deserves its own internet memes). The trick is to force all available,
unused pins (that being advertised as "speaker") to behave as microphone
pins instead.
Reported / Tested by: Dmitry Kutsenko <kutsenko.truebsd.org>
MFC after: 3 days
- Fix speaker issues with Dell Vostro 1500 (GPIO0)
Tested by: John Wright <jwright.gmail.com>
- Apply ridiculous quirk on Asus A8X series (A8JC, A8M, A8xx, etc). These
different laptop series share simmilar pci id, hardware codecs, etc.
but works differently. A slight difference in connection type for
widget #26 is used to differentiate it.
Tested by: eric baumbach <embaumbach.gmail.com>
- Apply GPIO0 quirk for ASUS G2K laptop
- Sort ASUS ids accordingly.
Submitted by: jkim
MFC after: 3 days
it's multi DAC / playback channels is not that good. Enabling vchans
make the bug more visible since playback allocation will look for
possible free hardware channels first (i.e: the next DAC, the very first
has been consumed by vchan mixer) which in this case has been proven faulty.
Tested by: Dominic Fandrey <LoN_Kamikaze at gmx dot de>
URL: http://lists.freebsd.org/pipermail/freebsd-stable/2007-December/039022.html
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
- Enable pcbeep control for Acer + ALC268 (nid 29). Give enough (fake)
hints so the parser will grab it and allocate "speaker" control.
- Fix regression while preparing DAC and ADC for multichannel
format. Since playback policy is to output to every possible path,
ensure that each DAC is started.
Reported / Tested by: Guy Brand
o Acer Aspire 4520 laptop
- jack sensing / automute
o Toshiba Satellite A135-S4527 laptop
- jack sensing / automute
Tested by: lioux
o Apple Macbook 3 (is it?)
- require gpio0 (for speakers) and ovref50 (for headphone)
to make it works
- jack sensing / automute
Tested by: Ed Schouten
* Add Nvidia MCP67 controller ids.
* Be sensible about simmilar controller with multiple pci ids.
* Connect unused DAC/ADC to stream#0 rather than forcing each of them
managing their own stream.
MFC after: 3 days
The reliability of it's multi DAC / playback channels is
not that good. Enabling vchans make the bug more visible
since playback allocation will look for possible free
hardware channels first (i.e: the next DAC, the very first
has been consumed by vchan mixer) which in this case has
been proven faulty.
Reported / Tested by: Sascha Klauder
MFC after: 3 days
codecs. Codec at address 0 seems purely digital, or perhaps an HDMI
interface. Let the driver skip it and continue scanning the codecs
starting with address 2 (Realtek ALC885).
* Due to possibilities of future similar cases, put enough logic
in hdac_scan_codecs() to force codec scanning starting from
XX address via tunable "hint.pcm.%d.codec_index".
Reported / Tested by: Toomas Pelberg <toomasp@gmx.net>
- Trivial headphone / speaker automute fixup for Fujitsu-Siemens
AMILO Si 1848 laptop.
Reported / Tested by: Ed <ed@bsd.it>
- Trivial headphone / speaker automute fixup for Fujitsu-Siemens
Lifebook S7020D laptop.
Reported / Tested by: Jaromir Dvoracek <jarek@ataxo.com>
- Some smart vendor trying to create interplanetary wormhole by
screwing pci config space during their BIOS update. The side effects
of their failure attempt includes mutilated hardware id, broken
speaker automuting and loosing the entire analog CD connectivity,
thus causing enough collateral damages to collapse the entire
universe. Move along with it.
Please exercise extra cautious when applying BIOS updates.
Reported / Tested by: Pietro Cerutti <gahr@gahr.ch>
- assembled laptop, based on the MSI-1034
(662) which is now becoming MSI-034A.
- Fix no sound issues (on headphones) for Lenovo ThinkCentre A55 due
to global automute table entry which is not applicable for
non-laptops.
Reported / Tested by: Piotr Smyrak <piotr.smyrak@heron.pl>
- Speaker mute control for HP DC7700 since the front headphone jack
does not generate any interesting unsolicited signal/response.
Reported / Tested by: tyop @ irc.freenode.net
Approved by: re (kensmith)
MFC after: 3 days
other changes too).
(without any real order)
1. Use device_get_nameunit for mutex naming
2. Add timer for low-latency playback
3. Move most mixer controls from sysctls to mixer(8) controls.
This is a largest part of this patch.
4. Add analog/digital switch (as a temporary sysctl)
5. Get back support for low-bitrate playback (with help of (2))
6. Change locking for exclusive I/O. Writing to non-PTR register
is almost safe and does not need to be ordered with PTR operations.
7. Disable MIDI until we get it to detach properly and fix memory
managment problems.
8. Enable multichannel playback by default. It is as stable as
single-channel mode. Multichannel recording is still an
experimental feature.
9. Multichannel options can be changed by loader tunables.
10. Add a way to disable card from a loader tunable.
11. Add new PCI IDs.
12. Debugger settings are loader tunables now.
14. Remove some unused variables.
15. Mark pcm sub-devices MPSAFE.
16. Partially revert (bus_setup_intr -> snd_setup_intr) since it need
to be done independently
Submitted by: Yuriy Tsibizov (driver maintainer)
Approved by: re (bmah)
- Add controller id for Intel 82801I (ICH9).
PR: kern/114399
Submitted by: Michael Fuckner <michael@fuckner.net>
- MSI support. Disable by default due to various issues with too many
broken hardwares. MSI can be enabled through device.hints(5) or
kenv(8) by setting "hint.pcm.%d.msi=1".
Partially submitted by: kevlo
YAMAMOTO Taku <taku@tackymt.homeip.net>
Tested by: joel, kevlo, YAMAMOTO Taku
Approved by: re (hrs)
MFC after: 3 days
Submitted by: Simon Schubert <corecode@fs.ei.tum.de>
- Defer flushing unsolicited response into taskqueue thread rather
than handle it directly in interrupt handler, since few of its
operations (like measuring/calibrating jack impedance) are quite
expensive.
- Misc. debugging cleanups.
Tested by: joel
Approved by: re (hrs)
MFC after: 3 days
eradication in/from userland path, countless locking fixes, etc.
- General sleep call through msleep(9) has been converted to condvar(9)
with better consistencies.
- Heavily guard every possible "slow path" entries (open(), close(),
few ioctl()s, sysctls), but once it entering "fast path" (io, interrupt
started), they are free to fly on their own.
- Rearrange locking sequences, resulting better concurrency and
serialization. Large part doesn't even need locking at all, and will be
removed in future. Less clutter, except in few places due to lock
ordering.
- Anonymous mixer object creation/deletion to simplify mixer handling
beyond typical mixer ioctls.
Submitted by: chibis (with modifications)
- Add few mix_[get|set|..] functions to avoid calling mixer_ioctl()
directly using cryptic arguments.
- Locking fixes to avoid possible deadlock with (still under Giant) USB.
- Better simplex/duplex device handling.
- Recover mmap() functionality for recording, which has been lost
since 2.2.x - 3.x (the introduction of newpcm). Full-duplex mmap still
doesn't work (due to VM/page design), but people still can mmap
both by opening each direction separately. mmaped playback is guarantee
to work either way.
- New sysctl: "hw.snd.compat_linux_mmap" to allow PROT_EXEC page
mapping, due to recent changes in linux compatibility layer which
require it. All linux applications that using sound + mmap() (mostly games)
require this to be enabled. Disabled by default.
- Other goodies.. too many, that will increase releng7 shareholder value
and make users of releng6 (and below) cry ;)
* This commit should be atomic. If anything goes wrong (not counting problem
originated from elsewhere), I will not hesitate to revert everything back
within 12 hours. This substantial changes itself not a rocket science
and the process has begun for almost 2 years, and lots of incremental
changes are already in place during that period of time.
* Some issues does occur in snd_emu10kx (note the 'x') due to various
internal locking issues and it is currently being worked on by chibis.
Tested by: chibis (Yuriy Tsibizov), joel, Alexandre Vieira,
many innocent souls...
- Add codec id for AD1988B, along with fixing its line-in and other
issues (with proper quirks). [2]
Submitted by: [1] barbara.xxx1975@libero.it
[2] Oliver Brandmueller ob@e-Gitt.NET
MFC after: 3 days
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.
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.
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
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.
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
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
---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>
- Playback and headphone/speaker automute works.
- Recording untested due to me being deaf doing back-and-forth
remote debugging.
Free Macbook donation is highly appreciated :)
Tested by: Dennis Pielken <mips128@gmx.net>
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.
- 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)
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>
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
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
- [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
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
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
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
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
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)
- 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]
In case this causes trouble for some other chipsets add a comment how to
proceed. If we don't get bugreports, this should be removed after a while
(some releases?).
PR: 56617 [1], 29465, 39260, 40574, 68225
Submitted by: Matthew E. Gove <mgove@comcast.net> [1]
* Add kernel hint option to disable DXS channels entirely. Report
from several skype users / Pav Lucistnik indicate that disabling
DXS may fix lots of pop / crackling noise. To disable DXS add
hint.pcm.<unit>.via_dxs_disabled="1" to /boot/device.hints.
Further investigation of the issues regarding DXS showed, that
the problem is in another (more generic) place, but until the
right fix is tested/reviewed this may help a little bit.
Added sysctl's to aid testing/debugging:
hint.pcm.<unit>.via_dxs_disabled=X - Disable / Enable DXS channels entirely
hint.pcm.<unit>.via_dxs_channels=X - Limit DXS channels up to X
hint.pcm.<unit>.via_sgd_channels=X - Limit SGD channels up to X
hint.pcm.<unit>.via_dxs_src=X - Enable / Disable DXS sample rate
converter.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
especially for CT4730 / EV1938 chip, causing misconfigured mixer
(David Xu), crippled after power cycle (Kevin Oberman). Fixed.
* Incorporate locking/spdif patches from Jon Noack / matk. Not all
es137x can really do spdif, clean it up a bit to only let few capable
chip. This adds a "hw.snd.pcm<unit>.spdif_enabled" sysctl until
a more generic way of handling this from userland (by an ordinary
user) is designed/implemented.
* Convert all bus_space_(read|write) to use es_rd/es_wr, simmilar
with other drivers.
* Add tunable hw.snd.pcm<unit>.latency_timer sysctl to toggle pci
latency timer value on the fly. Much noise / pop / crackling
issues can be solved by increasing its value. Other people have
pointed out to use pciconf instead, but this is just an added
value specific for CT4730/EV1938.
* Remove es137x specific debug sysctl/code.
Several PRs can now be closed.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Submitted by: Jon Noack <noackjr@alumni.rice.edu> (implicit)
Submitted by: matk (implicit)
PR: 59349, 68594, 73498
Tested by: multimedia@
handle DMA addresses located above 1GB. The LBA(loop begin address)
register which holds DMA base address is 32bits register. But the
MSB 2bits are used for other purposes. This effectivly limits the
DMA address space up to 1GB.
Approved by: jake (mentor)
Reviewed by: truckman, matk
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
allocation. Notably, in this case, the driver tries to allocate several
pieces of memory and then fails if the pieces allocated after the first
do not come after it physically, and within a specific range (8MB I
believe). Of course, this could just as easily fail for any number of
reasons, but it almost always fails now that contiguous allocations start
at the end of possible specified memory locations rather than the beginning.
Allocate all the possibly-needed memory up front, even though it's a waste,
to get around this. The least bogus solution would be to take the physical
address from the first allocation and create a new tag that specified that
further allocations must follow it within that 8MB window, then use that
when allocating new channels, but that's left for anyone else that really
feels like doing it.
Tested by: Erwin Lansing <erwin@lansing.dk>
- `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
exactly as done in the cmi driver. I am quite confident this is
safe since I'm runing this for more than two weeks now, on an SMP
box. A few people tested this patch for me successfully as well.
of you with other cards, please do review and test the drivers for
MP-safety and disable Giant in the interrupt routines when you are
sure of proper functionality.
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)
This change has not been tested.
This change was triggered by a gcc(1) warning on ia64 at -O2. The
variable v was not used after being computed, which resulted in enough
dead code elimination (DCE) to confuse the compiler and emit a bogus
warning about the use of the variable i without prior definition. The
variable i is the loop variable.
Submitted by: des
Responsibility: marcel
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@
I started with a year-old patch by Orlando Bassotto
<orlando.bassotto@ieo-research.it>, and ported it to 5.2-CURRENT along with
fixing the problems working with pre-Audigy cards.
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
round the result up to a multiple of 4 bytes so that it will always
be a multiple of the sample size. Also use the actual buffer size
from sc->bufsz instead of the default DS1_BUFFSIZE.
This fixes panics and bad distortion I have seen on Yamaha DS-1
hardware, mainly when playing certain Real Audio media.
Reviewed by: orion (an earlier version of the patch)
tested for playback.
* modify device name strings for ich chips to better conform with their
common names.
* remove superflous 'AC97 controller' from nforce device names.
MFC after: 1 week
Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma. At the moment, this is used for the
asynchronous busdma_swi and callback mechanism. Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg. dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create(). The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.
sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms. The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.
If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.
Reviewed by: tmm, gibbs
codec during initialization. This code mirrors the reset code used on
the VIA82c686 and fixes a codec initialization failure (SoundMAX
AD1885) reported by Matthias Schuendehuette.
actual address of buffer descriptor. This should fix the reported
calibration failures and subsequent speed problems with ich chipsets.
Minor calibration comment updates.
and be prepared to wait much longer for codec to become ready.
Credit to Hugo D. Valentim <hvalentim@gmx.net> for reporting the
problem, providing useful pointers, and repeated diff testing.
channel and disable DXS3. Several users have reported DXS3 as playing
at half speed on the 8233A revision of the chipset. This implicitly
means no SPDIF for VIA8233A users.
o Make DXS3 the primary playback channel. It may be the only
universally supported channel with the assorted revisions of this
chipset.
o Add sysctl and handler for enabling s/pdif output from DXS3.
