- 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)
Add support for Intel High Definition Audio Controller.
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.
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>
commit.
1) sys/dev/sound/pcm/sound.h
sys/dev/sound/pcm/channel.c
* Be more specific: SD_F_SOFTVOL -> SD_F_SOFTPCMVOL
2) sys/dev/sound/pcm/mixer.[ch]
* Implement
mix_setparentchild()
mix_setrealdev()
mix_getparent()
mix_getchild()
The purpose of these functions is implement relative volume
adjustment, such as to tie two or more mixer device into a
single logical device. Usefull for the upcoming HDA driver
and few AC97 codec (such as AD1981B) where the master volume
"vol" need to be implemented using this logical manner.
3) sys/dev/sound/pcm/ac97_patch.[ch]
* Patch for AD1981B codec to enable (automuting) headphone jack sense.
4) sys/dev/sound/pcm/ac97.c
* Implement proper logical master volume for AD9181B codec
through various mix_set{parentchild,realdev}(). Tie both
"ogain" (headphone volume) and "phone" (speaker/lineout) to
a logical "vol".
5) sys/dev/sound/pcm/usb/uaudio_pcm.c
* ditto, for "vol" -> { "pcm" }.
MFC after: 1 month
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
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
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
--------------------
- Seal the fate of long standing memory leak (4 years, 7 months) during
pcm_unregister(). While destroying cdevs, scan / detect possible
children and free its SLIST placeholder properly.
- Optimize channel allocation / numbering even further. Do brute cyclic
checking only if the channel numbering screwed.
- Mega vchan create/destroy cleanup:
o Implement pcm_setvchans() so everybody can use it freely instead
of implementing their own, be it through sysctl or channel auto
allocation.
o Increase vchan creation/destruction resiliency:
+ it's possible to increase/decrease total vchans even during
busy playback/recording. Busy channel will be left alone, untouched.
Abusive test sample:
# play whatever...
#
while : ; do
sysctl hw.snd.pcm0.vchans=1
sysctl hw.snd.pcm0.vchans=10
sysctl hw.snd.pcm0.vchans=100
sysctl hw.snd.pcm0.vchans=200
done
# Play something else, leave above loop running frantically.
+ Seal another 4 years old bug where it is possible to destroy (virtual)
channel even when its cdevs being referenced by other process.
The "First Come First Served" nature of dsp_clone() is the main
culprit of this issue, and usually manifest itself as dangling
channel <-> process association. Ensure that all of its cdevs
are free from being referenced before destroying it (through
ORPHAN_CDEVT() macross).
All these fixes (including previous fixes) will be MFCed, later.
to avoid possible device unregister race (impossible to reproduce, yet
possible).
- Extra sanity check to ensure proper parent channel is being selected.
- Reset parent channel once all of its children gone.
- [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
- Determine open direction using 'flags', not 'mode'. This bug exist since
past 4 years.
- Don't allow opening the same device twice, be it in a same or different
direction.
- O_RDWR is allowed, provided that it is done by a single open (for example
by mixer(8)) and the underlying hardware support true full-duplex operation.
- Do various paranoid checking in case other process/thread trying to hijack
the same device twice (or more).
MFC after: 5 days
especially for vchans. It turns out that channel numbering always depend
on d->devcount counter (which keep increasing), while PCMMKMINOR() truncate
everything to 8bit length. At some point the truncation cause the newly
created character device overlapped with the existence one, causing erratic
overall system behaviour and panic. Easily reproduce with something like:
(Luckily, only root can reproduce this)
while : ; do
sysctl hw.snd.pcm0.vchans=200
sysctl hw.snd.pcm0.vchans=100
done
- Enforce channel/chardev numbering within 8bit boundary. Return E2BIG
if necessary.
- Traverse d->channels SLIST and try to reclaim "free" counter during channel
creation. Don't rely on d->devcount at all.
- Destroy vchans in reverse order.
Anyway, this is not the fault of vchans. It is just that vchans are so cute
and begging to be abused ;) . Don't blame her.
Old, hidden bugs.. sigh..
MFC after: 3 days
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
dereferencing) since a NULL value would be a bug here.
Note: Both affected functions look very similar. A refactoring may
be beneficial.
CID: 483, 485
Found with: Coverity Prevent(tm)
Discussed with: ariff
MFC after: 5 days
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
This is supposed to fix some Coverity Prevent errors (Ariff didn't
looked at the CID's (ENOTIME), I just told him that there are some problems
in function dsp_ioctl()).
CID: 215-218
Found with: Coverity Prevent(tm)
Submitted by: ariff
MFC after: 5 days
(1) Fix DMA alignment, based on bytes per sample.
feeder_rate.c:
Handle strayed bytes (mostly caused by #1) better.
This DMA alignment issues are extremely hard to reproduce unless
the user happen to have a 32bit capable soundcards (ATI IXP) and
knowledgeable enough to force it to operate under pure 32bit
operations on both record and play directions.
feeder.h:
feeder.c:
- Implement scoring mechanisme to select best format for conversion.
This is actually part of newer format chaining procedures which
will be commited someday. Confusion during chaining process solved
by this scoring since it will try to reduce list of from/to formats
to a single, best format.
Related PR: kern/91683
channel.c:
- Simplify feeder building process since we have smarter format
chaining.
feeder_fmt.c:
- Add few more sign conversion feeders for 24 and 32 bit format.
feeder_rate.c:
- Force buffer / bytes allignment. Unaligned buffer may cause
panics during recording on pure 32bit sample format if it
involves feeder_rate as part of feeders chain.
Tested on: ATI IXP, force 32bit recording.
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
This should reduce huge playback / recording latency for
applications that try to act smarter and manage their own
buffering (XMMS, Skype, etc.).
Note to Skype + via8xxx users: Remove previous hackish
"hint.pcm.<unit>.via_dxs_disabled" from kernel hint and see
whether this changes cure all those annoying sound issues.
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
This one simply tries to simplify the logic to select the
buffer sizes. I am not sure it is necessary but the code
seems a bit more readable to me. And at least i have tried
to document how the buffer sizes are computed.
Thanks to luigi for deciphering one of the most cryptic part of
sound driver.
Submitted by: luigi
Approved by: netchild (mentor)
In SNDCTL_DSP_SETFRAGMENT, if you specify both read and
write channels, the existing code first acts on the
read channel, but as a side effect it updates the
arguments (maxfrags, fragsz) passed by the caller according
to acceptable values for the read channel, and then uses the
modified values to act on the write channel.
The problem with this approach is that, given a
(maxfrags, fragsz) user-specified value, the actual
values computed by the read and write channels may differ:
e.g. the read channel might want to allocate more fragments
than what the user specified because it has no side-effects
on the delay and it helps in case of slow readers,
whereas the write channel needs to use as few fragments
as possible to keep the audio latency low (very important
with telephony apps).
This patch stores the values computed by the read channel
into temproary variables so the write channel will use
the actual arguments of the ioctl.
This patch is very helpful with telephony apps such as asterisk.
Submitted by: luigi
Approved by: netchild (mentor)
- Added new codec id for CX20468-21 and VIA1617A.
Submitted by: Chen Lihong <lihong.chen@gmail.com>
- Re-enable SOUND_MIXER_IGAIN, but set the default level as 0 (mute)
Suggested by: luigi
mixer.c:
- Set default value for SOUND_MIXER_IGAIN as 0 (mute) to avoid
feedback problems on some laptops (was disabled by jhb during
ac97.c revision 1.42).
Approved by: netchild (mentor)
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)
mask to recdev_l and recdev_r, since each have its own unique mask.
Submitted by: Watanabe Kazuhiro <CQG00620@nifty.ne.jp>
Approved by: netchild (mentor)
It may be the case that you may hear some unwanted noise while
playing back with 24/32 bit. This is a problem in the USB system.
Explanation from Hans Petter Selasky:
---snip---
The current USB sound driver only uses one isochronous
buffer, that is restarted when it is completed. This will lead to a short
period of time, +1ms, where no sound data is sent to the external USB device.
Depending on the load of your computer, this can be as much as 50ms. So the
USB sound driver must use 2 isochronous transfers. At the beginning one will
queue both. Then these are restarted on completion. This will result in a
constant-rate data stream to the external sound device, a minimum sound
buffer equal to the size of the isochronous buffer, and possibly the sound
will reach your ears with less delay. Little delay is a result of constant
data rate. Currently only my USB driver will support that. If one tries that
with the USB driver in *BSD, then it will crash at the first moment one gets
a buffer underrun.
---snip---
Submitted by: Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp>
Mono-recording still not tested by: julian
- Return EINVAL if play_format or rec_format is set but the corresponding
sample rate is 0.
- Don't try to set the playback or recording format to 0. Previously,
issuing an AIOSFMT ioctl with an all-zeroes snd_chan_param would
trigger a KASSERT in chn_fmtchain(); I'm unsure about the effects on
a kernel without INVARIANTS. After this commit, issuing AIOSFMT with
an all-zeroes snd_chan_param is equivalent to issuing AIOGFMT.
MFC after: 2 weeks
sampling rate:
- Improve vchan chn_setspeed() strategy. Try to avoid FEEDER_RATE
on parent channel if the requested value is not supported
by the hardware.
- Fix vchan default speed calculation. In any case, vchan should
rely on parent bufsoft speed instead of bufhard since it is
possible that the entire feeder chain might involve FEEDER_RATE.
This is possible under extreme, rare condition if the above
chn_setspeed() strategy failed.
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]
* General spl* cleanup. It doesn't serve any purpose anymore.
* Nuke sndstat_busy(). Addition of sndstat_acquire() /
sndstat_release() for sndstat exclusive access. [1]
sys/dev/sound/pcm/sound.c:
* Remove duplicate SLIST_INIT()
* Use sndstat_acquire() / release() to lock / release the entire
sndstat during pcm_unregister(). This should fix LOR #159 [1]
sys/dev/sound/pcm/sound.h:
* Definition of SD_F_SOFTVOL (part of feeder volume)
* Nuke sndstat_busy(). Addition of sndstat_acquire() /
sndstat_release() for exclusive sndstat access. [1]
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
LOR: 159 [1]
Discussed with: yongari [1]
* Added codec id for CMI9761.
* feeder_volume *whitelist* through ac97_fix_volume()
sys/dev/sound/pcm/ac97.h:
* Added AC97_F_SOFTVOL definition.
sys/dev/sound/pcm/channel.c:
* Slight changes for chn_setvolume() to conform with OSS.
* FEEDER_VOLUME is now part of feeder building process.
sys/dev/sound/pcm/mixer.c:
* General spl* cleanup. It doesn't serve any purpose anymore.
* Main hook for feeder_volume.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
