[Alsa-user] Good quality resampling, flash and amarok

[Samtek <sam...@gm...>]

I have been trying to enable the best possible resampling quality to go from my
44100Hz music to my 48000Hz sound card. I am using the onboard sound on my asus
P5B using a AD1988 chip (according to alsamixer) and alsa version 1.0.16. I have
created a .asoundrc with the following:

defaults.pcm.rate_converter "samplerate_best"

This works fine with mplayer and aplay and I can see the expected increase in
CPU usage when playing 44100Hz content with these players. However when I have
the above in my .asoundrc any flash content in firefox causes flash to crash
with (I am using archlinux x64 with nspluginwrapper):

npviewer.bin[26668]: segfault at 0 ip f49fac14 sp ffab39d0 error 4 in
libasound.so.2.0.0[f49ad000+c1000]

I would also like to know what I should set amarok's output engine to or its
xine-config file to force it to use this resampling method. 

When using the udial sound from:
http://www.hydrogenaudio.org/forums/index.php?showtopic=9772
I get distortion using amarok's default settings but using front or hw:0 in the
alsa device settings fixes this. However the CPU usage is too low for it to be
using the best quality alsa resampling.

Even if you can't help me fix this please help me understand what the heck is
going on!

Thanks in advance