- Copy stable/10 (r259064) to releng/10.0 as part of the
10.0-RELEASE cycle.
- Update __FreeBSD_version [1]
- Set branch name to -RC1

[1] 10.0-CURRENT __FreeBSD_version value ended at '55', so
start releng/10.0 at '100' so the branch is started with
a value ending in zero.

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.

Drain all callout handlers during driver detach appropriately.

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

Despite several examples in the kernel, the third argument of
sysctl_handle_int is not sizeof the int type you want to export.
The type must always be an int or an unsigned int.

Remove the instances where a sizeof(variable) is passed to stop
people accidently cut and pasting these examples.

In a few places this was sysctl_handle_int was being used on 64 bit
types, which would truncate the value to be exported. In these
cases use sysctl_handle_quad to export them and change the format
to Q so that sysctl(1) can still print them.

Don't rely on pcm_channel index numbering while doing sequential channel
allocation.

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

Memory leak killing spree, mostly bus_dma(9) related.

[stage: 8/9]

Implement CHANNEL_SETFRAGMENTS() for snd_atiixp, snd_es137x, snd_hda
and snd_via8233. CHANNEL_SETBLOCKSIZE() will basically call
CHANNEL_SETFRAGMENTS() internally using conservative blocksize /
blockcount hints. Other drivers will be converted later.

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

Reduce maximum DMA segments from 128 to 64. We don't need more than that.

Total DMA segments should include total number of record channel(s).

Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.

General
-------

- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.

- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.

CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/

- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)

- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.

Driver specific
---------------

- Ditto for sysctls.

- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.

- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>

Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.

Joel Dahl will do the manpage update.

- Rename hw.snd.unit to hw.snd.default_unit to make the purpose more obvious.
- Enable 4 automatic vchan's by default.
- Add some comments which provide ides/questions for improvement.
- Prefix some temporary sysctl's with an underscore to denote that it is not
an official API but a workaround until the real solution is implemented.

Add support for (latest) VIA VT8251 (rev. 0x07) audio controller.

A slight difference of this chip from its previous siblings is that
it need a gentle "wake up" on every (full) DMA buffer completion to
avoid stalled interrupt handler.

Thanks to George Hartzell for permission on doing remote debugging.

Prime MFC candidate for 6.1-RELEASE. Please reply to this commit if
there are any objections (so I won't bug re@), since the changes
are too small and only specific to VT8251.

PR: i386/95949
Tested by: [1] George Hartzel
myself (remotely)
MFC after: 3 days

[1] http://lists.freebsd.org/pipermail/freebsd-multimedia/2006-April/004003.html

Fix a long standing unhandled interrupt bug which can cause
erratic system slowdown (beaten to a pulp) and possible panic. This
issue has bugged me for as long as I could remember, until I
realized that it is possible for register base offset to hold zero
value which is definitely a "FALSE".

Approved by: netchild (mentor)

- Locking improvements.
- Don't keep the SPDIF state in the driver private struct since it
can be overriden by hand with pciconf(8), query it when needed instead.

Regarding the locking I let Ariff explain it himself:
---snip---
About the locking, that is what I'm intended to do since the beginning.
The reason I'm not putting that along since my first patchset was
because several people especially from amd46 camp reported that it cause
lots of LORs, which is weird considering that I've never encounter such
in a pretty much strict locking environment (i386). However, since our
previous discussion with Pyun YongHyeon about strict locking, I've
decided to bring it back for all the affected drivers, not just for
es137x. It turns out that the root of the problem was within dsp.c
during device open, which has been fixed since dsp.c revision 1.84.
---snip---

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

* Add locking / MPSAFE
* Add kernel hint option to disable DXS channels entirely. Report
from several skype users / Pav Lucistnik indicate that disabling
DXS may fix lots of pop / crackling noise. To disable DXS add
hint.pcm.<unit>.via_dxs_disabled="1" to /boot/device.hints.
Further investigation of the issues regarding DXS showed, that
the problem is in another (more generic) place, but until the
right fix is tested/reviewed this may help a little bit.

Added sysctl's to aid testing/debugging:
hint.pcm.<unit>.via_dxs_disabled=X - Disable / Enable DXS channels entirely
hint.pcm.<unit>.via_dxs_channels=X - Limit DXS channels up to X
hint.pcm.<unit>.via_sgd_channels=X - Limit SGD channels up to X
hint.pcm.<unit>.via_dxs_src=X - Enable / Disable DXS sample rate
converter.

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

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

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

Add PCI ID for via 8237.

Submitted by: Josh Elsasser <jre@vineyard.net>
Approved by: tanimura (mentor)
PR: kern/61730

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

Perform warm or cold reset AC97 per the spec depending on state of the
codec during initialization. This code mirrors the reset code used on
the VIA82c686 and fixes a codec initialization failure (SoundMAX
AD1885) reported by Matthias Schuendehuette.

- Don't call pci_enable_io() in drivers (unless needed for resume).
- Don't test memory/port status and emit an error message; the PCI
bus code will do this now.

For the VIA8233A use multi-sgd register set for primary playback
channel and disable DXS3. Several users have reported DXS3 as playing
at half speed on the 8233A revision of the chipset. This implicitly
means no SPDIF for VIA8233A users.

Distinguish between register sets that depend on ac97 caps (wr0, wr1,
msgd) and those that don't (dxs0-3) when reporting format caps.

o Add back support for DXS channels.

o Make DXS3 the primary playback channel. It may be the only
universally supported channel with the assorted revisions of this
chipset.

o Add sysctl and handler for enabling s/pdif output from DXS3.

Fix assignment of record sgd_addr and clean up via8233chan_init.

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

Reviewed by: orion

Disable DRA to fix problems with recording.

Fix VRA configuration.

Reported and fixed by: Grzybowski Rafal <grzybek@inferno.mikrus.pw.edu.pl>

Support for VIA VT8233 audio controller.