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.
Rename MAX_SAMPLE_RATES macro to OSS_MAX_SAMPLE_RATES. The old
macro clashed with those used in other applications and libaries
(ex: RtAudio). 4Front responded by updating their spec, so we
will follow suit.
Submitted by: ryanb
Noticed by: pointyhat/kris
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.
- 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.
--------------------
- 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.
- 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
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
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
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.
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)
- 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)
* 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@
While I'm here add KASSERT(9) to notify failure of SYSUNINIT handler.
Reported by: Ben Kaduk < minimarmot AT gmail DOT com >
Tested by: Ben Kaduk < minimarmot AT gmail DOT com >
- Remove an assertion in sound.c, it's not needed (and causes a panic now).
From the conversation via mail between glebius and Ariff:
---snip---
> Well, but which mutex protects now? Do we own anything else
> in pcm_chnalloc()? I see some queue(4) macros in pcm_chnalloc(),
> they should be protected, shouldn't they?
Queue insertion/removal occur during
1) driver loading (which is pretty much single thread /
sequential) or unloading (mutex protected, bail out if there is
any channel with refcount > 0 or busy).
2) vchan_create()/destroy(), (which is *sigh* quite complicated), but
somehow protected by 'master'/parent channel mutex. Other
thread cannot add/remove vchan (or even continue traversing
that queue) unless it can acquire parent channel mutex.
---snip---
Fix the locking in dsp.c to prevent a LOR (AFAIK not on the LOR page).
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested with: INVARIANTS[1] and DIAGNOSTICS[2]
Tested by: netchild [1,2], David Reid <david@jetnet.co.uk> [1]
* New definition CHN_F_HAS_VCHAN.
- channel.c
* Use CHN_F_HAS_VCHAN to mark channel with vchan capability instead
of relying on SLIST_EMPTY(&channel->children) == true for better
clarification and future possible usages of children (like
'slave' channel).
* Various fixes, including blocksize / format bps allignment,
better 24bit seeking (mplayer, others).
* Improve format chain building, it's now possible to record something
to a format non-native to the soundcard through various feeder format
converters or to higher sampling rate. This also gains another feature,
like doing vchan mixing on non s16le soundcard such as sb8.
- sound.c
* Increase robustness within various function that handle vchan
creation / termination (these function need a total rewrite, but
that would cause other major rewrite within various places too!).
As far as its robustness can be guaranteed, leave it as is.
* Optimize channel ordering, prefer *real* hardware playback
channels over virtual channels. cat /dev/sndstat should look
better.
* Increase sndstat verbosity to include bufsoft/bufhard allocation.
- vchan.c
* Fix LOR 119.
- http://sources.zabbadoz.net/freebsd/lor.html#119
* Reorder / increase robustness of vchan_create() / destroy().
Enforce destroy_dev() during destroy operation, fix possible
panic / dangling character device.
- http://lists.freebsd.org/pipermail/freebsd-current/2005-May/050308.html
* Tolerate a little bit more during mixing process, this should help
non s16le soundcards.
Note: Recoring in a non-native rate/format may result in overruns. A friendly
application is wavrec from audio/wavplay. The problem is under
investigation.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
event handler, dev_clone, which accepts a credential argument.
Implementors of the event can ignore it if they're not interested,
and most do. This avoids having multiple event handler types and
fall-back/precedence logic in devfs.
This changes the kernel API for /dev cloning, and may affect third
party packages containg cloning kernel modules.
Requested by: phk
MFC after: 3 days
1. Support wide range sampling rate, as low as 1hz up to int32 max
(which is, insane) through new feeder_rate, multiple precisions
choice (32/64 bit converter). This is indeed, quite insane, but it
does give us more room and flexibility. Plenty sysctl options to
adjust resampling characteristics.
2. Support 24/32 bit pcm format conversion through new, much improved,
simplified and optimized feeder_fmt.
Changes:
1. buffer.c / dsp.c / sound.h
* Support for 24/32 AFMT.
2. feeder_rate.c
* New implementation of sampling rate conversion with 32/64 bit
precision, 1 - int32max hz (which is, ridiculous, yet very
addictive). Much improved / smarter buffer management to not
cause any missing samples at the end of conversion process
* Tunable sysctls for various aspect:
hw.snd.feeder_rate_ratemin - minimum allowable sampling rate
(default to 4000)
hw.snd.feeder_rate_ratemax - maximum allowable sampling rate
(default to 1102500)
hw.snd.feeder_rate_buffersize - conversion buffer size
(default to 8192)
hw.snd.feeder_rate_scaling - scaling / conversion method
(please refer to the source for explaination). Default to
previous implementation type.
3. feeder_fmt.c / sound.h
* New implementation, support for 24/32bit conversion, optimized,
and simplified. Few routines has been removed (8 to xlaw, 16 to
8). It just doesn't make sense.
4. channel.c
* Support for 24/32 AFMT
* Fix wrong xruns increment, causing incorrect underruns statistic
while using vchans.
5. vchan.c
* Support for 24/32 AFMT
* Proper speed / rate detection especially for fixed rate ac97.
User can override it using kernel hint:
hint.pcm.<unit>.vchanrate="xxxx".
Notes / Issues:
* Virtual Channels (vchans)
Enabling vchans can really, really help to solve overrun
issues. This is quite understandable, because it operates
entirely within its own buffering system without relying on
hardware interrupt / state. Even if you don't need vchan,
just enable single channel can help much. Few soundcards
(notably via8233x, sblive, possibly others) have their own
hardware multi channel, and this is unfortunately beyond
vchan reachability.
* The arrival of 24/32 also come with a price. Applications
that can do 24/32bit playback need to be recompiled (notably
mplayer). Use (recompiled) mplayer to experiment / test /
debug this various format using -af format=fmt. Note that
24bit seeking in mplayer is a little bit broken, sometimes
can cause silence or loud static noise. Pausing / seeking
few times can solve this problem.
You don't have to rebuild world entirely for this. Simply
copy /usr/src/sys/sys/soundcard.h to
/usr/include/sys/soundcard.h would suffice. Few drivers also
need recompilation, and this can be done via
/usr/src/sys/modules/sound/.
Support for 24bit hardware playback is beyond the scope of
this changes. That would require spessific hardware driver
changes.
* Don't expect playing 9999999999hz is a wise decision. Be
reasonable. The new feeder_rate implemention provide
flexibility, not insanity. You can easily chew up your CPU
with this kind of mind instability. Please use proper
mosquito repellent device for this obvious cracked brain
attempt. As for testing purposes, you can use (again)
mplayer to generate / play with different sampling rate. Use
something like "mplayer -af resample=192000:0:0 <files>".
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
This mostly to help CT4730, but apparently it does help other
cards too (especially via8233x). This probably need further test
and confirmation from other people with ac97 cards other than via
/ es137x.
* Aggresive dac power wake up call, again, to help CT4730 (and
probably others).
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
Affects to people WITH an AD1888 codec, the system will output to the port
labeled "speaker" instead of microphone. System will work the same in
multiple operating systems.
If people are currently using their systems with this codec they will need
to swap their output ports.
I have _not_ checked audio input or line input (basically, I have checked
nothing other than line-out).
I believe this is an appropriate change, it makes us consistent with
documentation, and other operating systems. Furthermore, this feature
(playing) is the vast majority of sound activities, so if this makes is
right for playback and wrong for recording... playback is more important,
and we can fix recoding in the future without worries of screwing people
again in the future (since we'll be "right" on the playback).
Submitted by: David Cross
holds sndstat_lock across a call to uiomove(), which is not legal
to do with a mutex because of the possibility that the data transfer
could sleep because of a page fault. It is not possible to just
unlock the mutex for the uiomove() call without introducing another
locking mechanism to prevent the body of sndstat_read() from being
re-entered. Converting sndstat_lock to an sx lock is the least
complicated change.
This is a candidate for RELENG_5.
LOR: 030
MFC after: 4 days
- `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
The big lines are:
NODEV -> NULL
NOUDEV -> NODEV
udev_t -> dev_t
udev2dev() -> findcdev()
Various minor adjustments including handling of userland access to kernel
space struct cdev etc.
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>
rid of the MTX_DUPOK flag on channel mutexes, which allows witness to
do a better job of lock order checking. Nuke snd_chnmtxcreate() since
it is no longer needed.
Tested by: matk
channel at a time unless it is actually necessary to lock both.
This avoids problems with lock order reversal and malloc() calls
with a mutex held when lower level code unlocks a channel, calls malloc(),
and relocks the channel. This also avoids the cost of some unnecessary
locking and unlocking.
Tested by: matk
Introduce d_version field in struct cdevsw, this must always be
initialized to D_VERSION.
Flip sense of D_NOGIANT flag to D_NEEDGIANT, this involves removing
four D_NOGIANT flags and adding 145 D_NEEDGIANT flags.
panic() so that the buffer overflow just beyond this point is always
caught, even when the code is not compiled with INVARIANTS.
Change chn_setblocksize() buffer reallocation code to attempt to avoid
the feed_vchan16() buffer overflow by attempting to always keep the
bufsoft buffer at least as large as the bufhard buffer.
Print a diagnositic message
Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE()
if our best attempts fail. If feed_vchan16() were to be called by
the interrupt handler while locks are dropped in chn_setblocksize()
to increase the size bufsoft to match the size of bufhard, the panic()
code in feed_vchan16() will be triggered. If the diagnostic message
is printed, it is a warning that a panic is possible if the system
were to see events in an "unlucky" order.
Change the locking code to avoid the need for MTX_RECURSIVE mutexes.
Add the MTX_DUPOK option to the channel mutexes and change the locking
sequence to always lock the parent channel before its children to avoid
the possibility of deadlock.
Actually implement locking assertions for the channel mutexes and fix
the problems found by the resulting assertion violations.
Clean up the locking code in dsp_ioctl().
Allocate the channel buffers using the malloc() M_WAITOK option instead
of M_NOWAIT so that buffer allocation won't fail. Drop locks across
the malloc() calls.
Add/modify KASSERTS() in attempt to detect problems early.
Abuse layering by adding a pointer to the snd_dbuf structure that points
back to the pcm_channel that owns it. This allows sndbuf_resize() to do
proper locking without having to change the its API, which is used by
the hardware drivers.
Don't dereference a NULL pointer when setting hw.snd.maxautovchans
if a hardware driver is not loaded. Noticed by Ryan Sommers
<ryans at gamersimpact.com>.
Tested by: Stefan Ehmann <shoesoft AT gmx.net>
Tested by: matk (Mathew Kanner)
Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
device that doesn't exists. I'm using my discretion and
committing without mentor approval since Seigo is away.
Noticed by: Maxime Henrion <mux@freebsd.org>
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@
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
thread being waken up. The thread waken up can run at a priority as
high as after tsleep().
- Replace selwakeup()s with selwakeuppri()s and pass appropriate
priorities.
- Add cv_broadcastpri() which raises the priority of the broadcast
threads. Used by selwakeuppri() if collision occurs.
Not objected in: -arch, -current
first sample in the buffer to be ignored. The bug caused a repetitive
glitch in one of the stereo channels when playing mono sound on
configurations that use the monotostereo16 feeder.
Reviewed by: orion
o AD1980 hook.
o ac97_fix_auxout.
and:
o Associate AC97_MIX_AUXOUT with SOUND_MIXER_OGAIN rather than
SOUND_MIXER_MONITOR.
o Add ac97_fix_tone to remove tone controls from mixer if invalid.
when the user specifies a maximum fragment size < 2.
This is the behavior that Linux provides and fixes the problem I've
observed in Tribes2 where sounds effects are delayed by 1/2 a second.
where physical addresses larger than virtual addresses, such as i386s
with PAE.
- Use this to represent physical addresses in the MI vm system and in the
i386 pmap code. This also changes the paddr parameter to d_mmap_t.
- Fix printf formats to handle physical addresses >4G in the i386 memory
detection code, and due to kvtop returning vm_paddr_t instead of u_long.
Note that this is a name change only; vm_paddr_t is still the same as
vm_offset_t on all currently supported platforms.
Sponsored by: DARPA, Network Associates Laboratories
Discussed with: re, phk (cdevsw change)
branches:
Initialize struct cdevsw using C99 sparse initializtion and remove
all initializations to default values.
This patch is automatically generated and has been tested by compiling
LINT with all the fields in struct cdevsw in reverse order on alpha,
sparc64 and i386.
Approved by: re(scottl)
- Get rid of the useless atop() / pmap_phys_address() detour. The
device mmap handlers must now give back the physical address
without atop()'ing it.
- Don't borrow the physical address of the mapping in the returned
int. Now we properly pass a vm_offset_t * and expect it to be
filled by the mmap handler when the mapping was successful. The
mmap handler must now return 0 when successful, any other value
is considered as an error. Previously, returning -1 was the only
way to fail. This change thus accidentally fixes some devices
which were bogusly returning errno constants which would have been
considered as addresses by the device pager.
- Garbage collect the poorly named pmap_phys_address() now that it's
no longer used.
- Convert all the d_mmap_t consumers to the new API.
I'm still not sure wheter we need a __FreeBSD_version bump for this,
since and we didn't guarantee API/ABI stability until 5.1-RELEASE.
Discussed with: alc, phk, jake
Reviewed by: peter
Compile-tested on: LINT (i386), GENERIC (alpha and sparc64)
Runtime-tested on: i386
without waiting, since they are called from a system-call context only.
This appears to fix all sorts of problems with open("/dev/dsp", O_WRONLY)
randomly returning ENXIO.
Found by: cognet
Sync with userland test framework which now deals better with pcm feeder kobj
emulation.
Reduce max rate from 96kHz to 48kHz as userland tests found a few bad
points about 90kHz and we don't care about operating up there for now.
of knowing data size transformations of feeder chain and in some cases
this means too much data is pulled through chain, eg converting input
stream from 16bits to 8bits on 16bit only h/w.
PR: kern/37831
Submitted by: Harti Brandt <brandt@fokus.fraunhofer.de>
between any pair of values in range 4-96kHz. Thanks to Ken Marks for
discovering there were problems with the previous version.
o Use a non-recursive gcd routine.
