The higher the sample-rate, the better any approximations that assume that the sample-rate is infinite (or that logic propagation speed is infinite) will be.
That's interesting, so basically if you want to "remaster" a track :let's say denoising, eq, exciters, reverb, stereo ehancers , and why not slight loudness compression, it would be better to upsample first, and then downsample once the work is done ?
True, but the worse any approximations that assume that the sample-rate is say 44-48kHz will be.
... So having one plugin that works better (or exclusively) at a high rate is not a green light to run the whole chain at that rate—check the specs of each plugin.
non-linear effects, such as distortion, will almost always alias. Certain effects that seem linear are actually not, as fast modulation of parameters such as gain (for a compressor/expander) or filter cutoff (wah pedal) will introduce aliasing as well. Oversampling reduces aliasing in the audio band, as the aliased signal typical falls off 6dB/octave.
This is all very interesting, though a bit over my head. Let me ask another way: Considering only "typical" recording studio plug-in effects such as EQ and compression, is there a practical advantage to processing files at, say, 96 KHz rather than at 44.1 KHz? The reason I stress "practical" is because I just tried a simple test in Sound Forge using the Sonitus EQ plug-in bundled with Cakewalk SONAR I use, and I saw no added distortion
If your software is good, it should be upsampling when needed without you having to do it automatically.
Try repeating the test with white noise and some high frequency EQ and see if you see changes in the shape of the EQ curves.
So unless someone has an example that proves otherwise, I'll consider the notion that higher sample rates allow typical audio plug-ins to work better as just another audio myth.
You've tested a couple of DSP operations on extremely trivial single-tone samples and are now willing to make such a sweeping generalization?!?
Me, I'll be satisfied to hear a good explanation of why EQ works better on 96 KHz audio files than files at 44.1 KHz. I'll be even happier to see an FFT or other data, or even a pair of audio files to compare. Whaddya got? --Ethan
Me, I'll be satisfied to hear a good explanation of why EQ works better on 96 KHz audio files than files at 44.1 KHz.
Of course if the EQ is already properly designed for its application, changing it is unlikely to improve things.
Quote from: Ethan Winer on 12 June, 2012, 02:35:28 PMMe, I'll be satisfied to hear a good explanation of why EQ works better on 96 KHz audio files than files at 44.1 KHz.Resampling is accomplished by applying a filter. If you resample to a higher sampling rate, then apply an EQ, effectively the length of the filter in the EQ is expanded by the length of the filter in the resampler. Thus, you're essentially using a filter with more taps, which if properly designed (e.g. a good resampler), will give 'better' (or at least different) results then a filter with fewer taps. Of course if the EQ is already properly designed for its application, changing it is unlikely to improve things.