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I use a sound sampler (Hauptwerk) that loads tens of thousand of wavefiles and needs a polyphony (simultaneous samples being played) of thousands of voices to s

I use a sound sampler (Hauptwerk) that loads tens of thousand of wavefiles and needs a polyphony (simultaneous samples being played) of thousands of voices to simulate a pipe organ. The minimum configuration for the smallest instrument (one voice) with natural acoustics (ie. original ambient reveberation recorded) is 4gb. The normal requirements for some good instruments are 64-128gb for 24-bit 6-channel uncompressed audio.

semi-extrinsic 10y ago
> 24-bit (...) uncompressed audio Do you really need this much headroom, e.g. for producing music professionally? (If so, why are you arguing over the $50-100 extra for a Xeon?) Because for listening, it's proven fact that even professional musicians and mixing engineers using equipment costing >$100k can't tell 24 bit from 16 bit in a proper double-blind A/B test. Presumably you use a high sample rate as well (maybe 192 kHz), which is equally useless. Drop both of those, and you're within 64GB easily with no noticable SQ loss.
- smallnamespace 10y ago
  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.
- TheOtherHobbes 10y ago
  Most people who believe this is a proven fact have never worked in pro audio and know a lot less about it than they assume. While people here like to quote that infamous looks-like-science-but-isn't Xiph video, the reality is that a lot of professional engineers absolutely can hear the difference. If you have good studio monitors and you know what to listen for the difference between a 24-bit master and a downsampled and dithered 16-bit CD master is very obvious indeed, and there are peer reviewed papers that explain why. Dither was developed precisely because naively interpolated or truncated 16-bit audio is audibly inferior to a 24-bit source. Many people certainly can hear the effect of dither on half-decent hardware, even though it's usually only applied to the LSB of 16-bit data. From a naive understanding of audio a single bit difference should be completely inaudible, because it's really, really quiet. From a more informed view it isn't hard to hear at all - and there are good technical reasons for that. For synthesis, you don't want dither. You want as much bit depth as possible so you can choose between dynamic range and low-level detail. So 24-bit data is the industry standard for orchestral samples.
  - spamizbad 10y ago
    There are professional audio "engineers" who claim they can hear the difference in the types of audio cables used. And when when challenge them on it, rather than defend it, or demonstrate it, they'll just name-drop musicians they've worked with... most of which are producing albums these days that sound like garbage because of the loudness war. Their words mean very little. I totally get going for 24 bit over 16 for the noise floor, though.
    - gtani 10y ago
      I knew some audio engineers who had different cables (cardas copper, pure silver, some others i can't remember) on identical headphones (Beyerdynamic) and everybody agreed you could hear not insignificant differences that they would describe as sharper transients, tubelike saturation etc. I consider that they were pro "engineers", they designed hardware and software for Avid.
      - 10y ago
- emersonrsantos 10y ago
  > Because for listening, it's proven fact that even professional musicians and mixing engineers using equipment costing >$100k can't tell 24 bit from 16 bit in a proper double-blind A/B test. Digital samples can and will generate intermodulation distortion, quantization and other audio artifacts when mixed. Using 24-bit lessens or eliminates the effect versus 16-bit. Experiments and results on this software are here: http://www.sonusparadisi.cz/en/blog/do-not-load-in-16-bit/ http://www.sonusparadisi.cz/en/blog/do-not-load-in-16-bit/
- jdietrich 10y ago
  Yes. Bit depth is almost entirely about dynamic range. Run out of bits at the top of the range and you get clipping; run out of bits at the bottom end and the signal disappears below the noise floor of dithering. For mixed and mastered music, the ~120dB perceived dynamic range of properly dithered 16-bit audio is more than sufficient. If you're listening at sufficient volume for the dither noise floor to be higher than the ambient noise floor of the room, you'll deafen yourself in minutes. For production use, the extra dynamic range of 24 bit recording is invaluable. You're dealing with unpredictable signal levels and often applying large amounts of gain, so you'll run into the limits of 16-bit recording fairly often. In a professional context, noise is unacceptable and clipping is completely disastrous. Most DAW software mixes signals at 64 bits - the computational overhead is marginal, it minimises rounding errors and it frees the operator from having to worry about gain staging. You can probably get away with 16 bit recordings for a sample library, but it's completely unnecessary. Modern sampling engines use direct-from-disk streaming, so only a tiny fraction of each sample is stored in RAM to account for disk latency. The software used by OP (Hauptwerk) is inexplicably inefficient, because it loads all samples into RAM when an instrument is loaded.

semi-extrinsic 10y ago

> 24-bit (...) uncompressed audio Do you really need this much headroom, e.g. for producing music professionally? (If so, why are you arguing over the $50-100 extra for a Xeon?) Because for listening, it's proven fact that even professional musicians and mixing engineers using equipment costing >$100k can't tell 24 bit from 16 bit in a proper double-blind A/B test. Presumably you use a high sample rate as well (maybe 192 kHz), which is equally useless. Drop both of those, and you're within 64GB easily with no noticable SQ loss.

smallnamespace 10y ago
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.
TheOtherHobbes 10y ago
Most people who believe this is a proven fact have never worked in pro audio and know a lot less about it than they assume. While people here like to quote that infamous looks-like-science-but-isn't Xiph video, the reality is that a lot of professional engineers absolutely can hear the difference. If you have good studio monitors and you know what to listen for the difference between a 24-bit master and a downsampled and dithered 16-bit CD master is very obvious indeed, and there are peer reviewed papers that explain why. Dither was developed precisely because naively interpolated or truncated 16-bit audio is audibly inferior to a 24-bit source. Many people certainly can hear the effect of dither on half-decent hardware, even though it's usually only applied to the LSB of 16-bit data. From a naive understanding of audio a single bit difference should be completely inaudible, because it's really, really quiet. From a more informed view it isn't hard to hear at all - and there are good technical reasons for that. For synthesis, you don't want dither. You want as much bit depth as possible so you can choose between dynamic range and low-level detail. So 24-bit data is the industry standard for orchestral samples.
- spamizbad 10y ago
  There are professional audio "engineers" who claim they can hear the difference in the types of audio cables used. And when when challenge them on it, rather than defend it, or demonstrate it, they'll just name-drop musicians they've worked with... most of which are producing albums these days that sound like garbage because of the loudness war. Their words mean very little. I totally get going for 24 bit over 16 for the noise floor, though.
  - gtani 10y ago
    I knew some audio engineers who had different cables (cardas copper, pure silver, some others i can't remember) on identical headphones (Beyerdynamic) and everybody agreed you could hear not insignificant differences that they would describe as sharper transients, tubelike saturation etc. I consider that they were pro "engineers", they designed hardware and software for Avid.
    - 10y ago
emersonrsantos 10y ago
> Because for listening, it's proven fact that even professional musicians and mixing engineers using equipment costing >$100k can't tell 24 bit from 16 bit in a proper double-blind A/B test. Digital samples can and will generate intermodulation distortion, quantization and other audio artifacts when mixed. Using 24-bit lessens or eliminates the effect versus 16-bit. Experiments and results on this software are here: http://www.sonusparadisi.cz/en/blog/do-not-load-in-16-bit/ http://www.sonusparadisi.cz/en/blog/do-not-load-in-16-bit/
jdietrich 10y ago
Yes. Bit depth is almost entirely about dynamic range. Run out of bits at the top of the range and you get clipping; run out of bits at the bottom end and the signal disappears below the noise floor of dithering. For mixed and mastered music, the ~120dB perceived dynamic range of properly dithered 16-bit audio is more than sufficient. If you're listening at sufficient volume for the dither noise floor to be higher than the ambient noise floor of the room, you'll deafen yourself in minutes. For production use, the extra dynamic range of 24 bit recording is invaluable. You're dealing with unpredictable signal levels and often applying large amounts of gain, so you'll run into the limits of 16-bit recording fairly often. In a professional context, noise is unacceptable and clipping is completely disastrous. Most DAW software mixes signals at 64 bits - the computational overhead is marginal, it minimises rounding errors and it frees the operator from having to worry about gain staging. You can probably get away with 16 bit recordings for a sample library, but it's completely unnecessary. Modern sampling engines use direct-from-disk streaming, so only a tiny fraction of each sample is stored in RAM to account for disk latency. The software used by OP (Hauptwerk) is inexplicably inefficient, because it loads all samples into RAM when an instrument is loaded.