- Playback and headphone/speaker automute works.
- Recording untested due to me being deaf doing back-and-forth
remote debugging.
Free Macbook donation is highly appreciated :)
Tested by: Dennis Pielken <mips128@gmx.net>
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
- Add support for the Conexant Waikiki/CX20551-22, found
in most Toshiba P100 series laptops. Despite of growing
urban legend of "unsupported Conexant", this codec is fully
supported in this driver.
Note: Toshiba P100 has broken (acpi) BIOS, thus rendering
its soundchip useless. Please disable ACPI, or get
BIOS updates (if any).
Found/tested by: Vulpes Velox <v.velox@vvelox.net>
URL: http://lists.freebsd.org/pipermail/freebsd-multimedia/2006-September/004896.html
- Parser cleanups to handle possible oss/mixer collision. Found
after parsing Conexant Waikiki nodes.
- Increase resilient against resource failure during attach/detach.
- Implement simple config through hint.pcm.<unit>.config. Supported
options:
gpio0 (default on Acer), gpio1, gpio2, softpcmvol,
fixedrate (default), forcestereo (default)
* Option prefixed with "no" (such as "nofixedrate") will do
the opposite.
* Options can be separated using space " " or comma ",".
* The "no" option will take precedence over anything else.
Example:
hint.pcm.0.config="gpio2,nofixedrate,noforcestereo,nogpio0,softpcmvol"
hint.pcm.0.config="softpcmvol noforcestereo"
- Fix support for ASUS M5200ae (buggy BIOS)
- Fix few problems, reported by Coverity Prevent (TM).
CID: 246991, 246676, 246675, 246674, 246477
Found by: Coverity Prevent (TM)
This driver make a special guarantee that "playback" works
on majority hardwares with minimal or without specific vendor
quirk.
This driver is a product of collaborative effort made by:
Stephane E. Potvin <sepotvin@videotron.ca>
Andrea Bittau <a.bittau@cs.ucl.ac.uk>
Wesley Morgan <morganw@chemikals.org>
Daniel Eischen <deischen@FreeBSD.org>
Maxime Guillaud <bsd-ports@mguillaud.net>
Ariff Abdullah <ariff@FreeBSD.org>
....and various people from freebsd-multimedia@FreeBSD.org
Refer to snd_hda(4) for features and issues.
Welcome To HDA.
Sponsored by: Defenxis Sdn. Bhd.
- fix multiple initialization of the first codec (support for more than
one codec should be added in the future)
- use spicds instead of ak452x module
Submitted by: "Konstantin Dimitrov" <kosio.dimitrov@gmail.com>
Reported by: Nick Withers < nick AT nickwithers DOT com >
Tested by: Nick Withers < nick AT nickwithers DOT com >
No objection from: ariff
MFC after: 1 week
is interaction between in-kernel sound buffer handling and hardware.
With small buffer, there are times when both harwdare reads and
kernel writes to the same buffer (it is only visible on slow machines, i
think). I'm digging in channel.c and buffer.c to find a solution that
allow use of large hardware buffers without sound lags - hardware can
handle buffers up to 32Mb."
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
- fix "No sound in KDE":
The problem is related to the implementation of Envy24(1712) hardware
mixer support in the driver. Envy24(1712) has very precise 36bit wide
hardware mixer, which is superior that vchans (software sound mixer in
the kernel). The driver supports Envy24(1712) hardware mixer, so up to
10 channels (5 stereo pairs) can be playback simultaneously.
However, there are problems with the implementation of Envy24(1712)
hardware mixer support in the driver, one of them is the problem with
"no sound in KDE":
When playing back several channels simultaneously and
stoping one of the channels, sound starts to stutter and
plays at very low speed.
Another problem is:
Playing back simultaneously more than one 24bit/32bit
sound file or 16bit sound file and 24bit/32bit sound
file doesn't work as expected.
Submitted by: "Konstantin Dimitrov" <kosio.dimitrov@gmail.com>
from a semantic point of view, but I notified the author of the driver
for confirmation. So far it at least fixes the build and should only
lead to not identifying or wrongly identifying a soundcard in the worst
case.
sound cards with optional pseudo-multichannel playback.
It's based on snd_emu10k1 sound driver. Single channel version is available
from audio/emu10kx port since some time.
The two new ALSA header files (GPLed), which contain Audigy 2 ("p16v") and
Audigy 2 Value ("p17v") specific interfaces, are latest versions from ALSA
Mercurial repository.
This is not connected to the build yet.
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
latest version from Mercurial repository. It brings definition of some
additional Audigy 2 / Audigy 2 Value registers.
- Use new #defines from ALSA emu10k1.h
- Remove unused include files:
+ emu10k1-ac97.h was imported from ALSA and never used,
+ emu10k1.h was imported from Creative Linux emu10k1 driver, but only
AUDIGY_CODEBASE was used from it.
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
- Enable 4 automatic vchan's by default.
- Add some comments which provide ides/questions for improvement.
- Prefix some temporary sysctl's with an underscore to denote that it is not
an official API but a workaround until the real solution is implemented.
yet. More commits to follow.
I got no response from the author, but since the driver is BSD licensed
I don't think he will complain. :-)
I got it from http://people.freebsd.org/~lofi/envy24.tar.gz
Written by: Katsurajima Naoto <raven@katsurajima.seya.yokohama.jp>
but large parts are rewritten by matk and tanimura.
This is old code, it's not maintained since 2003. We also don't have a
maintainer for this! Yuriy Tsibizov took it and uses it in his emu10kx
driver. Since the emu10kx driver will enter the tree "soon" (some bugs
have to be fixed after Yuriy return from his holidays), I add it here
already.
This also contains some changes to emu10k1 and cmi, so if you're lucky,
you can now make some kind of use of midi with those soundcards.
To all those poor souls which don't have such a card: feel free to send
patches, we don't have a maintainer for this.
To those which miss a specific feature in the midi code: feel free to
submit patches, we don't have a maintainer for this.
Oh, did I already told that it would be nice if someone would take care
of it? Maintainer with midi equipment wanted! :-)
If you get LOR's, submit a PR and notify multimedia@ please. If you get
panics, submit a PR with a backtrace (compile the sound system into your
kernel instead of using modules in this case) and notify multimedia@
please.
Written by: matk, tanimura
Submitted by: "Yuriy Tsibizov" <Yuriy.Tsibizov@gfk.ru>
Based upon: code from NetBSD
A slight difference of this chip from its previous siblings is that
it need a gentle "wake up" on every (full) DMA buffer completion to
avoid stalled interrupt handler.
Thanks to George Hartzell for permission on doing remote debugging.
Prime MFC candidate for 6.1-RELEASE. Please reply to this commit if
there are any objections (so I won't bug re@), since the changes
are too small and only specific to VT8251.
PR: i386/95949
Tested by: [1] George Hartzel
myself (remotely)
MFC after: 3 days
[1] http://lists.freebsd.org/pipermail/freebsd-multimedia/2006-April/004003.html
- [1] Make the driver friendly towards kernel without PREEMPTION.
Use msleep(9) instead of simple unlock-check_variable-lock mechanisme
since the later not really effective in non-preemptible kernel
(especially during codec detection routine).
- Free most driver resources in a sane manner to avoid possible
double free and panics especially during device detach and codec
detection failure.
MFC after: 3 days
[1] http://lists.freebsd.org/pipermail/freebsd-questions/2006-March/116515.html
forcing DMA alignment to default buffer size.
- Make sure DMA pointer properly aligned to avoid being truncated by caller
which causing severe underruns and random popping (especially in 32bit
playback / recording).
- Add AC97 inverted external amplifier quirk for Maxselect x710s
- http://maxselect.ru/
MFC after: 1 week
needed here, except there's a bug which results in detaching the device
twice.
Move the NULL pointer check to the beginning of the function and convert
it into a KASSERT.
CID: 420
Found with: Coverity Prevent(tm)
Discussed with: ariff
MFC after: 5 days
The minimum / maximum speed was way too low / high!
minspeed = 2000 - is this for real ?
maxspeed = 767999 - is this for real ?????
Wrap everything into 8000 - 48000 boundary, just to be safe.
MFC after: 3 days
- Mark MPSAFE since most of the locking procedures already implemented.
- Turn on inverted external amplifier sense flag for selected boards.
Tested by: bland
MFC after: 1 week
Instead of dragging the entire ICH4/82801DB into this mess, select
only few boards based on pci subdevice / subvendor.
Tested by: Daisuke Orikasa <luxury-acura-3.5rl at nifty.com>
MFC after: 3 days
- MPSAFE
- Fix / reorganize attach routine. Device specific initialization must
be done after generic bus / DMA setup. At last, Virtual Channels
(vchan) works as expected.
Note: Recent commit / fix against this driver proves that major enhancements
on the generic sound layer does indeed help to expose flaw within
device specific code. There are probably other drivers that need to
be addressed as well.
Tested by: barner
MFC after: 1 week
that enabling busmastering would result in PCR bit ON after codec
reset.
While I'm here add DELAY(1) to codec access routine to give reasonable
time to codec operation. Without the delay, it would cause problems on
super-fast machines(> 2GHz). Also enable legacy audio for all 6300ESB,
82801[D-G]B chips. Previously, it enabled legacy audio for 82801DB(ICH4)
chip only.
Reported by: Maxim Maximov mcsi AT mcsi DOT pp DOT ru
Andrew Bliznak andriko.b AT gmail DOT com
Tested by: brueffer, Maxim Maximov, Andrew Bliznak
erratic system slowdown (beaten to a pulp) and possible panic. This
issue has bugged me for as long as I could remember, until I
realized that it is possible for register base offset to hold zero
value which is definitely a "FALSE".
Approved by: netchild (mentor)
compatible AC97 codec.
- As the driver supports so many variants, create a table ids for
ease of probing and maintenance.
Submitted by: yongari
Reviewed/Tested by: multimedia@
- From luigi:
The code to compute fragment sizes in the ich driver almost
invariably ends up using the full buffer available, no matter
how the user specifies fragment size and number.
With audio telephony (8khz, 16bit-stereo) and the 16k buffer
size this results in an unbearable 500ms delay.
This patch makes sure that we never use more than 4 fragments,
(i don't think we need more unless there are huge interrupt
servicing latencies), and obey to the requested fragment size,
so that latency is acceptable.
Based on this (and after much regression tests), I can conclude
that this driver works best with 2 fragments, thus solving various
long standing issues of ICH driver not capable to flush or play
short files perfectly.
Suggested by: luigi (the idea of smaller fragments)
- MPSAFE conversion.
Approved by: netchild (mentor)
distinct hardware playback channels. DAC configuration can be
accessed through kernel hint - hint.pcm.<unit>.dac="val" with
following possible values:
0 = Enable both DACs (default)
1 = Enable single DAC (DAC1)
2 = Enable single DAC (DAC2)
3 = Enable both DACs, swap position (DAC2 comes first instead
of DAC1)
Special case for ES1370:
Unlike ES1371,2,3/CT5880, volume for each DAC 1 and 2 can be
controlled indepedently (synth for DAC1, pcm for DAC2). It is
possible that user will confuse by this behaviour, since both
DACs are enabled by default. Thus, provide a knob through sysctl
hw.snd.pcm<unit>.single_pcm_mixer:
0 = each DACs will be controlled separately (synth/pcm).
1 = combine both DACs volume mixer controller into a single
"pcm" (default)
As a side note, fixed rate operation (provided by previous
commit) is not a mandatory if the configuration space does not
involve DAC2 (perhaps disabled by user through the above kernel
hint). Unlike DAC2, DAC1 has its own register / control space,
not affected by the speed settings of ADC.
Tested by: multimedia@
Approved by: netchild (mentor)
- Don't keep the SPDIF state in the driver private struct since it
can be overriden by hand with pciconf(8), query it when needed instead.
Regarding the locking I let Ariff explain it himself:
---snip---
About the locking, that is what I'm intended to do since the beginning.
The reason I'm not putting that along since my first patchset was
because several people especially from amd46 camp reported that it cause
lots of LORs, which is weird considering that I've never encounter such
in a pretty much strict locking environment (i386). However, since our
previous discussion with Pyun YongHyeon about strict locking, I've
decided to bring it back for all the affected drivers, not just for
es137x. It turns out that the root of the problem was within dsp.c
during device open, which has been fixed since dsp.c revision 1.84.
---snip---
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
code which may help.
People with a ich compatible soundcard which want to help out should
change the "#if 1" to a "#if 0" and try if the soundcard still works.
Reports about working or not-working soundcards with this change to
multimedia@ please.
PR: 73987
sampling rate between playback and recording. This can be
disabled / enabled via kernel hints
(hint.pcm.<unit>.fixed_rate=0/4000-48000) or sysctl
hw.snd.pcm<unit>.fixed_rate=0/4000-48000). Default to 48khz
fixed rate. [1]
* Basic cleanup. *_es1371x_* -> *_es137x_*.
* Some locking fixes. [2]
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Discussed with: yongari [2]
See also: http://lists.freebsd.org/pipermail/freebsd-multimedia/2005-September/002758.html [1]
Reported by: Jos Backus <jos at catnook.com> [1]
In case this causes trouble for some other chipsets add a comment how to
proceed. If we don't get bugreports, this should be removed after a while
(some releases?).
PR: 56617 [1], 29465, 39260, 40574, 68225
Submitted by: Matthew E. Gove <mgove@comcast.net> [1]
* Add kernel hint option to disable DXS channels entirely. Report
from several skype users / Pav Lucistnik indicate that disabling
DXS may fix lots of pop / crackling noise. To disable DXS add
hint.pcm.<unit>.via_dxs_disabled="1" to /boot/device.hints.
Further investigation of the issues regarding DXS showed, that
the problem is in another (more generic) place, but until the
right fix is tested/reviewed this may help a little bit.
Added sysctl's to aid testing/debugging:
hint.pcm.<unit>.via_dxs_disabled=X - Disable / Enable DXS channels entirely
hint.pcm.<unit>.via_dxs_channels=X - Limit DXS channels up to X
hint.pcm.<unit>.via_sgd_channels=X - Limit SGD channels up to X
hint.pcm.<unit>.via_dxs_src=X - Enable / Disable DXS sample rate
converter.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
especially for CT4730 / EV1938 chip, causing misconfigured mixer
(David Xu), crippled after power cycle (Kevin Oberman). Fixed.
* Incorporate locking/spdif patches from Jon Noack / matk. Not all
es137x can really do spdif, clean it up a bit to only let few capable
chip. This adds a "hw.snd.pcm<unit>.spdif_enabled" sysctl until
a more generic way of handling this from userland (by an ordinary
user) is designed/implemented.
* Convert all bus_space_(read|write) to use es_rd/es_wr, simmilar
with other drivers.
* Add tunable hw.snd.pcm<unit>.latency_timer sysctl to toggle pci
latency timer value on the fly. Much noise / pop / crackling
issues can be solved by increasing its value. Other people have
pointed out to use pciconf instead, but this is just an added
value specific for CT4730/EV1938.
* Remove es137x specific debug sysctl/code.
Several PRs can now be closed.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Submitted by: Jon Noack <noackjr@alumni.rice.edu> (implicit)
Submitted by: matk (implicit)
PR: 59349, 68594, 73498
Tested by: multimedia@
handle DMA addresses located above 1GB. The LBA(loop begin address)
register which holds DMA base address is 32bits register. But the
MSB 2bits are used for other purposes. This effectivly limits the
DMA address space up to 1GB.
Approved by: jake (mentor)
Reviewed by: truckman, matk
assign DMA address to the wrong address. It can cause system lockup
or other mysterious errors. Since most sound cards requires low DMA
address(BUS_SPACE_MAXADDR_24BIT) sndbuf_alloc() would fail when the
audio driver is loaded after long running of operations.
Approved by: jake (mentor)
Reviewed by: truckman, matk
allocation. Notably, in this case, the driver tries to allocate several
pieces of memory and then fails if the pieces allocated after the first
do not come after it physically, and within a specific range (8MB I
believe). Of course, this could just as easily fail for any number of
reasons, but it almost always fails now that contiguous allocations start
at the end of possible specified memory locations rather than the beginning.
Allocate all the possibly-needed memory up front, even though it's a waste,
to get around this. The least bogus solution would be to take the physical
address from the first allocation and create a new tag that specified that
further allocations must follow it within that 8MB window, then use that
when allocating new channels, but that's left for anyone else that really
feels like doing it.
Tested by: Erwin Lansing <erwin@lansing.dk>
- `sound'
The generic sound driver, always required.
- `snd_*'
Device-dependent drivers, named after the sound module names.
Configure accordingly to your hardware.
In addition, rename the `snd_pcm' module to `sound' in order to sync
with the driver names.
Suggested by: cg
exactly as done in the cmi driver. I am quite confident this is
safe since I'm runing this for more than two weeks now, on an SMP
box. A few people tested this patch for me successfully as well.
of you with other cards, please do review and test the drivers for
MP-safety and disable Giant in the interrupt routines when you are
sure of proper functionality.
because they bogusly check for defined(INTR_MPSAFE) -- something which
never was a #define. Correct the definitions.
This make INTR_TYPE_AV finally get used instead of the lower-priority
INTR_TYPE_TTY, so it's quite possible some improvement will be had
on sound driver performance. It would also make all the drivers
marked INTR_MPSAFE actually run without Giant (which does seem to
work for me), but:
INTR_MPSAFE HAS BEEN REMOVED FROM EVERY SOUND DRIVER!
It needs to be re-added on a case-by-case basis since there is no one
who will vouch for which sound drivers, if any, willy actually operate
correctly without Giant, since there hasn't been testing because of
this bug disabling INTR_MPSAFE.
Found by: "Yuriy Tsibizov" <Yuriy.Tsibizov@gfk.ru>
read-only. Need to enable "legacy support", by poking
into pci config space. (comment from the patch)
Submited by: Autrijus Tang <autrijus@autrijus.org>
Approved by: tanimura (mentor)
This change has not been tested.
This change was triggered by a gcc(1) warning on ia64 at -O2. The
variable v was not used after being computed, which resulted in enough
dead code elimination (DCE) to confuse the compiler and emit a bogus
warning about the use of the variable i without prior definition. The
variable i is the loop variable.
Submitted by: des
Responsibility: marcel
This takes us a lot closer to refcounting dev_t.
This patch originally by cg@ with a few minor changes by me.
It is largely untested, but has been HEADSUP'ed twice, so presumably
people have not found any issues with it.
Submitted by: cg@
I started with a year-old patch by Orlando Bassotto
<orlando.bassotto@ieo-research.it>, and ported it to 5.2-CURRENT along with
fixing the problems working with pre-Audigy cards.
dsp_open: rearrange to only hold one lock at a time
dsp_close: ditto
mixer_hwvol_init: delete locking, the only consumer seems to
be the ess driver and it only call it a creation time, I
think the device will be stable across the sleepable malloc.
cmi interrupt routine: Release locks while caller chn_intr,
either this or do what emu10k1 does which is have no locks
at in the interrupt handler.
Submitted by: mat@cnd.mcgill.ca
round the result up to a multiple of 4 bytes so that it will always
be a multiple of the sample size. Also use the actual buffer size
from sc->bufsz instead of the default DS1_BUFFSIZE.
This fixes panics and bad distortion I have seen on Yamaha DS-1
hardware, mainly when playing certain Real Audio media.
Reviewed by: orion (an earlier version of the patch)
tested for playback.
* modify device name strings for ich chips to better conform with their
common names.
* remove superflous 'AC97 controller' from nforce device names.
MFC after: 1 week
Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma. At the moment, this is used for the
asynchronous busdma_swi and callback mechanism. Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg. dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create(). The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.
sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms. The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.
If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.
Reviewed by: tmm, gibbs
codec during initialization. This code mirrors the reset code used on
the VIA82c686 and fixes a codec initialization failure (SoundMAX
AD1885) reported by Matthias Schuendehuette.
actual address of buffer descriptor. This should fix the reported
calibration failures and subsequent speed problems with ich chipsets.
Minor calibration comment updates.
and be prepared to wait much longer for codec to become ready.
Credit to Hugo D. Valentim <hvalentim@gmx.net> for reporting the
problem, providing useful pointers, and repeated diff testing.
channel and disable DXS3. Several users have reported DXS3 as playing
at half speed on the 8233A revision of the chipset. This implicitly
means no SPDIF for VIA8233A users.
o Make DXS3 the primary playback channel. It may be the only
universally supported channel with the assorted revisions of this
chipset.
o Add sysctl and handler for enabling s/pdif output from DXS3.
- add bus capabilities into the driver, so that it is possible for a radio
driver to attach to it to use shared resources of fm801 chip. The radio
driver itself will be committed later.
MFC after: 1 week
circumstances. The problem was only reported with -stable, but it's
obviously wrong in -current also. MFC is forthcoming.
Submitted by: doconnor@dsoft.com.au
Attempt to determine what function of AUX_OUT is: "True line level
out", "Headphone out", or "4-Channel out" and frig OSS mixer label
accordingly.
Addresses problem raised by Randy Bush on -multimedia of not being
able to hear audio on ich2 m/b which was eventually found to be
because the mixer monitor value was 0. On this h/w the label
"monitor" should now be presented as the marginally more intuitive
"ogain".
is that some cards built around fm801 chip have the same device ID, only
have radio tuner onboard, but no sound capabilities. Therefore, with such
card inserted and `device pcm' in kernel the user has a big problem, as
the fm801 driver effectively hangs the machine when trying to initialise
nonexistent ac97 codecs (it does 500 retries with 1 second interval!).
It would be better if MediaForte's engeneers were smart enough to put
different device ID into such cards, but it isn't an option.
MFC after: 2 weeks
calibrated. This fixes the problem where playback and recording do
not run at the correct speed. It probably also eliminates the
need for the hacks/workarounds/sysctl's that were previously
devised to deal with this, but I will leave that for a different
time.
Reviewed by: orion
otherwise we might get interrupts and are unable to
handle them properly, which results in a page fault.
PR: kern/39549
Submitted by: Gil Kloepfer <gil@arlut.utexas.edu>
the per-channel bus_addr_t offset. Also, cast the offset to (long long)
and use %#llx instead of %#x to fix printf warnings on architectures where
sizeof(bus_addr_t) != sizeof(int).
es137x.c: In function `es1371_rdcd':
es137x.c:598: warning: `x' might be used uninitialized in this function
PR: kern/35408
Submitted by: Thomas Quinot <thomas@cuivre.fr.eu.org>
The extra microphone channel capability is part of the "normal" ac97
capabilities and not an extended ac97 capability. Now recording on
codecs without a seperate mic channel works.
MFC after: 1 week
most cases NULL is passed, but in some cases such as network driver locks
(which use the MTX_NETWORK_LOCK macro) and UMA zone locks, a name is used.
Tested on: i386, alpha, sparc64
