configuration registers directly.
Remove pci_enable_io calls where they are redundant. The PCI bus driver
will set the right bits when the corresponding bus resource is activated.
Remove redundant pci_* function calls from suspend/resume methods. The
bus driver already saves and restores the PCI configuration.
Reviewed by: jhb
Approved by: kib (mentor)
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
that favours true hardware channel, the first instance of recording
request will grab this channel (the first channel is being used as
vchan master). In many cases, it is not really work as intended and give
false impression of broken recording.
PR: kern/118546
MFC after: 3 days
- 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>
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.
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
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.
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
- 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.
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
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
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)
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
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
- `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
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)
