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From Marc Winiger <mw@msys.ch>
ok dlg, claudio.
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the mixer layer to the audio layer, differentiate between the
'monitor' class and the 'outputs' class instead of lumping them
together. not all mixers have a monitor class entry which was
causing the outputs class to be ignored.
- look for monitor port in the monitor class, but if the monitor
port is not found in the monitor class look in the outputs class
as well. the code was effectively looking in both the monitor and
outputs classes before, so this is no real change.
sets up audio_info.play.[gain|balance] correctly on at least emu(4),
auich(4), auvia(4) and snapper(4) ... probably most other AC'97 devices
as well. as a result, audioctl(1) works much better and the overall
volume can be set with /dev/audio instead of needing to go through
/dev/mixer.
from deanna@
thanks again to those who tested this change
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pointed out by mglocker@
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to allow buffer sizes and pointer to the buffer to be set to reasonable
values for the sample size of the data to be read/written.
fixes kernel corruption seen with azalia and probably other drivers
that are using something other than 8kHz monaural mulaw as the default
encoding when doing 'dd if=/dev/audio of=foo' as the first audio
operation after boot.
problem reported by deanna@
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one that can be used with other ciphers than WEP.
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and the userland sample size matters.
there may be different sample size factors for play and record,
so use the appropriate factor when userland sets the blocksize.
the reported blocksize, hiwat and lowat are only for the play buffer.
IMO, it would be good to report the record blocksize, hiwat and lowat
as well, but that is another issue.
tested with emu(4) and an auvia(4) modified to only do stereo in
hardware.
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may not be the same sample size as what userland is to receive.
account for the discrepency between the sample sizes when comparing
how much to copy out with what userland expects.
- only copy out the part of the buffer that is in the format userland
expects.
emu(4) can now record with all encodings it claims to be able to.
that is, dd if=/dev/audio of=foo does as audio(4) says (monaural
mu-law) and does it correctly.
AFAIKS, the only other driver that sets a read factor is uaudio(4),
and that depends on what the hardware supports.
tested on i386, amd64 and sgi with 3 different emu(4) cards
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2x on each loop, only loop .5x times
fixes hard hangs on i386 and "hiccups" on sgi when these functions are
used through e.g. dd if=/dev/audio of=foo, since mulaw is the default
encoding.
the commit history wrt this code is interesting. I am responsible for
introducing this braindamage to OpenBSD ... and note that
linear16_to_linear8_[lb]e had it right ...
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eeproms was backwards. This resulted in a bogus MAC address being
read from the eeprom on certain cards. From FreeBSD. OK pvalchev@
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The variety of different serial devices is just too big to make a decision up
front. This should make the second serial port on four-digit B/C/J-class
workstation work as a serial console too.
ok miod@
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- remove argument names from function prototypes
OK jasper@, ray@, "Didn't notice any errors" marc@
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the correct size for my 4.TB volume. Original diff from dlg@ with some
tweaks by me; ok dlg@
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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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while i'm here, fix allocation of the Rx ring descriptors so we don't
waste memory.
IMPORTANT: you must upgrade your wpi-firmware package to rev. 2.14.4
or the driver will stop working.
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data is expanded before it goes in the buffer. all sizes
and offsets used internally are of this expanded data.
there's no need to compensate for this fact except when
showing information to the user in case their actions
depend on how many bytes are in the buffer.
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try to go past the end of the buffer.
this and the last commit allow ffplay and gnash to work properly
on devices that may need to do mono to stereo conversion, like
auich(4).
problem reported by and much debugging deanna@
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by taking into account expansion we may have done in cases such as
playing monoaural media on a device that only plays stereo, like
auich(4).
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defined.
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ok krw@
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successful compilation of assembler or generating microcode.
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ALC882, and differs only by lacking an ADC and a mixer, so have it use
ALC882's get_port(), set_port() and a slightly modified version of its
mixer init.
This also works around a problem where the generic mixer init creates
truncated mixer item names which can't be manipulated (noticed by
steven@).
Tested by steven@, ckuethe@ and jmc@.
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reported a while back by Laurence Tratt, reminded more recently by brad@
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read-write and it supports full-duplex operation.
- update documentation to match this behaviour.
see http://marc.info/?l=openbsd-tech&m=118310788926431&w=2 for
more background on this change
"patches look correct to me" marc@
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buffer, not the record buffer.
inspired by recent similar change in NetBSD.
"patches look correct to me" marc@
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originally from Alexandre Ratchov
"patches look correct to me" marco@
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attribute to '0'. this function may be called when executing an
AUDIO_SETINFO ioctl where the pause attribute was set to '1', and
setting it to '0' in this function will cause unexpected bahaviour.
- according to audio(4), the AUDIO_FLUSH ioctl "restarts recording
and playback". therefor, it should set the pause attributes to '0'.
this was previously accomplished indirectly, because it calls
audio_init_ringbuffer(), which was setting pause to '0'.
- the pause attribute must be set to '0' in audio_open() so that
recording and playback buffers can be "activated" when the device
is opened. this was apparently forgotten when audiosetinfo()
was split off of audio_open() and the logic in audiosetinfo()
was changed to accomodate being run at times other than open().
also, this only really affects read() with poll(), which may
explain why the problem was not noticed for some time.
originally part of a patch Alexandre Ratchov sent to tech@ a
while back
fixes PR 3813
"patches look correct to me" marco@
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For instance, it may not set POLLIN when samples become available,
or may set POLLOUT when write(2) will block.
Fix this by making conditions used by poll (FILTREAD and FILTWRITE
macros) match the blocking conditions in audio_read() and audio_write().
originally from Alexandre Ratchov, with a small tweak by me.
"patches look correct to me" marco@
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the ESS bit must be set in (re)assocation requests.
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needed or not to the callers (instead of doing nothing in add_xrates).
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from brad, ok millert beck miod krw igor deraadt
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offload", and associated man page change. To use IPv4 transmit TCP/UDP
checksum offloading you must again define BNX_CSUM.
As requested by mbalmer@ via deraadt@ on suggestion of reyk@ in
response to PR #5437.
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can provide its own attach glue to ahci.
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that some vendors will ship their chip with two functions enabled, so now
the code will only configure the functionality that is enabled on the
current function. two function setups will now work as the oem intended.
requested by kettenis@
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