seems not enough to verify its consistencies.
- Define AC97_MIXER_SIZE as SOUND_MIXER_NRDEVICES (25), since we
don't need more than that. Stop doing wild and random guess about
its size since we're stricly bound to it.
sysctl_handle_int is not sizeof the int type you want to export.
The type must always be an int or an unsigned int.
Remove the instances where a sizeof(variable) is passed to stop
people accidently cut and pasting these examples.
In a few places this was sysctl_handle_int was being used on 64 bit
types, which would truncate the value to be exported. In these
cases use sysctl_handle_quad to export them and change the format
to Q so that sysctl(1) can still print them.
Things can get ugly without it due to uninitialized class. RELENG_6 need
a simmilar, but different treatment as well.
err.. perhaps we should teach devclass_get_maxunit() to return -1 ?
MFC after: 1 day
- Rework the entire pcm_channel structure:
* Remove rarely used link placeholder, instead, make each pcm_channel
as head/link of each own/each other. Unlock - Lock sequence due to
sleep malloc has been reduced.
* Implement "busy" queue which will contain list of busy/active
channels. This greatly reduce locking contention for example while
servicing interrupt for hardware with many channels or when virtual
channels reach its 256 peak channels.
- So I heard you like v chan ... O RLY?
Welcome to Virtual **Record** Channels (vrec, rec vchans, vchans for
recording, Rec-Chan, you decide), the ultimate solutions for your
nagging O_RDWR full-duplex wannabe (note: flash plugins) monopolizing
single record channel causing EBUSY. Vrec works exactly like Vchans
(or, should I rename it to "Vplay" :) , except that it operates on the
opposite direction (recording). Up to 256 vrecs (like vchans) are
possible.
Notes:
* Relocate dev.pcm.%d.{vchans,vchanformat,vchanrate} to each of its
respective node/direction:
dev.pcm.%d.play.* for "play" (cdev = dsp%d.vp%d)
dev.pcm.%d.rec.* for "record" (cdev = dsp%d.vr%d)
* Don't expect that it will magically give you ability to split
"recording source" (eg: 1 channel for cdrom, 1 channel for mic,
etc). Just admit that you only have a *single* recording source /
channel. Please bug your hardware vendor instead :)
- Bump maxautovchans from 4 to 16. For a full-fledged multimedia
desktop/workstation with too many soundservers installed (esound,
artsd, jackd, pulse/polypaudio, ding-dong pling plong mudkip fuh fuh,
etc), 4 seems inadequate. There will be no memory penalty here, since
virtual channels are allocate only by demand.
- Nuke/Rework the entire statically created cdev entries. Everything is
clonable through snd own clone manager which designed to withstand many
kind of abusive devfs droids such as:
* while : ; do /bin/test -e /dev/dsp ; done
* jot 16777216 0 | while read x ; do ls /dev/dsp0.$x ; done
* hundreds (could be thousands) concurrent threads/process opening
"/dev/dsp" (previously, this might result EBUSY even with just
3 contesting threads/procs).
o Reusable clone objects (instead of creating new one like there's no
tomorrow) after certain expiration deadline. The clone allocator will
decide whether to reuse, share, or creating new clone.
o Automatic garbage collector.
- Dynamic unit magic allocator. Maximum attached soundcards can be tuned
using tunable "hw.snd.maxunit" (Default to 512). Minimum is 16, and
maximum is 2048.
- ..other fixes, mostly related to concurrency issues.
joel@ will do the manpage updates on sound(4).
Have fun.
by the subsequent mix_setdevs() and friends.
- Minor style(9) declaration arrangement nit.
Requested by: joeld
Submitted by: pluknet <pluknet@gmail.com>
Neither me nor Ariff have access to any of this hardware, so all tests
have been made by Konstantin and Artem. Commit message mostly written
by Konstantin.
envy24:
- Add test code to support rear line-in input on 'Terratec DMX 6fire'
audio card. This code is also intended to be used in the future for
support of cards, that have I2C-to-GPIO expanders wired between the
control line of the audio codec and the Envy24, however such cards
are too complex and i can't add that support without hardware sample
of such board, i've already tried and failed.
envy24ht:
- Add support for 'AudioTrak Prodigy HD2'.
- Add support for 'AudioTrak Prodigy 7.1 XT'.
- Add support for 'ESI Juli@' (Works ok, DAC volume is hard-coded for
the time being, so 'mixer vol ...' doesn't work, only 'mixer pcm
...' works). [1]
- Fix bug in the init data for M-Audio Revolution 5.1, that
results in distorted sound.
- Add software volume control (now 'mixer pcm' works, thanks to Ariff).
- Add support for more samples rates - 176.4kHz and 192kHz.
- Fix problem with the 192kHz samples rate playback when 24.576MHz
crystal is used on the board instead of 49.152MHz crystal.
spicds:
- Add support for Asahi Kasei flagship DAC - AK4396 (used in AudioTrak
Prodigy HD2).
Submitted by: Konstantin Dimitrov <kosio.dimitrov@gmail.com>
Tested by: Artem Antonov [1]
Reviewed by: ariff
- http://www.intel.com/design/chipsets/specupdt/245051.htm
AC97 Soft Audio and Soft Modem Master Abort Errata
Issue:
Use of either soft audio or soft modem on an Intel® 82443MX PCISet
based platform running a 100 MHz Processor System Bus and an AC97 codec
may result in failures. The system continues to function normally while
the AC97 hardware may not resume and may require a cold-boot to
recover. As a result of the failure, the Master Abort Status bit will
be set in the audio or modem function PCI header space.
Workaround:
Force uncacheable DMA on both BDL and pcm buffers.
Tested by: Emil Holmstr|m <emil@linux.se>
- Remove explicit call to pmap_change_attr(), since we now have proper
and functional definition of BUS_DMA_NOCACHE.
- Enable PCI(e) bus snooping for non i386/amd64 as an alternative for
uncacheable DMA.
- Codecs changes:
* Analag Device -> Analog Devices, AD1988.
* New codec: VIA VT1708 and VT1709, Realtek ALC262, ALC861-VD and
ALC885.
* Various fixups for Conexant Waikiki, fix recording (read: microphone)
on various Analog Devices codecs due to vendor BIOS mess, various
quirks for several ASUS laptops/boards.
- Fix connection list handling, closely following the specification to
handle range of nids.
- Basic Jack sense polling infrastructure for possible hardwares with
broken unsolicited response interrupt.
Ideas/Submitted/Tested by: Andriy Gapon <avg@icyb.net.ua>,
#freebsd-azalia, many.
and should only be applied on certain specific card / vendor, hence the
addition of ac97_getsubvendor().
- Fix low volume issue on several MSI laptops through ALC655 quirk.
Reported/Tested by: Christian Mueller
<raptor-freebsd-multimedia@xpls.de>
MFC after: 1 week
execution should help us avoiding potential deadlock and illegal locking
while sleeping in various mixer -> usb calls. To enable it, use
hint.uaudio.%d.async="1" or sysctl dev.uaudio.%d.async=1. Default is
disable, to remain compatible with old behaviour (with slight risk of
potential deadlock).
on amd64 and i386) until we gain proper BUS_DMA_NOCACHE support.
(in progress).
Tested by: rafan, infofarmer, Nguyen Tam Chinh <unixvn@gmail.com>
Tested on: amd64, i386
- SWAPLR quirk for (unknown, luckily it is mine) broken uaudio stick.
Fixing by rewiring is impossible without damaging it. Luckily,
we can fix it using "other" methods :) .
- Add uaudio_get_vendor(), _product() and _release() in uaudio.c
(currently used by uaudio_pcm quirk).
- Implement CHANNEL_SETFRAGMENTS().
- Drop channel locking in few places where it is about to sleep
somewhere. This should help eliminating illegal locking acquisition
where the current thread is about to sleep, and also few deadlock
cases. Dropping it right here is quite safe since it is already
protected by CHN_F_BUSY flag and other threads won't bother to touch it.
Solving other illegal locking issues are quite tricky without converting
most usbd_do_request() calls to its equivalent _async() calls,
which I intend to do it later after getting full test report from
other people with different uaudio hardwares.
- Fix memory leak issues during detach. This seems common to any drivers
(notably emu10kx, csapcm?) with bridge functions.
Implement CHANNEL_SETFRAGMENTS() for snd_atiixp, snd_es137x, snd_hda
and snd_via8233. CHANNEL_SETBLOCKSIZE() will basically call
CHANNEL_SETFRAGMENTS() internally using conservative blocksize /
blockcount hints. Other drivers will be converted later.
- Disable stray buffer management, since sample size aligned buffering
are pretty much guaranteed through out the entire feeder_* chain
processes.
- Few style(9) cleanups.
channel.c/channel_if.m:
- Macros cleanups, prefer inlined min() over MIN().
- Rework chn_read()/chn_write() for better dead interrupt detection
policy. Reduce scheduling overhead by doing pure 5 seconds sleep
before giving up, instead of several cycle of brute micro sleeping.
- Avoid calling wakeup_one() for non-sleeping channel (for example,
vchan parent channel).
- EWOULDBLOCK -> EAGAIN.
- Fix possible divide-by-zero panic on chn_sync().
- Re-enforce ^2 blocksize policy, since there are too many broken
userland apps that blindly assume it without even trying to do
serious calculations.
- New channel method - CHANNEL_SETFRAGMENTS(), a refined version of
CHANNEL_SETBLOCKSIZE(). It accept _both_ blocksize and blockcount
arguments, so the driver internals will have better hints for
buffering and timing calculations.
- Hook FEEDER_SWAPLR into feederchain building process.
feeder_fmt.c:
- Unified version of various filters, avoiding duplications.
- malloc()less feeder_fmt. Informations can be retrieved dynamically
by doing table lookup on static data. For cases such as converting
from stereo to mono or reducing bit depth where input data is larger
than output, cycle remaining available free space until it has been
exhausted and start kicking 8 bytes reservoir space from there to
complete the remaining requested count.
- Introduce FEEDER_SWAPLR. Few super broken hardwares (found on several
extremely cheap uaudio stick, possibly others) mistakenly wired left
and right channels wrongly, screwing output or input.
- Rearrange FEEDER_* constants starting from 0 to 31, so the future
additions will be much easier and consistent.
- Introduce FEEDER_SWAPLR. Few super broken hardwares (found on several
extremely cheap uaudio stick, possibly others) mistakenly wired left
and right channels wrongly, screwing output or input.
malloc()less feeder_vchan. Informations can be retrieved dynamically
by doing table lookup on static data. Reduce mixing overhead by
doing direct copy on first channel. Mixing process will begin starting
from second channel onwards.
malloc()less feeder_volume. Informations can be retrieved dynamically
by doing table lookup on static data. Increase resolution from 6bit
to PCM_FXSHIFT (8bit) for better resolution and finer volume changes.
- Convert sx lock to plain mutex. Since the access of /dev/sndstat
is pretty much exclusive and protected by toggling sndstat_isopen,
plain mutex is more than enough.
- Enable SBUF_AUTOEXTEND to avoid buffer truncation.
cache coherency, besides of causing train wreck in other places
(especially on amd64, possibly on i386).
Discussed with: kib@, rafan@
Tested by: rafan@
confusions and panic provided that the following conditions are met:
1) WITNESS is enabled (watch/trace).
2) Using modules, instead of statically linked (Not a strict
requirement, but easier to reproduce this way).
3) 2 or more modules share the same mtx type ("sound softc").
- They might share the same name (strcmp() == 0), but it always
point to different address.
4) Repetitive kldunload/load on any module that shares the same mtx
type (Not a strict requirement, but easier to reproduce this way).
Consider module A and module B:
- From enroll() - subr_witness.c:
* Load module A. Everything seems fine right now.
wA-w_refcount == 1 ; wA-w_name = "sound softc"
* Load module B.
* w->w_name == description will always fail.
("sound softc" from A and B point to different address).
* wA->w_refcount > 0 && strcmp(description, wA->w_name) == 0
* enroll() will return wA instead of returning (possibly unique)
wB.
wA->w_refcount++ , == 2.
* Unload module A, mtx_destroy(), wA->w_name become invalid,
but wA->w_refcount-- become 1 instead of 0. wA will not be
removed from witness list.
* Some other places call mtx_init(), iterating witness list,
found wA, failed on wA->w_name == description
* wA->w_refcount > 0 && strcmp(description, wA->w_name)
* Panic on strcmp() since wA->w_name no longer point to valid
address.
Note that this could happened in other places as well, not just sound
(eg. consider lots of drivers that share simmilar MTX_NETWORK_LOCK).
Solutions (for sound case):
1) Provide unique mtx type string for each mutex creation (chosen)
or
2) Put "sound softc" global variable somewhere and use it.
their latest Compaq V3000 BIOS (revision F.22). As a result, analog CD
connectivity is gone to the oblivion. Even if they decide to fix it in
future revisions, the damage has been done.
excessive interrupt clock timer reset, screwing interrupt generation
for already active channels. Track moving DMA pointer and call buffer
interrupt on each blocksize boundary.
PR: kern/109791
MFC after: 3 days
(external) microphone pin tend to screw it. Internal microphone (found
on several laptops) still need high VRef.
Tested by: Pietro Cerutti <pietro.cerutti@gmail.com>
lenix <irc.freenode.net>
changes. This should ease the job of maintaining codebase since much
of the regression tests are done across os versions.
- bus_setup_intr() -> snd_setup_intr().
Use bus_get_dma_tag() to obtain the parent DMA tag to make the drivers
a little bit more non-ia32/amd64 friendly.
There is no man page for bus_get_dma_tag, so this is modelled after
rev. 1.62 of src/sys/dev/sound/pci/es137x.c by marius.
Inspired by: commit by marius
approval, change the copyright statement to point at him instead of
"FreeBSD, Inc".
Encouraged by: rwatson
Reviewed by: imp
Discussed with and approved by: orion
/usr/share/examples/etc/bsd-style-copyright. I've fixed a
few minor wording and formatting differences.
Approved by: luigi, Hannu Savolainen <hannu@opensound.com>
buffer resizing, etc.) that was here since eon. Free all (unmanaged)
allocated buffer through sndbuf_destroy() in case we forgot to call
sndbuf_free(). For a managed buffer (mostly hw specific managed buffer),
either provide CHANNEL_FREE() method with appropriate return value to
invoke semi-automatic sndbuf_free() or simply do it on their own. If
everything is failed, sndbuf_destroy() will come to the rescue as a
final measure.
MFC after: 3 days
/usr/share/examples/etc/bsd-style-copyright. I've fixed a
few minor wording and formatting differences.
Approved by: matk, Hannu Savolainen <hannu@opensound.com>
Reviewed by: imp
unsolicited pin sense event and need manual control to turn off speaker
volume while attaching headphone.
Tested by: Ingeborg Hellemo <Ingeborg.Hellemo@cc.uit.no>
Disable global Acer + ALC883 headphone automute settings since there are
few models that does not respect this and causing broken behaviour.
Reported/Tested by: Pavel Argentov <argentoff@rtelekom.ru>
sparc64 GENERIC and the sound device drivers known working on sparc64
to use bus_get_dma_tag() to obtain the parent DMA tag so we can get rid
of the sparc64_root_dma_tag kludge eventually. Except for ath(4), sk(4),
stge(4) and ti(4) these changes are runtime tested (unless I booted up
the wrong kernels again...).
laptops.
Tested by: [1] Lion G. <liontanker@hotmail.com>
[2] Pietro Cerutti <pietro.cerutti@gmail.com>
Specialized mixer initialization for STAC9221, much like STAC9220.
Tested by: Devon H. O'Dell
revision 1.98 is NOT merged, because FreeBSD does not support this
syntax.
revision 1.99 is NOT merged, "const poisoning" part is not applicable
to FreeBSD. There is no variable shadowing, GCC can't find
this one (but there are others)
revision 1.100 is NOT merged, because it was null patch (no changes)
revision 1.101 is NOT merged, there is no BIT() macro in FreeBSD
revision 1.102 is merged
revision 1.103 is partially merged. There is no ai.ifaceh in FreeBSD
revision 1.104 is NOT merged
revision 1.105 is merged
revision 1.106 is not merged, because of rev. 1.107
revision 1.107 is a backuout of 1.106
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
---snip---
New features:
1. Optional multichannel recording (32 channels on Live!, 64 channels
on Audigy).
All channels are 16bit/48000Hz/mono, format is fixed.
Half of them are copied from sound output, another half can be
used to record any data from DSP. What should be recorded is
hardcoded in DSP code. In this version it records dummy data, but
can be used to record all DSP inputs, for example..
Because there are no support of more-than-stereo sound streams
multichannell stream is presented as one 32(64)*48000 Hz 16bit mono
stream.
Channel map:
SB Live! (4.0/5.1)
offset (words) substream
0x00 Front L
0x01 Front R
0x02 Digital Front L
0x03 Digital Front R
0x04 Digital Center
0x05 Digital Sub
0x06 Headphones L
0x07 Headphones R
0x08 Rear L
0x09 Rear R
0x0A ADC (multi-rate recording) L
0x0B ADC (multi-rate recording) R
0x0C unused
0x0D unused
0x0E unused
0x0F unused
0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0)
0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0)
0x12..-0x1F dummy
Audigy / Audigy 2 / Audigy 2 Value / Audigy 4
offset (words) substream
0x00 Digital Front L
0x01 Digital Front R
0x02 Digital Center
0x03 Digital Sub
0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards)
0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards)
0x06 Digital Rear L
0x07 Digital Rear R
0x08 Front L
0x09 Front R
0x0A Center
0x0B Sub
0x0C Side L
0x0D Side R
0x0E Rear L
0x0F Rear R
0x10 output to AC97 input L (muted)
0x11 output to AC97 input R (muted)
0x12 unused
0x13 unused
0x14 unused
0x15 unused
0x16 ADC (multi-rate recording) L
0x17 ADC (multi-rate recording) R
0x18 unused
0x19 unused
0x1A unused
0x1B unused
0x1C unused
0x1D unused
0x1E unused
0x1F unused
0x20..0x3F dummy
Fixes:
1. Do not assign negative values to variables used to index emu_cards
array. This array was never accessed when index is negative, but
Alexander (netchild@) told me that Coverity does not like it.
After this change emu_cards[0] should never be used to identify
valid sound card.
2. Fix off-by-one errors in interrupt manager. Add more checks there.
3. Fixes to sound buffering code now allows driver to use large playback
buffers.
4. Fix memory allocation bug when multichannel recording is not
enabled.
5. Fix interrupt timeout when recording with low bitrate (8kHz).
Hardware:
1. Add one more known Audigy ZS card to list. Add two cards with
PCI IDs betwen old known cards and new one.
Other changes:
1. Do not use ALL CAPS in messages.
Incomplete code:
1. Automute S/PDIF when S/PDIF signal is lost.
Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only).
---snip---
This commits enables a little bit of debugging output when the driver is
loaded as a module. I did a cross-build test for amd64.
The code has some style issues, this will be addressed later.
The multichannel recording part is some work in progress to allow playing
around with it until the generic sound code is better able to handle
multichannel streams.
This is supposed to fix
CID: 171187
Found by: Coverity Prevent
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
