infrastructure, not us. This appears to be a leftover from an older
version of the driver.
Submitted by: avg
Tested by: Anton Shterenlikht <mexas bristol.ac.uk>
MFC after: 1 week
X-MFC-Note: To stable/8 and stable/7 only
volumes were incorrectly calculated.
I've tested this with one of my es1370 cards and I can confirm that it
works.
PR: 98167
Submitted by: Joseph Terner <jtsn@gmx.de>
Approved by: kib
loopback.
- Change the meaning of "mix" OSS control. Now it controls loopback level,
according to comments in soundcard.h.
- Allow AD1981HD codecs to use playback mixer. Now driver should be able to
really use it.
- Fix bug in shared muters operation.
now due to unidentified synchonization problem. For 7.1 soundcards 5.1
support handled correctly via software upmix done by sound(4).
Stereo stream is no more duplicated to all ports. If you loose sound, check
you are using right connectors. Front speakers connector is usually green,
center/LFE - orange, rear - black, side - gray.
30-bit like the reset of the controllers supported by this driver.
Actually ALi M5451 can be setup up to generate 32-bit addresses by
setting the 31st bit via the accompanying ISA bridge, which allows
it to work in sparc64 machines whose IOMMU require at least 32-bit
DMA. Even though other architectures would also benefit from 32-bit
DMA, enabling this bit is limited to sparc64 as bus_dma(9) doesn't
generally guarantee that a low address of BUS_SPACE_MAXADDR_32BIT
results in a buffer in the 32-bit range.
- According to Tatsuo YOKOGAWA's ali(4), the the DMA transfer size of
ALi M5451 is fixed to 64k and in fact using the default size of 4k
- The 4DWAVE DX and NX require the recording buffer to be 8-byte
aligned so adjust the bus_dma_tag_create(9) accordingly.
- Unlike the rest of the controllers supported by this driver, the
ALi M5451 only has 32 hardware channels instead of 64 so limit the
loop in tr_intr() accordingly. [1]
Submitted by: yongari [1]
Reviewed by: yongari (superset of what is committed)
MFC after: 3 days
The newbus lock is responsible for protecting newbus internIal structures,
device states and devclass flags. It is necessary to hold it when all
such datas are accessed. For the other operations, softc locking should
ensure enough protection to avoid races.
Newbus lock is automatically held when virtual operations on the device
and bus are invoked when loading the driver or when the suspend/resume
take place. For other 'spourious' operations trying to access/modify
the newbus topology, newbus lock needs to be automatically acquired and
dropped.
For the moment Giant is also acquired in some key point (modules subsystem)
in order to avoid problems before the 8.0 release as module handlers could
make assumptions about it. This Giant locking should go just after
the release happens.
Please keep in mind that the public interface can be expanded in order
to provide more support, if there are really necessities at some point
and also some bugs could arise as long as the patch needs a bit of
further testing.
Bump __FreeBSD_version in order to reflect the newbus lock introduction.
Reviewed by: ed, hps, jhb, imp, mav, scottl
No answer by: ariff, thompsa, yongari
Tested by: pho,
G. Trematerra <giovanni dot trematerra at gmail dot com>,
Brandon Gooch <jamesbrandongooch at gmail dot com>
Sponsored by: Yahoo! Incorporated
Approved by: re (ksmith)
- honor parent DMA tag limitations, as man page requires,
- allow data buffer to be allocated within full 64bit address range, when
support is announced by hardware,
- add quirk, disabling 64bit addresses for broken chips, use it for MCP78.
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.
mic inputs. I have no idea what for it was made that time, but now I have
several reports that it should be removed to make microphones work. If
this quirk is still required for some systems then they should be identified
and specified explicitly.
only for mic-type inputs. This gives better chances to use it.
Change default configuration for some AD1986A codec based ASUS boards,
use it also for ASUS P5PL2 board. This makes front mic preamplifier working.
Tested by: Vadim Frolov <frolov@frolov.ck.ua>
implement CD input in hardware, while unconditional showing it confuse users.
Also it was made in the way that sometimes improper with present driver.
Add patch for ALC268 based Acer TM5320 to make headphones jack sensing work.
Default configuration defines two separate playback associations, which
current driver unable to trace properly due to order they are defined and
limited codec uniformity.
Submitted by: G. Mirov <g.mirov AT gmail.com>
Disable MSI for nVidia MCP51 controller. Enabling MSI there leads to
unexpected errors and timeouts, that should not happen even if interrupts
are not working completely.
use patches so far:
+ Envy24:
- fix: broken init data for M Audio Delta DiO 2496
- add: support for M Audio Delta 44
- add: support for M Audio Delta 1010LT
Tested by: Dominique Goncalves, dominique.goncalves at gmail.com
- add: support for Terratec EWX 2496
Tested by: Stefan Sperling, stsp at stsp.name
- add: support for M Audio Delta 66
Tested by: Richard Bown, richard.bown at blueyonder.co.uk
- add: support for M Audio Delta 1010
Tested by: Andrew Reilly, areilly at bigpond.net.au
+ Envy24HT:
- add: support for Terrasoniq TS22PCI
- fix: M-Audio Revolution 5.1 sound volume is very low
Reported by: Oliver Hartmann, ohartman at zedat.fu-berlin.de
Andrey Slusar, anrays at gmail.com
Tested by: Andrey Slusar, anrays at gmail.com
Rusu Silviu, arol.the at gmail.com
- fix: M-Audio Revolution 7.1 sound is distorted and very quiet
Reported by: Olev Hannula, hannula at gmail.com
Tested by: Olev Hannula, hannula at gmail.com
Stanislav Belansky, stanislav at icmail.ru
- fix: Terratec PHASE 22 codec is power-off due to wrong init data
Reported by: Philipp Ost, pj at smo.de
Tested by: Philipp Ost, pj at smo.de
+ SpicDS:
- fix: AK4381 produce hiss sound on 192kHz sample rate
- fix: stupid bug with volume control for AK4396
Submitted by: Konstantin Dimitrov <kosio.dimitrov@gmail.com>
module. These files cause manual interaction when building
ports/audio/aureal-kmod which provides a usable i386-only driver (it requires
linking against some linux object files distributed by vendor which bankrupted
back in 2000).
MFC after: 1 week
Disable some unneeded pathes in overcomplicated input mixer to help parser
to handle the rest better. This gives mic input boost control in some
configurations and just more predictable operation in others.
nodes capabilities. Add "Analog"/"Digital" marks to the pcm device names.
I hope it will help new users easier accept concept of several PCM devices
and understand exact purposes of that devices.
with several points unappropriate for the present parser. This patch
disables input-to-output analog monitoring but instead fixes recording.
Tested by Tobias Grosser on ThinkPad T61p.
When I changed kern_conf.c three months ago I made device unit numbers
equal to (unneeded) device minor numbers. We used to require
bitshifting, because there were eight bits in the middle that were
reserved for a device major number. Not very long after I turned
dev2unit(), minor(), unit2minor() and minor2unit() into macro's.
The unit2minor() and minor2unit() macro's were no-ops.
We'd better not remove these four macro's from the kernel, because there
is a lot of (external) code that may still depend on them. For now it's
harmless to remove all invocations of unit2minor() and minor2unit().
Reviewed by: kib
Left only parts surely required for basic troubleshooting and configuration
purposes. There is still very long output, but further shrinking makes it
less informative.
Original debugging can be enabled with hw.snd.verbose=4.
Because of using more clear and same time more functional codec parser
new driver is able to handle more codecs, use them better then before and
without most of previous quirks. All of tested codecs itself manage playback,
record, input mixing and monitoring quite fine. In all of investigated
trouble cases problem was found or in nonstandard codec usage or incorrect
codec configuration made by BIOS. Most of that cases could be fixed using
device hints, some of which are already included to the driver.
New driver supports multiple codecs per HDA bus, multiple audio function
groups per codec and multiple logical sound devices per audio function group.
So don't worry when you get several PCM devices instead of one, it is normal.
It is usual situation with powerful codecs to provide, for example, 3 PCM
devices: one for 7.1 playback and main recording, one for headset and one
for digital SPDIF I/O.
New driver implements Universal Audio Architecture (UAA) much better then
previous one. Most information about recommended codec usage now taken from
the codec configuration registers initialized by BIOS. User may alter that
configuration using device hints to reconfigure logical audio devices to
his needs in a very broad range up to the limits of the codec functionality.
New driver supports digital PCM playback and AC3 pass-through. I am not sure
about completeness of this implementation, but I have several success stories
including my own. Vchans subsystem does not support AC3 pass-through so it
had to be disabled for that devices at this moment.
New driver is ready for multichannel playback, but until our OSS is unable
to use this it will just duplicate same stereo stream into all channel
pairs.
New driver supports suspend/resume. I am unable to really test this part
myself, but I have got several success stories.
Driver has very informative verbose boot messages. So if you have any
questions or problems - enable and read them first.
Discussed on: freebsd-multimedia@
Tested by: many
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
