[linux-audio-dev] Looking for fast integer resampling code

[Dave Chapman]

Hi,

I'm working on the Rockbox - http://www.rockbox.org - project, which is
a project to develop an open source (GPL'd) replacement firmware for
portable digital audio p[ayers.

Rockbox has been running on various Archos MP3 players for about 3
years.  The latest hardware that Rockbox is in the process of being
ported to is the iRiver H120/H140 players.  These consist of a 120MHz
Coldfire (m68k-based) processor and 32MB of RAM.  There is no
floating-point support in the Coldfire.

We currently have real-time playback of MPEG audio (via libmad), Ogg
Vorbis (via libOgg), FLAC (via libFLAC), A52/AC3 (via liba52) and (of
course) WAV files.

Unfortunately, the audio hardware only supports a 44.1KHz samplerate -
meaning that we have to resample other frequencies.

The most common case will be downsampling 48KHz->44.1KHz, but there are
also users who need upsampling from various lower frequencies (e.g. for
use with audio books).

So we're looking for pointers to any high quality (but fast!) resampling
code that could be used.

Thanks in advance,

Dave.