* add new channels to the end of the list so channels used in order of
addition
* de-globalise definition of struct snddev_info and provide accessor
functions where necessary.
* move the $FreeBSD$ tag in each .c file into a macro and allow the
/dev/sndstat handler to display these when set to maximum verbosity to aid
debugging.
* allow each device to register its own sndstat handler to reduce the amount
of groping sndstat must do in foreign structs.
worked before.
mixer, dsp and sndstat are seperate devices - give them their own cdevsws
instead of demuxing requests sent to a single cdevsw.
use the si_drv1/si_drv2 fields in dev_t structures for holding information
specific to an open instance of mixer/dsp.
nuke /dev/{dsp,dspW,audio}[0-9]* links - this functionality is now provided
using cloning.
various locking fixes.
instead of using two malloced arrays for storing channel lists, use an
slist. convert the sndstat device to use sbufs and optionally provide more
detail about channel state.
vchans are software mixed playback channels. they are not enabled by this
commit. they use the feeder infrastructure to emulate normal playback
channels in a manner transparent to applications, whilst providing as many
channels are desired, especially suitable for devices with only one hardware
playback channel. in the future they will provide additional features.
those wishing to test this functionality will need to add vchan.c to
sys/conf/files and use 'sysctl -w hw.snd.pcm0.vchans' to enable it.
blocksize and auto-rate selection are not yet supported.
this introduces a new buffering mechanism which results in dramatic
simplification of the channel manager.
as several structures have changed, we take the opportunity to move their
definitions into the source files where they are used, make them private and
de-typedef them.
the sound drivers are updated to use snd_setup_intr instead of
bus_setup_intr, and to comply with the de-typedefed structures.
the ac97, mixer and channel layers have been updated with finegrained
locking, as have some drivers- not all though. the rest will follow soon.
devices. opening /dev/{dsp,dspW,audio}0 and then opening a different device
from that list and closing it resulted in a panic when any operation is
performed on the first fd.
we prevent this happening by denying the second open unless it uses the same
minor device as the first.
PR: kern/25519
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
this gives us several benefits, including:
* easier extensibility- new optional methods can be added to
ac97/mixer/channel classes without having to fixup every driver.
* forward compatibility for drivers, provided no new mandatory methods are
added.
modify driver capability reporting format to list every audio format
seperately- required for above and because we could not previously indicate
that mono was unsupported.
there should be no functional impact.
opens if the reference count is not decremented on close.
Note that this may result in the reference count being corrupted
on full duplex devices (due to mismatching opens/closes), but the
code doesn't use the reference count for anything on full duplex
devices.
mutate some panics to kasserts
add more spl protection
PR: kern/14990
Partially Submitted by: Vladimir N.Silyaev <vns@delta.odessa.ua>
Reviewed by: dfr
Without this, ioctl commands for setting formats and speeds were
essentially ignored for simplex devices until the application actually
performed a read or write.
* Make sure that both channels are set in the SB mixer code and provide a
mixer table specifically for the ess18xx which supports the extended
accuracy available on this part.
* Fix a stupid bug in ess_format() which ignored the passed-in format and
changed the hardware based on the value which was set last time. This
meant that the hardware setting was often not set correctly at all.
* Add a custom identify driver for the ESS1888 which automagically detects
and adds the device in a pseudo-PnP way. This driver also emits the magic
sequence which enables the sound hardware after a hard reset, allowing
it to work correctly for the sound hardware of a PWS 433au (and probably
all other PWS class alpha machines).
With these changes, I was able to play back simple sounds on my 433au. I
have not tested recording or any other formats other than 8bit ulaw and
16bit stereo.
will have to mknod yourself for now.
* don't eat the first write()
* partial rvplayer fix- don't panic on unaligned writes unless our
feeder chain requires them for downconversion. a fuller fix is
on the way.
