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The property bits of audio(9) are obsolete and ought to be removed
completely.
sys/dev/audio.c:audio_open() currently uses get_props() to bail out if
read *and* write was requested on a non-duplex driver.
Drivers that currently support playing but not recording need adjustment
before the API can be cleaned up.
Drivers that advertise themselves as full duplex, i.e. those that always
return AUDIO_PROP_FULLDUPLEX unconditionally in their get_props() currently
always succeed this check.
As this is the only property, losen audio_open()'s DIAGNOSTIC check and only
do the duplex check if the driver provides get_props().
This allows for simple removal of get_props() from full-duplex drivers
without adding any other code or without changing functionality.
This includes all audio drivers under sys/dev/pci/ (maestro(4) being the
only unfinished exception here).
Other drivers as well as the API change can then follow in smaller diffs.
This builds on amd64, arm64, i386, macppc and sparc64.
amd64 with azalia(4) still plays, records as well as plays and records
at the same time on my X230 as tested with
$ aucat -i play.wav [-o rec.wav]
alpha and hppa tests by miod
OK ratchov miod
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This audio(9) struct will lose a member, but drivers init their struct quite
inconsistently, most pre-C99 style.
Use C99 style everywhere, to get rid of all annoying differences and allow
for easy member removals/additions:
- don't change current order of members
- no explicit NULL members
- no comments or blank lines
- trailing comma in last member line
GENERIC.MP builds fine with this on arm64, amd64, i386 and sparc64.
macppc and alpha build-tested by miod
OK ratchov miod
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AUDIO_PROP_FULLDUPLEX is the only audio(9) in use, the other two died with
commit 1cf2860827c8ca659d8097d8da94a5ae5b888c53
Author: ratchov <ratchov@openbsd.org>
Date: Thu Jun 25 06:43:45 2015 +0000
Reimplement the audio driver in a simpler way, removing unused/unusable
functionality. Same API and ABI except for the removed bits and no
behaviour change for programs using libsndio. With help from armani@
and mpi@, thanks.
but remained defined and set in drivers.
and the following merely moved them when they were dead code already:
commit 9215aa3dfad387bca877a805534df6dcfe8722eb
Author: ratchov <ratchov@openbsd.org>
Date: Wed Aug 31 07:22:43 2016 +0000
Delete unused ioctls and associated macros. Move macros that are still
used internally by low-level drivers from sys/audioio.h to
dev/audio_if.h instead of deleting them.
None of this is used in base or ports; codesearch.debian.net only shows
AUDIO_PROP_{CAPTURE,PLAYBACK} in firefox-esr, mozjs and cubeb.
ratchov points out that audio_if.h and audioio.h are private interfaces
and the codesearch shows SunOS and NetBSD bits (#ifdef'd out on OpenBSD).
OK ratchov
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ok mpi@ ratchov@ "More const is good" deraadt@
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unused but intended to be used to correlate audio and wskbd devices.
ok ratchov@
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feedback and ok tb@ jmc@ ok ratchov@
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Prompted by warning from clang 10.
ok patrick@
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*activate().
ok ratchov@
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methods from all audio drivers and from the audio_if structure as they
are never called.
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with no benefit.
ok millert, "correct" deraadt
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the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi
ok armani@
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have any direct symbols used. Tested for indirect use by compiling
amd64/i386/sparc64 kernels.
ok tedu@ deraadt@
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ok tedu
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after discussions with beck deraadt kettenis.
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kernel resumes normal (non-cold, able to run processes, etc) operation.
Previously we were relying on specific DVACT_RESUME op's in drivers
creating callback/threads themselves, but that has become too common,
indicating the need for a built-in mechanism.
ok dlg kettenis, tested by a sufficient amount of people
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Pointed out by LLVM.
ok ratchov@
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with the IPL_MPSAFE flag. In certain circumstances, this prevents
interrupt handlers from waiting the global kernel_lock possibly (hold
by slower code) to be released.
ok kettenis
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structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).
ok and help from kettenis, a lot of work from armani
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that's ever used it, and it's long since been changed to use
DVACT_{QUIESCE,SUSPEND,RESUME} instead.
ok deraadt@, dlg@; miod@ also agreed with this idea when I brought it
up a few weeks ago
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1) Allocating with M_WAITOK, checking for NULL, and calling panic() is
pointless (malloc() will panic if it can't allocate) so remove the check
and the call.
2) Allocating with M_WAITOK, checking for NULL, and then gracefully
handling failure to allocate is pointless. Instead also pass M_CANFAIL
so malloc() doesn't panic so we can actually handle it gracefully.
1) was done using Coccinelle.
Input from oga.
ok miod.
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ok claudio@
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DVACT_{QUIESCE,RESUME}. not 100% right yet. after resume, interrupts,
DMA, and the mixer appear to be working, but no sound is produced.
cookie for whoever finds the problem.
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the AUOPEN_ macros from sys/dev/audiovar.h are for something else.
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for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.
discussed with ratchov, deraadt
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between instances, saving space in the kernel. feedback from many (some
incorporated, some left for future work).
ok deraadt, kettenis, "why not" miod.
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flags field and use the flags as appropriate.
ok jakemsg@
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(openlunatic at gmail).
tested by Dawe (dawedawe at gmx de), thanks, and me. does not disturb
currenlty working emu(4) devices.
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values of the audio_params structure during AUDIO_SETINFO if the
hardware cannot be set to exactly the requested mode.
some drivers do this sometimes. others always return EINVAL if there
isn't an exact match.
be more consistent. only return EINVAL if an absurd parameter was
requested, otherwise return a supported set of parameters, as close
as possible to what was requested.
with/ok ratchov@
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Not sure what's more surprising: how long it took for NetBSD to
catch up to the rest of the BSDs (including UCB), or the amount of
code that NetBSD has claimed for itself without attributing to the
actual authors.
OK deraadt@
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this makes the emulation expansion factor 1 by default since there's no
emulation. previously the factor was 2 because these drivers emulate
8-bit ulaw playback with 16-bit linear encodings. having a factor
other than 1 by default creates block size and hiwat/lowat issues
with applications that set the block size and hiwat before other
parameters.
ok ratchov@
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instead of 8-bit mono mulaw @ 8kHz.
this is just the infrastructure; no drivers are specifying a default
yet.
ok ratchov@, deanna@
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SB Live!/5.1/Audigy/2/4 cards that both OSS and ALSA provide
separate modules from their emu10k1 drivers to support them.
and in PR 5143, we see that emu(4) does not support these Dell
SBLive! cards, so don't attach emu(4) to them.
ok ratchov, brad
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be set separately on each channel. if changing the number of channels
make sure to set the sample rate on each channels.
fixes problems with src/regress/sys/dev/audio/autest.c, and
the hydrogen port/package, which set stereo mode either after or
without changing sample rates.
ok ratchov@
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it to 16-bit slinear_le.
this appears to be the result of a mistake in porting the driver
from NetBSD. when this driver was ported, NetBSD had some members
in it's 'audio_params' structure that OpenBSD doesn't have. in
particular, NetBSD had 'hw_encoding' and 'hw_precision', which store
what the hardware is presenting. the 'encoding' and 'precision' members
store what the audio layer is presenting (both in NetBSD and
OpenBSD) and can be different than what the hardware is presenting.
the original NetBSD code was setting 'hw_encoding' and 'hw_precision',
and in porting was changed to 'encoding' and 'precision'.
with this change the hardware is still presenting 16-bit slinear_le;
having the variable 'b16' set to '1' is enough to ensure this.
however now the audio layer will convert it to the requested format,
instead of always outputting 16-bit slinear_le.
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- figure sample size into start of voice buffer location
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- add big-endian support
- fix a few typos
From NetBSD
- use pci_matchbyid()
Known working SB Live's tested by numerous users with no reported regressions;
though no one has stepped up to test out the Audigy support yet.
ok deraadt@
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userland-visible sys/select.h. Consistent with what Net and Free do.
OK deraadt@, tested with full ports build by naddy@.
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pcisubmatch(); kettenis@ testing; brad@ ok
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From NetBSD via Antoine Jacoutot <ajacoutot at lphp.org>
Ok millert@ mickey@ deraadt@
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From: NetBSD
ok mickey@
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(mvme68k/mvme88k) vs.c and (vax) if_le.c ok miod@
isakmpd ones ok ho@
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ok mickey@
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ok mickey@
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