Revert r330897:

This was intended to be a non-functional change. It wasn't. The commit
message was thus wrong. In addition it broke arm, and merged crypto
related code.

Revert with prejudice.

This revert skips files touched in r316370 since that commit was since
MFCed. This revert also skips files that require $FreeBSD$ property
changes.

Thank you to those who helped me get out of this mess including but not
limited to gonzo, kevans, rgrimes.

Requested by: gjb (re)

Partial merge of the SPDX changes

These changes are incomplete but are making it difficult
to determine what other changes can/should be merged.

No objections from: pfg

Copy head@r302406 to stable/11 as part of the 11.0-RELEASE cycle.
Prune svn:mergeinfo from the new branch, as nothing has been merged
here.

Additional commits post-branch will follow.

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

Replace 0 with NULL for pointers in misc. device drivers.

Found with devel/coccinelle.

Use uintmax_t (typedef'd to rman_res_t type) for rman ranges.

On some architectures, u_long isn't large enough for resource definitions.
Particularly, powerpc and arm allow 36-bit (or larger) physical addresses, but
type `long' is only 32-bit. This extends rman's resources to uintmax_t. With
this change, any resource can feasibly be placed anywhere in physical memory
(within the constraints of the driver).

Why uintmax_t and not something machine dependent, or uint64_t? Though it's
possible for uintmax_t to grow, it's highly unlikely it will become 128-bit on
32-bit architectures. 64-bit architectures should have plenty of RAM to absorb
the increase on resource sizes if and when this occurs, and the number of
resources on memory-constrained systems should be sufficiently small as to not
pose a drastic overhead. That being said, uintmax_t was chosen for source
clarity. If it's specified as uint64_t, all printf()-like calls would either
need casts to uintmax_t, or be littered with PRI*64 macros. Casts to uintmax_t
aren't horrible, but it would also bake into the API for
resource_list_print_type() either a hidden assumption that entries get cast to
uintmax_t for printing, or these calls would need the PRI*64 macros. Since
source code is meant to be read more often than written, I chose the clearest
path of simply using uintmax_t.

Tested on a PowerPC p5020-based board, which places all device resources in
0xfxxxxxxxx, and has 8GB RAM.
Regression tested on qemu-system-i386
Regression tested on qemu-system-mips (malta profile)

Tested PAE and devinfo on virtualbox (live CD)

Special thanks to bz for his testing on ARM.

Reviewed By: bz, jhb (previous)
Relnotes: Yes
Sponsored by: Alex Perez/Inertial Computing
Differential Revision: https://reviews.freebsd.org/D4544

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.

Spelling fix for interupt -> interrupt

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.

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.

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

Clean up the BSD license to match the preferred license in
/usr/share/examples/etc/bsd-style-copyright. I've fixed a
few minor wording and formatting differences.

Approved by: luigi, Hannu Savolainen <hannu@opensound.com>

Recover (?) support for AD1815 based ISA soundcards.

PR: kern/94388
Submitted by: Krzysztof Kotlenga <piernik at gmail dot com>
MFC after: 3 days

- Fixup the locking.
- Don't mark MPSAFE (yet).

Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>

Use BUS_PROBE_DEFAULT in preference to 0 and BUS_PROBE_LOW_PRIORITY in
preference to some random negative number to allow other drivers a
bite at the apple.

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

Add acpi attachments for ISA sound drivers. This is needed so they'll
probe and attach when ACPI is enabled.

Submitted by: takawata (sbc fix)
MFC after: 1 day

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)

update my email address.

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

- Clean up ISA DMA supports.
- Rename all sndbuf_isadma* functions to sndbuf_dma* and move them into
sys/dev/sound/isa/sndbuf_dma.c.

No response from: sound

Do not return(foo()) in void function.

Submitted by: marius@alchemy.franken.de
MFC after: 3 days

(hopefully) fix build breakage some people are seeing

Approved by: re

Fix driver to re-enable sound output on AD1816 based cards caused by an
obviously bogous return value of ad1816chan_setformat().
PR: 37932
Submitted by: Martin Kaeske <Martin.Kaeske@Stud.TU-Ilmenau.DE>
Reviewed by: hm
MFC after: 10 days

Add lock type arguments to callers of snd_mtxcreate().

allow the hardware buffer size to be controlled with hints
release isa dma channels on unload (ad1816, ess, sb8)

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.

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.

lock the mutex, not the softc pointer.

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.

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

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

make drivers start at beginning of buffer when triggered - improves mmap.

not all tested.

not sure about aureal.c or csapcm.c

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

Submitted by: tanimura

fix a bug in sndstat output, tidy up a few bits

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)

fix mixer to return properly scaled values

repo-copied to make way for newmidi, this commit updates include paths

Fix a typo in the DRIVER_MODULE() declaration which caused a collision
with the real mss module.

seperate the ad1816 driver from the mss driver since it shares no code