Hello.
In alsa-lib-1.0.23/src/pcm/pcm_plug.c, there are two assertions of the form:
assert(snd_pcm_format_linear(slv->format));
in snd_pcm_plug_change_rate() and snd_pcm_plug_change_channels()
functions, respectively. While in snd_pcm_plug_change_rate() this looks
reasonable, it is IMHO invalid in snd_pcm_plug_change_channels(). I say
this because this can be triggered with the following .asoundrc:
pcm.jackplug {
type plug
slave.pcm {
type jack
playback_ports {
0 system:playback_1
1 system:playback_2
}
capture_ports {
0 system:capture_1
1 system:capture_2
}
client_name "alsa"
}
}
and the following command:
aplay -r 44100 -c 1 -f FLOAT_LE -D jackplug /dev/zero
JACK is configured to accept stereo audio, with 44100 Hz sampling rate,
so only the number of channels has to be changed by the plug here.
I.e., rerouting channels by copying samples makes perfect sense not only
for linear formats, but also for floating-point ones.
--
Alexander E. Patrakov
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