Hello!
Do you want to reduce/increase the samplerate live? If so alsa just has
plug
devices. Those devices just convert the audio sent to it into one desired
samplingrate, that of the soundcard. It can work like this:
aplay -Dplughw:0,0 file.wav
or
aplay -Dplug:default file.wav
You can create such devices of type plug with more complicated settings in
your .asoundrc. But this is a little more advanced magic. :-)
If you don't need to do it live, if you just need a file converted into a
different samplerate take one of the following tools:
sox (also the package name on your distribution)
libsndfile (it originally comes with some tools) you'd also need libsamplerate
(libsrc a.k.a Secret Rabbit Code :-) )
With libsndfile there is libsndfile-resample (which has really better
sound-quality, if you need high-end audio).
Yes you can mix alsa capture and alsa playback. This is also a question of
alsa, but more a question of the software. Typically this is a harddisk
recording/digital audio workstation (DAW) software. For the bigger settings
I'd always use JACK (jack comes with all newer distributions I know. It's an
audio server for realtime. It just "hides" ALSA from you and gives you named
input and output ports. You start your software recorder: jack_capture for
simple things, audacity for nice graphic user interface (GUI) and not to
complicated work, ardour (for gui and big) or ecasound for no gui and still
rather big. What JACK does and offers is more, but you don't need to worry
about that now. You just start your application then use something like
jack_lsp and jack_connect to connect you software's inputs/outputs to the
soundcards outputs/inputs. It's like plugging virtual cables in a virtual
studio. You can also use qjackctl, a nice graphical application to wire
programs and soundcards.
I hope that helps. If you got more questions on this, please get back to us.
Kindest regards
Julien
--------
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