Vinyl -> 24/96: Rip once, rip right, never rip again
Reply #20 – 2006-10-25 02:20:46
If one records at 88.2 or higher, and then resamples to 44.1, there is (1) less energy in the alias images to begin with and (2) very little energy indeed by the time it reaches back to audible frequencies. Therefore, recording at 44.1 will contain this distortion (even if only with HF harmonic distortion as the source), and recording at a higher sample rate will eliminate most of it. A fair number of professionals who record live at higher sample rates have made statements to the effect that its all about alias distortion, not capturing higher frequency content. I'd strongly agree. For proper sampling you simply must not have present any frequencies above the Nyquist limit in the analogue signal that is passed to the sampler. This requires an analogue filter of some kind, and if you wish to preserve audible frequencies only about 10% below the Nyquist limit it's very difficult to filter well. Otherwise, if you sample at 44.1 kSa/s (kilosamples per second) your Nyquist limit being 44.1/2 = 22.05 kHz, a frequency above 22.05 kHz would be mirrored to a frequency below 22.05 kHz. Just imagine a 30 kHz sine wave that wasn't filtered being sampled at 44.1 kSa/s. At 7.95 kHz above the Nyquist limit it can be shown that it is indistinguishable from a sinusoid 7.95 kHz below the Nyquist limit. When playing back the waveform, the reconstruction filter has to assume it's below 22.05 kHz, so it comes out as 14.1 kHz (a mirrored alias tone), which is an audible frequency. The filter in a soundcard capable of 88.2 or 96.0 kSa/s is likely to be a good filter with fair response well above 22.05 kHz but very little at 40 kHz or more. I don't think it's likely that many or any soundcards would implement a different pre-sampling electronic filter for each sampling rate, so only the higher sampling rates are likely to be sampled properly according to Nyquist's theorem. You want the best quality and to do it only once, so respect Nyquist diligently and allow ample headroom to avoid clipping by using 24-bit sampling and not worrying if the peaks in your waveform are even only 1/4 or 1/2 of full-scale. You can use Replygain's Album Gain mode to achieve normal loudness for playback after you've made your recording. When you later downsample to 44.1 kHz, a brickwall pre-filter must be applied before generating the samples at the lower rate, as part of the algorithm. SSRC and the equivalent downsampler in Foobar2000 will certainly do this, as will good quality audio editing software. Foobar2000 has excellent DSP resampler and dithering algorithms for optimum conversion of your lossless high-sampling rate, high bit-depth files to 16-bit, 44.1 kSa/s for burning to CD. With vinyl sources, I'd say that "no noise shaping" dither is ideal when converting to 16-bit.