Is it possible to stretch 16 bits symmetrically?
Reply #206 – 2014-05-08 14:38:00
Question is anyone willing to show me a dsp/method that does this multiplication and will do it upto 192dB vst/foobar? If one does not exist then how easy would it be to make one? By the time the samples hit any plugin they have already been converted to 32-bit floating point samples by foobar2000 automatically. The plugin gets 32-bit floating point, usually processes them with at least 32-bit floating point precision and outputs 32-bit floating point samples. As such, any plugin could achieve a much higher dynamic range than 192 dB.Don't tell me 48 thousand adjustments per second is not enough to reproduce sound. With your DAC it isn't, for high fidelity anyway. Don't get me wrong. You can wonderfully store audible frequencies even with 44.1 kHz, but when it comes to reproduction your non-oversampling DAC will not perform very well. If you want to reconstruct a sampled waveform then you need to interpolate (that's basic knowledge, see sampling theorem). You just increase the sampling rate by less than 9%, so the DAC will still very much follow the samples and output square waves. Without knowing the details of your DAC, I cannot tell if it has steep and complex analog low pass filters or not. If it doesn't then you get square waves. If it does then you have lots of phase shift even in the audible range.The fact that there will now be content beyond 44.1 upto 48k (added by the last dithering down stage at output), dismisses ultrasonic/aliasing issues altogether - which by the way, in playback systems were only ever associated with the problematic 44.1 from the get go - which is very true only on the playback side - similar terms, different methods get confused with recording side). No, it does not dismiss those issues. Even if you have a perfect resampler that removes images from 22.05 to 24 kHz, you said yourself that you are adding shaped noise again. Also, images will still be created at ~26 kHz upward. Their suppression will depends on the analog low pass filters of your DAC. This is also not only a D/A-conversion problem, see sampling theorem.Oversampling in playback systems would not be needed at all if CDs were 48000 from the start I believe. Well, you're wrong. Oversampling is still used with even 96 kHz and higher for various reasons, like making the analog low pass filters in the DAC very simple and cheap, reducing phase shift in the audible range, ...