Thank you for both of your replies, Jean Marc and Stefan. Sorry that my
knowledge about the linux sound systems is quite limited, I just have some
knowledge in signal processing so I thought it would be helpful to point out
the problem. And I perfectly understand Jean-Marcs answer that it does not
directly relate to the codecs themselves, it is located before and after the
codec.
Stefan do I understand right that when I use dmix I cannot use the sample
conversion option you specify? Do I need to put exactly the option you state or
replace the argument "samplerate" by the corresponding rate number?
I have edited the alsa.conf file and seen that there are options relative to
dmix (defaults.pcm.dmix_rate 48000, so I guess I do use it?) fixing the sample
rate at 48kHz. I inserted your conversion option and noticed that this had no
effect after restarting alsa. Neither did changing the dmix rate to
defaults.pcm.dmix_rate 8000. And I guess in this case anyway, ALL of my sound
would be at 8kHz? So this does not look like an ideal workaround. And as long
as ALSA does linear interpolation (it actually sounds more like next neighbour
sampling) could this be a feature request for upcoming ekiga releases?
Once again thanks for your answers and neurons
Frank
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