That's a clear need IMO, but it'd be slightly better if the game could have 48 kHz audo files and downsampled them to 44.1 kHz playback than the other way around (better to downsample than upsample).
44.1kHz sampling is sufficient to perfectly describe all analog waves with no frequency component above 22050Hz, which is substantially above human hearing. You can then upsample this band limited signal (0-22050Hz) to any sampling rate you wish, perfectly, because the 44.1kHz sampling is lossless with respect to the analog waveform. (The 16 bits per sample is not, though for the purposes of human hearing it is sufficient for 99% of use cases.)
22050 Hz is an ideal unreachable limit, like the speed of light for velocities.
You cannot make filters that would stop everything above 22050 Hz and pass everything below. You can barely make very expensive analog filters that pass everything below 20 kHz while stopping everything above 22 kHz.
Many early CD recordings used cheaper filters with a pass-band smaller than 20 kHz.
For 48 kHz it is much easier to make filters that pass 20 kHz and whose output falls gradually until 24 kHz, but it is still not easy.
Modern audio equipment circumvents this problem by sampling at much higher frequencies, e.g. at least 96 kHz or 192 kHz, which allows much cheaper analog filters that pass 20 kHz but which do not attenuate well enough the higher frequencies, then using digital filters to remove everything above 20 kHz that has passed through the analog filters, and then downsampling to 48 kHz.
The original CD sampling frequency of 44.1 kHz was very tight, despite the high cost of the required filters, because at that time, making 16-bit ADCs and DACs for a higher sampling frequency was even more difficult and expensive. Today, making a 24-bit ADC sampling at 192 kHz is much simpler and cheaper than making an audio anti-aliasing filter for 44.1 kHz.
The analog source is never perfectly limited to 20 kHz because very steep filters are expensive and they may also degrade the signal in other ways, because their transient response is not completely constrained by their amplitude-frequency characteristic.
This is especially true for older recordings, because for most newer recordings the analog filters are much less steep, but this is compensated by using a much higher sampling frequency than needed for the audio bandwidth, followed by digital filters, where it is much easier to obtain a steep characteristic without distorting the signal.
Therefore, normally it is much safer to upsample a 44.1 kHz signal to 48 kHz, than to downsample 48 kHz to 44.1 kHz, because in the latter case the source signal may have components above 22 kHz that have not been filtered enough before sampling (because the higher sampling frequency had allowed the use of cheaper filters) and which will become aliased to audible frequencies after downsampling.
Fortunately, you almost always want to upsample 44.1 kHz to 48 kHz, not the reverse, and this should always be safe, even when you do not know how the original analog signal had been processed.
yeah but you can record it in 96kHz, then resample it perfectly to 44.1 (hell, even just 40) in digital domain, then resample it back to 48kHz before sending to DAC
If you have such a source sampled at a frequency high enough above the audio range, then through a combination of digital filtering and resampling you can obtain pretty much any desired output sampling frequency.
the point is that when down sampling from 48 to 44.1 you can for "free" do the filtering since the down sampling is being done digitally with an fft anyway
https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampli...