The human ear can hear sounds up to ~ 20KHz. Some ADC's oversample. For example, to capture frequencies up to 20KHz, an ADC converter might sample at 160KHz. The steps are: 1. Filter the audio with an analog low-pass filter. 2. Sample at 160KHz. 3. Digitally filter the audio with a low-pass filter. 4. Downsample to a lower sample rate.
I understand that it's difficult to implement low-pass analog filters with sharp cutoffs: required to maximize the available bandwidth without exceeding the Nyquist limit, which would otherwise result in aliasing from folded frequencies. Therefore, I might choose to oversample at 160KHz, with an low-pass analog filter at 20KHz that can rolloff in 30KHz and still be below the Nyquist limit at 80Khz.
However, once I've sampled at 160KHz, why would I want to apply a digital low-pass filter before decimation? What are the advantages of this?