CVS log for src/sys/arch/amiga/dev/aucc.c

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Default branch: MAIN

Mostly merge changes from HEAD upto 20200411

Merge changes from current as of 20200406

Pull up following revision(s) (requested by isaki in ticket #792):

	sys/arch/amiga/dev/aucc.c: revision 1.48
	sys/dev/pci/auixp.c: revision 1.49
	sys/dev/pci/gcscaudio.c: revision 1.19
	sys/dev/pci/auich.c: revision 1.159
	sys/dev/sbus/dbri.c: revision 1.42
	sys/dev/pci/auvia.c: revision 1.85
	sys/dev/pci/auacer.c: revision 1.39

Drop 512 bytes limit on auvia_round_blocksize().

This fixes attach on 6 channels device.
PR kern/55017.

round_blocksize must return a multiple of the framesize.
aucc(4) supports 3 channels mode.

round_blocksize must return a multiple of the framesize
even if passed blocksize is greater than the upper limit.

round_blocksize must return a multiple of the framesize.

It's not divisible when blk=GCSCAUDI_PRD_SIZE_MAX and channels=4.
round_blocksize must return a multiple of the framesize
even if 6 channels mode.

I believe that keeping "good alignment" is just a wish, not constraint.

Sync with head.

round_blocksize must return a multiple of the framesize.
aucc(4) supports 3 channels mode.

Pull up following revision(s) (requested by rin in ticket #189):
	sys/arch/amiga/dev/aucc.c: revision 1.47
The device does not support recording.
Fix diagnostic assertion failure in aucc_set_format().
OK isaki

The device does not support recording.
Fix diagnostic assertion failure in aucc_set_format().

OK isaki

XXX
pullup to netbsd-9

Sync with HEAD

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

Move dev/audio_if.h -> dev/audio/audio_if.h

Remove simple flags that indicate whether the device is opened.
These are handled in the upper layer now.

Adapt to audio2.
- Remove its own mulaw converter.
- XXX I'm not sure about 3,4ch mode vs 8/14bit mode.

Use C99 style struct initializer to audio_hw_if.

Rebase to HEAD as of a few days ago.

sync with head.

for a reference, the tree before this commit was tagged
as yamt-pagecache-tag8.

this commit was splitted into small chunks to avoid
a limitation of cvs.  ("Protocol error: too many arguments")

sync with head

kill sprintf
fix unused variables

Resync to 2012-11-19 00:00:00 UTC

sync with head

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

sync with head

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.

get rid of unused 'powerstate' callback in audio_hw_if

port these to audiomp:  support get_locks, and take the intr_lock
in the interrupt handler.

sync with head

sync with head.

Sync with head.

sync with head - hopefully this will work

sync with head.

Use the SI capitalization for "Hz", "kHz", and "MHz" in comments and strings.
Add a space between numbers and Hz unit.

merge ktrace-lwp.

Sync with HEAD. Here we go again...

No 0x in front of %p...

Sync with HEAD.

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

audio_hw_if change
 * make open() and close() optional
 * add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters
 to round_blocksize()

* adopt the filter pipeline framework.
* repulse.c follows ac97_attach() change

Sync with HEAD.

constify audio_hw_if, midi_hw_if, and radio_hw_if.

DMA, not dma nor Dma.

Respect the law: Use A-law and mu-law as spellings as far as easily possible.
Inspired by Igor Sobrado in PR 19680.

Catch up to -current.

sync kqueue with -current; this includes merge of gehenna-devsw branch,
merge of i386 MP branch, and part of autoconf rototil work

Use CFATTACH_DECL().

Declare all cfattach structures const.

Catch up to -current.

Sync w/ -current.

add __KERNEL_RCSID as suggested by Luke Mewburn

- ANSIfy
- remove some trailing spaces/tabs
- minor style nits

Sync kqueue branch with -current.

Catch up with -current. Fix some bogons in the sparc64 kbd/ms
attach code. cd18xx conversion provided by mrg.

file aucc.c was added on branch nathanw_sa on 2002-02-28 04:06:32 +0000

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).

Catch up to -current.

defopt LEV6_DEFER.

Update thorpej_scsipi to -current as of a month ago
A i386 GENERIC kernel compiles without the siop, ahc and bha drivers
(will be updated later). i386 IDE/ATAPI and ncr work, as well as
sparc/esp_sbus. alpha should work as well (untested yet).
siop, ahc and bha will be updated once I've updated the branch to current
-current, as well as machine-dependant code.

I updated the amiga SCSI drivers, but completely untested yet (may not even
compile)

Pull up revision 1.26 (requested by is):
  Use the correct DMA pointer for the second channel on 14-bit
  byte-swapped conversion.  Fixes NULL pointer reference crash
  when playing 16-bit little-endian audio.

Pullup revision 1.26 (requested by mhitch, approved by jhawk)

  Log Message:
  Use the correct DMA pointer for the second channel on 14-bit byte-swapped
  conversion.  Fixes NULL pointer reference crash when playing 16 bit little
  endian audio.

Use the correct DMA pointer for the second channel on 14-bit byte-swapped
conversion.  Fixes NULL pointer reference crash when playing 16 bit little
endian audio.

Kill more broken cf_unit bogons.

Modifications to aucc.c for more input formats and 14bit stereo output.
By Bernardo Innocenti, submitted as PR 6787.
Some KNF-ifying done by myself, so you might want to blame me instead
if the original patch worked for you.
	-is

* 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.

Adjust for changes to config.

pullup from trunk (requested by ig)

Workaround for the "xanim volume setting problem":
We play mono samples on all 4 channels.
However, we get the volume settings for mono samples as a symmetic two-channel
setting... the other two channels used to stay at max volume...
I believe that something else is wrong here, but dont want to change MI
code (which in turn influences a couple of MD driver) thus late in the release
cycle.

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.

Update marc-pcmcia branch from trunk.

AudioC[A-Z] -> AudioC[a-z]
AudioElinear -> AudioEslinear

Update marc-pcmcia branch from trunk.

- Change audio_hw_if a little: set_param now sets the play and record modes
  at the same time instead by using two different calls.  This enables
  it to check more easily if the combined mode is all right.
- Improve the error checking in audio.c.
- Add a new audio property, AUDIO_PROP_INDEPENDENT, show if the
  play and record settings are independent.
- Fix some buglets in audio.c.

Update marc-pcmcia branch from trunk.

Change the MI audio driver so it attaches to the MD driver in the
normal way.  This requires adding a line to the config files to
get audio to work again.

Audio changes:
	- Change the way attach and open works to allow multiple audio
	  devices.
	- Split the mulaw.c file into two to avoid dragging in mulaw
	  convertsion when they are not needed.  Add 16 bit alaw/mulaw tables.
	- Change the way audio properties are gotten.
	- Recognize more versions os SoundBlaster.

Audio: Remove the machine dependant code I put in audio.c by mistake.
This adds another method to audio_hw_if.  Also remove a field from
audio_hw_if that was not read-only.

Update forgotten audio driver for changes in audio_hw_if.

audio: Simplify handling of AUDIO_SETFD and committing of encoding mode.

Garbage collect some unused routines.

In the name of backwards compatibility AUDIO_ENCODING_LINEAR has been
renamed AUDIO_ENCODING_SLINEAR and AUDIO_ENCODING_LINEAR reverts to the
NetBSD 1.2 sematics.  A kernel with COMPAT_12 defined will accept
AUDIO_ENCODING_LINEAR and treat it as before, without COMPAT_12 it
will be rejected.

- Fix gain control.
- play mono stuff on all channels.

Make multichannel output work. The cc parameter to hw_start_output is
a byte counter, it seems. Somebody should really write man 9 audio_if.
While here, make debug output invisible #ifndef AUCCDEBUG.

Multichannel output still doesn't work, but while working on it, I repaired
the double buffering code, and setting a couple of other parameters works now.

Make aucc work in the presence of LEV6_DEFER.

Change last commit to use `#error'.

Make this fail to compile with LEV6_DEFER.
XXX This isn't nice, but stopping the  system clock isn't nice, either.
We will repair this soon.

Add RCS ids, and clean up some unused in newstyle audio hardware drivers
functions.

Actually commit the sample rate to the hardware.
While we're here, add missing parts of (software encoded) ULINEAR encoding.

A bit more of KNF.

Amiga custom chip audio hardware driver, by Stefan Thesing
(thesing@cs.uni-sb.de), heavily hacked upon by me to
- make it work with -current audio system
- make it shut off Amiga audio DMA only at appropriate places.
XXX A couple of bugs still remain, which well be handled later.
XXX Among them: only mono output; doesn't refuse to handle input, but chokes;
will not play last millichunk (is this 20 ms?) of data.