tricks in his bag should he really have to mix music in 16 bits, enough to produce something that doesn't sound horrible. The amateur would be more likely to struggle.
- semi-extrinsic 10y ago
  I understand in OPs case, someone else (being a professional) has already recorded and mastered the sound of a real live pipe organ. That person of course benefits from working at 24bit, but once he's done mastering, OP is "just" playing back the mastered audio and should be fine with 16bits, I think.
  - clarry 10y ago
    But OP is not just playing back a single mastered recording, it's thousands of simultaneous samples that get mixed together live, presumably. In that case the professional who recorded all these samples can do nothing to prevent the peaks from lining up during playback.