AIOGCAP ioctl reports software-emulated formats. It defaults to on. People
who use performance-sensitive audio software and do not want it to pick a
software-emulated audio format instead of one supported by their hardware
should turn it off.
This unbreaks isdnphone(1) on systems with PCM-only sound cards.
Approved by: cg
* be more specific in verbose boot messages
* allow the feeder subsystem to veto pcm* attaching if there is an error
initialising the root feeder
* don't free/malloc a new tmpbuf when resizing a snd_dbuf to the same size as
it currently is
* store the feeder description in the feeder structure instead of mallocing
space for it
extensively as none of my testboxes have speakers or an audio source at
present, but the chains built look correct and reading /dev/audio (ulaw,
translated from signed 16 bit little-endian) gives values within the
expected range for silence.
Note ALL MODULES MUST BE RECOMPILED
make the kernel aware that there are smaller units of scheduling than the
process. (but only allow one thread per process at this time).
This is functionally equivalent to teh previousl -current except
that there is a thread associated with each process.
Sorry john! (your next MFC will be a doosie!)
Reviewed by: peter@freebsd.org, dillon@freebsd.org
X-MFC after: ha ha ha ha
* 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.
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.
change channel interface - kobj implementation coming soonish
make pcm_makelinks not panic if modular
add pcm_unregister()
these changes support newpcm kld unloading, but this is only implemented
by ds1.c
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.