Sample rate conversion Options

[MannyCalavera]

Hello everyone.

I have a question regarding sample rate conversion algorightms.
I don't know how they work, but I guess you guys are more familiar with this

Let's say I have a 48 kHz wave file that I want to use in something I have to mix
with some 96 kHz material at certain places. This requires the 48 kHz file be resampled
and in the end back to 48.

So in short: does resampling 48 --> 96 --> 48 change the original or can the original
be restored 100% ?

I'm using Wavelab for this.

[Wombat]

Although you can never make a perfect sinc filter, its possible to build ones very very close to perfect, such that any difference is below quantization noise except for a tiny region right around the Nyquist limit, which of course won't include audio anyway because of the anti-alias filter on the original recording ADC.

I am pretty sure we can get close but from reading some earlier posts in this thread it is suggested that 2x upsampling will leave the source material intact which it is not. One can argue about such ultra-close filters have huge amounts of pre-ringing btw.
It is not a lossless operation.
I see it in the context of some audiophile claims about upsampling 2x from 44.1 to 88.2 for example sounds better as to 96kHz cause of only applying zeros on every second bit, but it isn´t.

This post has been edited by Wombat: Apr 14 2011, 01:15

[saratoga]

Although you can never make a perfect sinc filter, its possible to build ones very very close to perfect, such that any difference is below quantization noise except for a tiny region right around the Nyquist limit, which of course won't include audio anyway because of the anti-alias filter on the original recording ADC.

I am pretty sure we can get close but from reading some earlier posts in this thread it is suggested that 2x upsampling will leave the source material intact which it is not.

Obviously if the filter does not have unity transmittance at every frequency, then at least some samples cannot be the same . . .

[bandpass]

Obviously if the filter does not have unity transmittance at every frequency, then at least some samples cannot be the same . . .

Indeed, with 2× resampling, if the input signal frequency stays below the filter transition, half of the output samples (weird/asymmetric filters notwithstanding) will be the same as the input samples (as observed at the chosen bit depth). With sox, 16-bit, 48kHz ->96kHz, the odd-numbered output samples are bit-exact up to 22kHz; at 23kHz, the roll-off kicks in and the samples are no longer the same. At 24-bit, it's bit-exact to 16k; at 17k, attenuation of 0.0001dB is evident in the output.

[Axon]

Obviously if the filter does not have unity transmittance at every frequency, then at least some samples cannot be the same . . .

Indeed, with 2× resampling, if the input signal frequency stays below the filter transition, half of the output samples (weird/asymmetric filters notwithstanding) will be the same as the input samples (as observed at the chosen bit depth). With sox, 16-bit, 48kHz ->96kHz, the odd-numbered output samples are bit-exact up to 22kHz; at 23kHz, the roll-off kicks in and the samples are no longer the same. At 24-bit, it's bit-exact to 16k; at 17k, attenuation of 0.0001dB is evident in the output.

Strictly speaking, it is not very meaningful to describe the flatness of a frequency response as "bit-exact".