I'd like to convert a few applications that now have their own sound
driver to use the standard sound driver. However, currently I can't, as
a few features are missing, namely the following:
- "true queue length": the app should be able to query the true queue
length,
including on-card FIFO buffers etc. (and possibly including codec
filter
delay)
- the application should be able to query and eventually set (within
bounds)
the wakeup latency (this is related to the fragment size)
- I need the possibility to switch between input and output (and vice
versa)
as fast as possible on half duplex cards, that means I need a
possibility
to suppress lengthy autocalibration cycles when switching. Not
necessarily
a new API element, could be implicit when read follows write and vice
versa.
- It would be nice if it was possible to tell the sound driver something
like
"only wake me up when there are at least N samples (or bytes) in the
receive queue"
and "only wake me up if there are more than N samples free in the
transmit queue
or the transmit queue contains less than M samples".
Can we agree on an API for this?
Tom
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