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

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.