Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma. At the moment, this is used for the
asynchronous busdma_swi and callback mechanism. Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg. dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create(). The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.
sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms. The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.
If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.
Reviewed by: tmm, gibbs
* 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.
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.
in bogus hwptr warnings. Further comments sought on -hackers
were not forthcoming. I can only hope that calling DELAY() like
this doesn't cause any heartburn.
position, channel 1's dma position register must be quiescent. So
the driver will spl, pause the DMA, delay a bit and hold as still as
possible while snapping the picture.
I'm sure there HAS to be a better way to do this, but if there is, it's
not documented.
So far as I can tell, this fixes recording, which means the Solo is open
for business.
2. Offer half duplex with both playback and record on channel 1 or
full duplex with playback always on channel 2 as a compile-time option.
3. 16 bit record output is byte swapped for some dumb reason. Report the _BE
AFMTs for recording.
16 bit samples have some sort of choppiness, the nature of which
is not completely clear, but it clearly has something to do with
dma buffer synchronization. But at least channel 1 makes noise now.
appears to be the correct length, but quality of output has not yet
been tested. Also, full duplex audio (that is, playback on channel 1)
does not yet work. Two constants and I am there!
Obtained from: major hints from ALSA
