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

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

- Increase dynamic range of filter coefficients from 28bit to 30bit.
This cause dramatic effect in overall precision and conversion quality
by pushing down most aliasing artifacts around -180 dB.

Spectrogram analysis/comparison:

http://people.freebsd.org/~ariff/z_comparison/z_28vs30/

- Guard against possible 64bit overflow during accumulation process by
slightly normalize and saturate sample and coefficient multiplication,
possible during extreme 32bit downsampling (eg. 380KHz -> 8KHz) with
custom preset that require more than ~7000 taps filter (which is
overkill).

- Add knobs through FEEDER_RATE_PRESETS to set dynamic range of filter
coefficients/accumulator and prefered polynomial interpolator:

COEFFICIENT_BIT:X
(where 1 <= X <= 30, default: 30)

ACCUMULATOR_BIT:X
(where 32 <= X <=64, default: 58)

INTERPOLATOR:I
(where I = ZOH, LINEAR, QUADRATIC, HERMITE, BSPLINE,
OPT32X, OPT16X, OPT8X, OPT4X, OPT2X)

Approved by: re (kib)

Slightly increase amount of bandwidth of resampling filter for
feeder_rate_quality=3. This have the benefit of reducing aliasing
artifacts due to alias masking.

Spectrogram analysis:

o Old preset (100:36:0.90)
http://people.freebsd.org/~ariff/z_comparison/z_q3_old.png

o New preset (100:36:0.92):
http://people.freebsd.org/~ariff/z_comparison/z_q3_new.png

Approved by: re (kib)

- Add a way to change filter oversampling factor through
FEEDER_RATE_PRESET "OVERSAMPLING_FACTOR:X .. .." where
X = log2(oversampling factor).

- Lower down default filter oversampling factor from 128
(log2 = 7) to 32 (log2 = 5), saving worth of 80 Kb.
The use of better polynomial interpolator will raise
its conversion quality/accuracy to match (or slightly
better) with previous settings.

- Bump driver version.

Move all sound related scripts to its own 'sound' subdir.

Suggested by: jmallett

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.