CVS log for src/sys/arch/x68k/dev/vsvar.h

Up to [cvs.NetBSD.org] / src / sys / arch / x68k / dev

Request diff between arbitrary revisions

Keyword substitution: kv
Default branch: MAIN

TAB/space/indent cleanup.

Sync with HEAD.

Remove an unused variable.

Remove sc_active flag.  sc_[pr]intr can be used instead of it.

Sync with HEAD

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

Adapt to audio2.

update from HEAD

Avoid magic numbers and give appropriate names.

Pull up following revision(s) (requested by isaki in ticket #172):
	sys/arch/x68k/dev/vs.c: 1.38-1.39, 1.41-1.44, 1.47-1.48 via patch
	sys/arch/x68k/dev/vsvar.h: 1.12-1.15 via patch
Avoid panic when the device is closed when not playing.
Catch up vs_set_params() to recent MI audio (after in-kernel mixer).
Before that, MD part had to support all encodings I'd like to support,
but currently it's no longer necessary.  The hardware is
4bit/1ch/15.6kHz ADPCM but it behaves as 16bit/1ch/16.0kHz PCM.
For audio.c < 1.362, the device attach succeeded and playback is
still working.
For audio.c >= 1.363, the device attach fails again.
It does not work yet but I commit it for milestone.
Update confused vs_set_params().
play and rec are identical but pfil and rfil are independent.
XXX I introduce VS_USE_PREC8 option for debugging purposes
    temporarily.  I'll remove it if the problem is solved.
Remove temporary VS_USE_PREC8 option.
vs(4) now supports slinear16be and slinear8 (but now work yet).
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.
Merge prate and rrate.  These can not be separated.
- Revert temporary local conversion introduced at rev 1.43.
- But does not revert to trigger method.  trigger method is not suitable for
  x68k ADPCM+DMA mechanism.
- Don't (re)start ADPCM when DMA is running.  This solves the noise.
  From Y.Sugahara.
- Cache dmac xfer.
- Finally MI audio supports 4bit precision format without null_filter hack!
- Fix reusing play/rec argument as local variables.  It is in/out parameter.

- Revert temporary local conversion introduced at rev 1.43.
- But does not revert to trigger method.  trigger method is not suitable for
  x68k ADPCM+DMA mechanism.
- Don't (re)start ADPCM when DMA is running.  This solves the noise.
  From Y.Sugahara.
- Cache dmac xfer.

Sync with HEAD

Merge prate and rrate.  These can not be separated.

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.

Avoid panic when the device is closed when not playing.

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.

adapt to audiomp api changes

Switch to 2-clause license.

sync with head.

Sync with wrstuden-revivesa-base-2.

Sync with HEAD.

Sync with head.

Split device_t/softc.  Use aprint*() and device_xname().

sync with head.

Sync with HEAD.

Kill caddr_t; there will be some MI fallout, but it will be fixed shortly.

Add MP locking for audio drivers.

merge ktrace-lwp.

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

adopt the filter pipeline framework

Sync with HEAD.

Follow up on the constification of audio_hw_if, to fix build
problem for vs.c.

Fix the sync with head I botched.

Sync with HEAD.

Sync with HEAD

Remove an advertising clause.

Catch up to -current.

Correct playing mu-law and linear.

file vsvar.h was added on branch nathanw_sa on 2001-05-03 02:09:13 +0000

Correct playing mu-law and linear.

X68k built-in voice synthesizer.