[linux-audio-user] resampling question
Erik de Castro Lopo
erikd-lad at mega-nerd.com
Tue Oct 7 18:39:00 EDT 2003
On Tue, 7 Oct 2003 23:51:07 +0300 (EEST)
Tommi Sakari Uimonen <tuimonen at cc.hut.fi> wrote:
> Hi. If I want to resample from 96khz to 48khz, what would give the best
> result: use libsamplerate, or just drop off odd (or even) samples?
>
> I guess the libsamplerate and any of its interpolations would cause some
> digital garbage anyway,
Sample rate conversion using libsamplerate's SRC_SINC_* converters gives a
signal to noise ratio of 96dB or better. Yes there is some digital garbage
but its probably less than the noise picked when an instrument is recorded
with a microphone.
> or are they intellectual enough to detect that the
> rate is halved and just perform a drop off?
On most normal signals, dropping every second sample will sound considerably
worse than what libsamplerate does. I suggest that you read up on sampling
and aliasing for an explanation of why.
Erik
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