> Both your inputs (ADC) and outputs (DAC) are fixed-point. Why would you want to use a floating point in between?
Fixed point arithmetic is non-trivial and not well supported by CPU instruction sets.
(Hint: you can't just use integer add/multiply.)
> I think you’re underestimating the complexity of modern musician-targeted VST effects. I’ve never programmed that kind of software. But I’m positive such effects are overwhelmingly more complex than just multiply+add these sample values. Therefore, extra temporal resolution helps.
Indeed, many audio effects require upsampling to work well with common inputs, e.g highly non-linear effects like distortion/saturation or analog filter models.
However usually they perform upsampling and downsampling internally (commonly between 2x-4x-8x).
While upsampling/downsampling is expensive (especially if you are using multiple of these types of plugins) its not clear if running at a higher sample rate across the board is worth it just to save those steps.
Fixed point arithmetic is non-trivial and not well supported by CPU instruction sets. (Hint: you can't just use integer add/multiply.)
> I think you’re underestimating the complexity of modern musician-targeted VST effects. I’ve never programmed that kind of software. But I’m positive such effects are overwhelmingly more complex than just multiply+add these sample values. Therefore, extra temporal resolution helps.
Indeed, many audio effects require upsampling to work well with common inputs, e.g highly non-linear effects like distortion/saturation or analog filter models. However usually they perform upsampling and downsampling internally (commonly between 2x-4x-8x). While upsampling/downsampling is expensive (especially if you are using multiple of these types of plugins) its not clear if running at a higher sample rate across the board is worth it just to save those steps.