Clean get_props().
- Make get_props() return AUDIO_PROP_{PLAYBACK,CAPTURE} properly.
This eliminates need for audio.c to take care of such (old)
drivers which don't return both of PLAYBACK and CAPTURE.
- All get_props() doesn't need to return AUDIO_PROP_MMAP.
It is handled in the audio layer now.

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: 1.25.2;
Use C99 style struct initializer to audio_hw_if.

Make it compilable (remove unused variables).

split device_t/softc for all remaining drivers.
replace "struct device *" with "device_t".
use device_xname(), device_unit(), etc.

branches: 1.22.8;
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: 1.21.34; 1.21.36;
Remove clause 3 and 4 from TNF licenses

branches: 1.20.2; 1.20.4;
use aprint_*_dev and device_xname

branches: 1.19.16;
machine/{bus,cpu,intr}.h -> sys/{bus,cpu,intr}.h

branches: 1.18.28; 1.18.30; 1.18.44; 1.18.46; 1.18.50;
merge ktrace-lwp.

branches: 1.17.10;
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: 1.15.2;
constify audio_hw_if, midi_hw_if, and radio_hw_if.

Remove junk code.

sprintf -> snprintf

branches: 1.12.2;
interrupt with two rs.

don't need <sys/types.h> when including <sys/param.h>

add/cleanup RCSID

Add a new optional method, dev_ioctl, to the audio hardware driver interface.
It is called when an unrecognized ioctl() is performed on a device,
thus allowing ioctl()s that frob the hardware driver (like loading
microcode).

branches: 1.8.20; 1.8.22; 1.8.24;
Fix Copyright dates

Assign my files to The NetBSD Foundation, Inc.

* Redo the way the way the MIDI driver attaches to audio devices.
* Improve the midisyn layer a little.
* Add a driver for the Yamaha OPL[23] FM synths.
The opl driver is not finished yet; it sounds pretty awful.

For some strange reason I cannot get any FM sound from my SB64 cards,
but a regular SB16 works fine.

Dont waste time in delay()s... else we won't finish in time to get our first
lowwatermark interupt for AUDIO_BYPASS data.

Fix pasto in symbolic name for string

tav_write_word -> tav_write_short

Make the audio API (almost) SunOS compatible.
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.

TMS320AV110 (mpeg layer 2 decoder chip) and ZBUS Melody board, which uses it.
No optionboards yet for Melody, only audio driver.
MPEG data types for audio.c

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: 1.25.2;
Use C99 style struct initializer to audio_hw_if.

Make it compilable (remove unused variables).

split device_t/softc for all remaining drivers.
replace "struct device *" with "device_t".
use device_xname(), device_unit(), etc.

branches: 1.22.8;
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: 1.21.34; 1.21.36;
Remove clause 3 and 4 from TNF licenses

branches: 1.20.2; 1.20.4;
use aprint_*_dev and device_xname

branches: 1.19.16;
machine/{bus,cpu,intr}.h -> sys/{bus,cpu,intr}.h

branches: 1.18.28; 1.18.30; 1.18.44; 1.18.46; 1.18.50;
merge ktrace-lwp.

branches: 1.17.10;
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: 1.15.2;
constify audio_hw_if, midi_hw_if, and radio_hw_if.

Remove junk code.

sprintf -> snprintf

branches: 1.12.2;
interrupt with two rs.

don't need <sys/types.h> when including <sys/param.h>

add/cleanup RCSID

Add a new optional method, dev_ioctl, to the audio hardware driver interface.
It is called when an unrecognized ioctl() is performed on a device,
thus allowing ioctl()s that frob the hardware driver (like loading
microcode).

branches: 1.8.20; 1.8.22; 1.8.24;
Fix Copyright dates

Assign my files to The NetBSD Foundation, Inc.

* Redo the way the way the MIDI driver attaches to audio devices.
* Improve the midisyn layer a little.
* Add a driver for the Yamaha OPL[23] FM synths.
The opl driver is not finished yet; it sounds pretty awful.

For some strange reason I cannot get any FM sound from my SB64 cards,
but a regular SB16 works fine.

Dont waste time in delay()s... else we won't finish in time to get our first
lowwatermark interupt for AUDIO_BYPASS data.

Fix pasto in symbolic name for string

tav_write_word -> tav_write_short

Make the audio API (almost) SunOS compatible.
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.

TMS320AV110 (mpeg layer 2 decoder chip) and ZBUS Melody board, which uses it.
No optionboards yet for Melody, only audio driver.
MPEG data types for audio.c

branches: 1.25.2;
Use C99 style struct initializer to audio_hw_if.

Make it compilable (remove unused variables).

split device_t/softc for all remaining drivers.
replace "struct device *" with "device_t".
use device_xname(), device_unit(), etc.

branches: 1.22.8;
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: 1.21.34; 1.21.36;
Remove clause 3 and 4 from TNF licenses

branches: 1.20.2; 1.20.4;
use aprint_*_dev and device_xname

branches: 1.19.16;
machine/{bus,cpu,intr}.h -> sys/{bus,cpu,intr}.h

branches: 1.18.28; 1.18.30; 1.18.44; 1.18.46; 1.18.50;
merge ktrace-lwp.

branches: 1.17.10;
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: 1.15.2;
constify audio_hw_if, midi_hw_if, and radio_hw_if.

Remove junk code.

sprintf -> snprintf

branches: 1.12.2;
interrupt with two rs.

don't need <sys/types.h> when including <sys/param.h>

add/cleanup RCSID

Add a new optional method, dev_ioctl, to the audio hardware driver interface.
It is called when an unrecognized ioctl() is performed on a device,
thus allowing ioctl()s that frob the hardware driver (like loading
microcode).

branches: 1.8.20; 1.8.22; 1.8.24;
Fix Copyright dates

Assign my files to The NetBSD Foundation, Inc.

* Redo the way the way the MIDI driver attaches to audio devices.
* Improve the midisyn layer a little.
* Add a driver for the Yamaha OPL[23] FM synths.
The opl driver is not finished yet; it sounds pretty awful.

For some strange reason I cannot get any FM sound from my SB64 cards,
but a regular SB16 works fine.

Dont waste time in delay()s... else we won't finish in time to get our first
lowwatermark interupt for AUDIO_BYPASS data.

Fix pasto in symbolic name for string

tav_write_word -> tav_write_short

Make the audio API (almost) SunOS compatible.
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.

TMS320AV110 (mpeg layer 2 decoder chip) and ZBUS Melody board, which uses it.
No optionboards yet for Melody, only audio driver.
MPEG data types for audio.c