Copy stable/10@r272459 to releng/10.1 as part of
the 10.1-RELEASE process.

Approved by: re (implicit)
Sponsored by: The FreeBSD Foundation

Copy head (r256279) to stable/10 as part of the 10.0-RELEASE cycle.

Approved by: re (implicit)
Sponsored by: The FreeBSD Foundation

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.

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).

Discussed with: scottl

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

MFp4 (114068):
Use bus_get_dma_tag() to obtain the parent DMA tag to make the drivers
a little bit more non-ia32/amd64 friendly.

There is no man page for bus_get_dma_tag, so this is modelled after
rev. 1.62 of src/sys/dev/sound/pci/es137x.c by marius.

Inspired by: commit by marius

forced commit: previous commit was CID 483

A pointer was checked for NULL after dereferencing it. The check is not
needed here, except there's a bug which results in detaching the device
twice.

Move the NULL pointer check to the beginning of the function and convert
it into a KASSERT.

CID: 420
Found with: Coverity Prevent(tm)
Discussed with: ariff
MFC after: 5 days

Power up the external amplifiers additionally to powering up the DAC and ADC.

PR: 47029
Submitted by: Anish Mistry <mistry.7@osu.edu>
Tested by: David Murphy <dm@dmz.ie>

Implement suspend/resume operation for snd_csa(4)

PR: kern/82243
Submitted by: Serge Semenenko
Approved by: re (scottl)
MFC after: 1 month

Start each of the license/copyright comments with /*-, minor shuffle of lines

Audio drivers failed to detect failure condition and attempted to
assign DMA address to the wrong address. It can cause system lockup
or other mysterious errors. Since most sound cards requires low DMA
address(BUS_SPACE_MAXADDR_24BIT) sndbuf_alloc() would fail when the
audio driver is loaded after long running of operations.

Approved by: jake (mentor)
Reviewed by: truckman, matk

Rename the sound device drivers:

- `sound'
The generic sound driver, always required.

- `snd_*'
Device-dependent drivers, named after the sound module names.
Configure accordingly to your hardware.

In addition, rename the `snd_pcm' module to `sound' in order to sync
with the driver names.

Suggested by: cg

The newpcm headers currently #define away INTR_MPSAFE and INTR_TYPE_AV
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>

Convert callers to the new bus_alloc_resource_any(9) API.

Submitted by: Mark Santcroos <marks@ripe.net>
Reviewed by: imp, dfr, bde

Augment /dev/sndstat with the module names, if applicable.

Approved by: tanimura (mentor)

Use PCIR_BAR(x) instead of PCIR_MAPS.

Glanced over by: imp, gibbs
Tested by: i386 LINT

Prefer new location of pci include files (which have only been in the
tree for two or more years now), except in a few places where there's
code to be compatible with older versions of FreeBSD.

Mega busdma API commit.

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

Implement a "sndbuf_getbufaddr" function and use it instead of vtophys().

Reviewed by: orion

many changes:

* 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.

the turtle beach santacruz implements ac97 eapd in an inverted sense -
inform the ac97 code of this.

Use the M_ZERO flag to malloc(9)

Reviewed by: cg
MFC after: 1 week

use a global devclass for all drivers - i'm not entirely sure why this
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.

update this driver to use new firmware and incorporate many fixes.

this works on cs4630 chips, and should implement the clkrun hack for
thinkpads- this will display diagnostic messages when triggered until its
correctness is established.

mega-commit.

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.

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.

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.

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 this is only implemented
by ds1.c

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.

add module metadata. this is a hack, sound drivers will eventually present a
bus to which pcm, mixer, etc will attach.

handle emulated dma reads

don't try to get sample size from snd_dbuf

fail in attach if we seem to have no ac97 codec

update the ac97 layer:
* add a callback for initialising the mixer interface
* support ac97 2.1 variable rate audio feature

fix ac97-using drivers for the above

add suspend/resume support for neomagic

The pointer of the serial FIFO in the DSP seems to make an error
upon starting/stopping the DSP. Correct the pointer in addition
to clearing the FIFO.

update ac97 layer to use device_printf when printing messages

Do not start/stop DMA transfer if it is already started/stopped.

- Handle an interrupt for csa primarily in the bridge driver,
then invoke the children. As the value of HISR can be read
only once, pass the HISR to the children via struct
csa_bridgeinfo, stored in the ivars of them.
- Clear the contents of serial FIFO upon stopping the DMA for
playing. This may eliminate buzz on playing. Experimental.

- latest 2ndbuffer patch
- make chn_setdir work for rec on isa cards
- note: es1371 does not irq in smp

Submitted by: tanimura

Make the origins clear.

Pointed out by: Tatoku Ogaito <tacha@trap.fukui-med.ac.jp>

Update for pnp adjustments regarding NPNP.

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)

- Introduce the bridge drivers for Sound Blaser, GUS and Crystal
Semiconductor CS461x/428x.
- Add support for GUS and CS461x/428x pcm.

Bridges reviewed by: dfr, cg
GUS non-PnP support submitted by: Ville-Pertti Keinonen <will@iki.fi>
GUS PnP support tested by: Michiru Saito <mich@mtci.ne.jp>