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: 1.10.8;
Revert previous local conversion for vs(4).

vs(4) became to able to play audio again.
At the moment the encoding conversion using set_params() does
not seem to work for me. So vs(4) uses local conversion to/from
ADPCM instead of it. But this should be a temporary work.
XXX The playback quality is very poor compared to before...
XXX Recording is not tested.

branches: 1.8.30;
Switch to 2-clause license.

merge ktrace-lwp.

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: 1.5.4;
Remove an advertising clause.

branches: 1.4.10;
Rewrite encode/decode algorithm completely.
It improves playing/recording quality greatly
and it was almost done by Yosuke Sugahara <penta@fuchu.or.jp>.
Thanks a lot!

Add support of slinear8, slinear16_le, slinear16_be.

initialize codec variables every open().

Fix mis-increment bug in converter msm6258_ulinear8_to_adpcm().
This fix needs factor_denom.

branches: 1.1.2; 1.1.4;
Software codec for Oki MSM6258 voice synthesizer.
Recording untested.

Revert previous local conversion for vs(4).

vs(4) became to able to play audio again.
At the moment the encoding conversion using set_params() does
not seem to work for me. So vs(4) uses local conversion to/from
ADPCM instead of it. But this should be a temporary work.
XXX The playback quality is very poor compared to before...
XXX Recording is not tested.

branches: 1.8.30;
Switch to 2-clause license.

merge ktrace-lwp.

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: 1.5.4;
Remove an advertising clause.

branches: 1.4.10;
Rewrite encode/decode algorithm completely.
It improves playing/recording quality greatly
and it was almost done by Yosuke Sugahara <penta@fuchu.or.jp>.
Thanks a lot!

Add support of slinear8, slinear16_le, slinear16_be.

initialize codec variables every open().

Fix mis-increment bug in converter msm6258_ulinear8_to_adpcm().
This fix needs factor_denom.

branches: 1.1.2; 1.1.4;
Software codec for Oki MSM6258 voice synthesizer.
Recording untested.

vs(4) became to able to play audio again.
At the moment the encoding conversion using set_params() does
not seem to work for me. So vs(4) uses local conversion to/from
ADPCM instead of it. But this should be a temporary work.
XXX The playback quality is very poor compared to before...
XXX Recording is not tested.

Switch to 2-clause license.

merge ktrace-lwp.

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: 1.5.4;
Remove an advertising clause.

branches: 1.4.10;
Rewrite encode/decode algorithm completely.
It improves playing/recording quality greatly
and it was almost done by Yosuke Sugahara <penta@fuchu.or.jp>.
Thanks a lot!

Add support of slinear8, slinear16_le, slinear16_be.

initialize codec variables every open().

Fix mis-increment bug in converter msm6258_ulinear8_to_adpcm().
This fix needs factor_denom.

branches: 1.1.2; 1.1.4;
Software codec for Oki MSM6258 voice synthesizer.
Recording untested.