fix spelling mistakes, mainly in comments and log messages.

Merge isaki-audio2 branch, the overhaul of audio subsystem.
- Interrupt-oriented system rather than thread-oriented.
- Improve stability, quality and performance.
- Split playback and record cleanly. Improve halfduplex support.
- Many bugs are fixed including deadlocks, resource leaks, abuses, etc.
- Simplify audio filter mechanism. The encoding/channels/frequency
conversions are completely handled in the upper layer. So the hard-
ware driver only converts its hardware encoding (if necessary).
- audio_hw_if changes:
- Obsoletes query_encoding and add query_format instead.
- Obsoletes set_params and add set_format instead.
- Remove drain, setfd, mappage.
- The call sequences are changed.
- ioctl AUDIO_GETFD/SETFD, AUDIO_GETCHAN/SETCHAN are obsoleted.
- ioctl AUDIO_{QUERY,GET,SET}FORMAT are introduced.
- cleanup config attributes: au*conv and mulaw.
- All hardware drivers should follow it (I've done as much as possible).

Some file paths are changed:
- dev/audio.c -> dev/audio/audio.c (rewritten)
- dev/audiovar.h -> dev/audio/audiovar.h
- dev/audio_dai.h -> dev/audio/audio_dai.h
- dev/audio_if.h -> dev/audio/audio_if.h
- dev/audiobell.c -> dev/audio/audiobell.c
- dev/audiobellvar.h -> dev/audio/audiobellvar.h
- dev/mulaw.[ch] -> dev/audio/mulaw.[ch] + dev/audio/alaw.c

branches: 1.18.50; 1.18.54;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From
the original ad-audiomp branch notes:

Add MP locking to the audio drivers.

Making the audio drivers MP safe is necessary before efforts
can be made to make the VM system MP safe.

The are two locks per device instance, an ISR lock and
a character device lock. The ISR lock replaces calls to
splaudio()/splx(), and will be held across calls to device
methods which were called at splaudio() before (e.g.
trigger_output). The character device lock is held across
calls to nearly all of the methods, excluding some only
used for initialization, e.g. get_locks.

Welcome to 5.99.57.

branches: 1.17.12; 1.17.14;
device_t/softc split
patch presented on tech-kern@, no comments on it

branches: 1.16.4; 1.16.16; 1.16.18;
Detach self and children. Make some functions static. Use device_t
and accessors. Use PMF_FN_*.

branches: 1.15.6; 1.15.10;
Merge jmcneill-pm branch.

branches: 1.14.14; 1.14.16; 1.14.24; 1.14.26;
Kill caddr_t; there will be some MI fallout, but it will be fixed shortly.

branches: 1.13.26; 1.13.28;
merge ktrace-lwp.

branches: 1.12.10;
ansify and KNF

merge kent-audio1 branch, which introduces audio filter pipeline to the MI
audio framework


Summary of changes:

* struct audio_params
- remove sw_code, factor, factor_denom, hw_sample_rate,
hw_encoding ,hw_precision, and hw_channels. Conversion information
is conveyed by stream_filter_list_t.
- change the type of sample_rate: u_long -> u_int
- add `validbits,' which represents the valid data size in
precision bits. It is required in order to distinguish 24/32bit
from 24/24bit or 32/32bit.

* audio_hw_if
- add two parameters to set_params()
stream_filter_list_t *pfil, stream_filter_list *rfil
A HW driver should set filter recipes for requested formats
- constify audio_params parameters of trigger_output() and
trigger_input(). They represent audio formats for the hardware.
- make open() and close() optional
- add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters
to round_blocksize()

* sw_code is replaced with stream_filter_t.
stream_filer_t converts audio data in an input buffer and writes
into another output buffer unlike sw_code, which converts data in
single buffer.
converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c,
dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are
reimplemented as stream_filter_t

* MI audio
- audiosetinfo() builds filter pipelines from stream_filter_list_t
filled by audio_hw_if::set_params()
- audiosetinfo() returns with EINVAL if mmapped and set_params()
requests filters
- audio_write(), audio_pint(), and audio_rint() invoke a filter
pipeline.
- ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS,
AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for
AUDIO_GETINFO handle values for a buffer nearest to userland.

* add `struct device *' parameter to ac97_attach()

* all of audio HW drivers follow audio_hw_if and ac97 changes

branches: 1.10.2;
ac97_host_if::reset() returns non-zero value if codec reset fails, and
ac97 is not attached in that case.

PR: kern/26973

branches: 1.9.2;
netbsd.org -> NetBSD.org

This was the last commit of this kind to src/sys, which is now totally
"NetBSD.org clean". Thanks for the patiance, and sorry for all the commits.

branches: 1.8.2;
Add extensible malloc types, adapted from FreeBSD. This turns
malloc types into a structure, a pointer to which is passed around,
instead of an int constant. Allow the limit to be adjusted when the
malloc type is defined, or with a function call, as suggested by
Jonathan Stone.

Removed esm_freemem(), which isn't used any more.

First cut at recording support for the ESS Maestro-2 and 2e.
16-bit mono recording seems to work OK. 16-bit stereo recording
is missing the left channel for reasons unknown, fixes welcome.
8-bit recording still unsupported.

Avoid an "unmamed fields" warning (gcc 3.1).

branches: 1.4.2; 1.4.4; 1.4.6;
Power Management Hook
esm_powerhook();

No-op commit to force update to a non-"-kk" revision.

branches: 1.2.2;
Add a quirk table for specific implementations.
Change the NEC-specific subvendor test for GPIO initialization into a more
generic quirk table flag.
Add a quirk flag for hardware that has left and right channels reversed
(currently the COMPAL 38W2 a.k.a. Dell Inspiron 5000e)
Clean up PCM channel initialization and add channel swapping option.

Initial import of the Maestro 2/2e 'esm' PCI audio driver. This driver
is based on the FreeBSD 'maestro' driver by Taku YAMAMOTO
<taku@cent.saitama-u.ac.jp>