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.
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
- 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.
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().
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
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
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
