because they bogusly check for defined(INTR_MPSAFE) -- something which
never was a #define. Correct the definitions.
This make INTR_TYPE_AV finally get used instead of the lower-priority
INTR_TYPE_TTY, so it's quite possible some improvement will be had
on sound driver performance. It would also make all the drivers
marked INTR_MPSAFE actually run without Giant (which does seem to
work for me), but:
INTR_MPSAFE HAS BEEN REMOVED FROM EVERY SOUND DRIVER!
It needs to be re-added on a case-by-case basis since there is no one
who will vouch for which sound drivers, if any, willy actually operate
correctly without Giant, since there hasn't been testing because of
this bug disabling INTR_MPSAFE.
Found by: "Yuriy Tsibizov" <Yuriy.Tsibizov@gfk.ru>
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.
rewrite ess mixer to use native registers
rewrite play/rec code to use more accurate timer when available
add code to use audio2 for playback, but disable it as no irqs are generated
add support for non-pnp cards to sbc
move card identification to sbc
channel-swapping code is in sb now instead of dsp
vibra16x support is still broken, but will be fixed soon
note: sbc is now compulsory for sb cards
for pnp cards use:
device sbc0
for non-pnp cards eg:
device sbc0 at isa? port 0x240 irq 5 drq 3 flags 0x15
(hints as oldpcm)
both in addition to:
device pcm0
Reviewed by: tanimura,dfr
Said he liked it: peter
Also, optimize out a mess of #if's that were duplicating work already
done by config(8). For example, if a file is marked as
"dev/sound/pci/foo.c optional pcm pci" then it's only added if pcm *and*
pci are present, so #if NPCM > 0 and #if NPCI > 0 are totally redundant.
A bit more work is still needed.
Discussed with: cg (a few weeks ago)
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.
resource_list_release. This removes the dependancy on the
layout of ivars.
* Move set_resource, get_resource and delete_resource from
isa_if.m to bus_if.m.
* Simplify driver code by providing wrappers to those methods:
bus_set_resource(dev, type, rid, start, count);
bus_get_resource(dev, type, rid, startp, countp);
bus_get_resource_start(dev, type, rid);
bus_get_resource_count(dev, type, rid);
bus_delete_resource(dev, type, rid);
* Delete isa_get_rsrc and use bus_get_resource_start instead.
* Fix a stupid typo in isa_alloc_resource reported by Takahashi
Yoshihiro <nyan@FreeBSD.org>.
* Print a diagnostic message if we can't assign resources to a PnP
device.
* Change device_print_prettyname() so that it doesn't print
"(no driver assigned)-1" for anonymous devices.
