update code dealing with snd_dbuf objects to do so using a functional interface

modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will

update code dealing with snd_dbuf objects to do so using a functional interface

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.

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kobjify.

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 compat

kobjify.

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.

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add reinit functions to mixers
unstaticize chn_start()
add reset/resetdone functions to channels

Add suspend/resume hooks.

Submitted by: sean@stat.duke.edu

fix warnings

detach support
remove un-needed setdir functions
add bus_teardown_intr calls where necessary
destroy our dma tags where necessary
destroy ac97 before releasing resources

change mixer api slightly
change channel interface - kobj implementation coming soonish
make pcm_makelinks not panic if modular
add pcm_unregister()

these changes support newpcm kld unloading, but t

change mixer api slightly
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

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rework feeder sytem to allow feeders in klds

modify driver capability reporting format to list every audio format
seperately- required for above and because we could not previously indicate
that mon

rework feeder sytem to allow feeders in klds

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.

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don't complain about bad irqs if using a shared irq

This patch appears to solve the problems with recording resulting
in bogus hwptr warnings. Further comments sought on -hackers
were not forthcoming. I can only hope that calling DELAY() like
this doe

This patch appears to solve the problems with recording resulting
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.

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A couple of cosmetic fixes, plus a biggie: When checking the DMA
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

A couple of cosmetic fixes, plus a biggie: When checking the DMA
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.

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1. Increase the size of the DMA buffer.

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 o

1. Increase the size of the DMA buffer.

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.

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Undo some of my own damage. With this patch, it appears that both
recording and playback now work correctly.

Fix channel 1 playback the rest of the way. There are a couple of
hiccups, but playback now proceeds mostly normally using both channel 1
and 2.

Get playback on channel 1 closer to working. 8 bit samples work.
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

Get playback on channel 1 closer to working. 8 bit samples work.
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.

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Fix channel 1, sort of. recording now generates IRQs and the output
appears to be the correct length, but quality of output has not yet
been tested. Also, full duplex audio (that is, playback on chan

Fix channel 1, sort of. recording now generates IRQs and the output
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

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Reverse the sense of the signed/unsigned formatting bit.

This dramatically cleans up playback quality, at least with mxaudio.

almost-there ess solo-1 driver, committed so people can tell me why it
doesn't generate irqs.

- Merge from HEAD

Sound Mega-commit. Expect further cleanup until code freeze.

For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .

Summary of changes includ

Sound Mega-commit. Expect further cleanup until code freeze.

For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .

Summary of changes includes:

1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.

Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.

Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).

Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.

2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.

Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.

Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/

3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.

4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.

5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.

6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.

Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO

Manual page updates are on the way.

Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.

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Flush remaining malloc() cleanups (M_NOWAIT -> M_WAITOK).

Filter/compress the amount of channel trigger. This should reduce
much of lock/unlock contentions within the interrupt handler. Most
of these drivers only need PCMTRIG_START or STOP (ABORT).

Discuss

Filter/compress the amount of channel trigger. This should reduce
much of lock/unlock contentions within the interrupt handler. Most
of these drivers only need PCMTRIG_START or STOP (ABORT).

Discussed with: scottl

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Fix broken binary issues with latest gcc 4.x due to bitfield signess
mishaps for emu10k1 [1] and few other places.

Reported/Submitted/Tested by: Ed Schouten <ed@fxq.nl> [1]

sndbuf_alloc() now accept dmaflags argument which will be forwarded to
internal bus_dmammem_alloc() for greater flexibility on setting up DMA /
page attributes.

Fix severe out-of-bound mtx "type" pointer, causing WITNESS refcount
confusions and panic provided that the following conditions are met:

1) WITNESS is enabled (watch/trace).
2) Using modules, i

Fix severe out-of-bound mtx "type" pointer, causing WITNESS refcount
confusions and panic provided that the following conditions are met:

1) WITNESS is enabled (watch/trace).
2) Using modules, instead of statically linked (Not a strict
requirement, but easier to reproduce this way).
3) 2 or more modules share the same mtx type ("sound softc").
- They might share the same name (strcmp() == 0), but it always
point to different address.
4) Repetitive kldunload/load on any module that shares the same mtx
type (Not a strict requirement, but easier to reproduce this way).

Consider module A and module B:
- From enroll() - subr_witness.c:
* Load module A. Everything seems fine right now.
wA-w_refcount == 1 ; wA-w_name = "sound softc"
* Load module B.
* w->w_name == description will always fail.
("sound softc" from A and B point to different address).
* wA->w_refcount > 0 && strcmp(description, wA->w_name) == 0
* enroll() will return wA instead of returning (possibly unique)
wB.
wA->w_refcount++ , == 2.
* Unload module A, mtx_destroy(), wA->w_name become invalid,
but wA->w_refcount-- become 1 instead of 0. wA will not be
removed from witness list.
* Some other places call mtx_init(), iterating witness list,
found wA, failed on wA->w_name == description
* wA->w_refcount > 0 && strcmp(description, wA->w_name)
* Panic on strcmp() since wA->w_name no longer point to valid
address.

Note that this could happened in other places as well, not just sound
(eg. consider lots of drivers that share simmilar MTX_NETWORK_LOCK).

Solutions (for sound case):
1) Provide unique mtx type string for each mutex creation (chosen)
or
2) Put "sound softc" global variable somewhere and use it.

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