sizeof(codec->reg_cache) is just the size of the pointer. Elsewhere in the
file, codec->reg_cache is used with sizeof(wm8900_reg_defaults), so the
code is changed to do the same here.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression *x;
expression f;
type T;
@@
*f(...,(T)x,...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert table of pointers to mixer controls into tables
of the mixer controls. It saves about 20% of the snd-sb-common
module size reported by lsmod.
The als4000 uses part of sb16's control table.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sizeof(share_id) is just the size of the pointer. On the other hand,
block->share_id is an array, so its size seems more appropriate.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression *x;
expression f;
type T;
@@
*f(...,(T)x,...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/461062
The original reporter states that PCM maxes at +12 dB and results in
very bad distortion. Cap PCM at 0 dB to resolve this symptom.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/418627
The original reporter states that this quirk is necessary to obtain
reasonable gain for playback. Without it, sound is inaudible. Tested
with playback (spkr and hp) and capture.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames function names like twl4030_i2c_write_u8,
twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8
and also common variable in twl-core.c
Signed-off-by: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
The upcoming TWL6030 is companion chip for OMAP4 like the current TWL4030
for OMAP3. The common modules like RTC, Regulator creates opportunity
to re-use the most of the code from twl4030.
This patch renames few common drivers twl4030* files to twl* to enable
the code re-use.
Signed-off-by: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Bring the WM8350 IRQ API more in line with the generic IRQ API by
masking and unmasking interrupts as they are requested and freed.
This is mostly just a case of deleting the mask and unmask calls
from the individual drivers.
The RTC driver is changed to mask the periodic IRQ after requesting
it rather than only unmasking the alarm IRQ. If the periodic IRQ
fires in the period where it is reqested then there will be a
spurious notification but there should be no serious consequences
from this.
The CODEC drive is changed to explicitly disable headphone jack
detection prior to requesting the IRQs. This will avoid the IRQ
firing with no jack set up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
This is done as simple code transformation, the semantics of the
IRQ API provided by the core are are still very different to those
of genirq (mainly with regard to masking).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Overwrite pin config on intel DG45ID board.
intelhdmi - dont power off HDA link
ALSA: hrtimer - Fix lock-up
ALSA: intelhdmi - add channel mapping for typical configurations
ALSA: intelhdmi - channel mapping applies to Pin
ALSA: intelhdmi - accept DisplayPort pin
ALSA: hda - show HBR(High Bit Rate) pin cap in procfs
ALSA: hda - Fix LED GPIO setup for HP laptops with IDT codecs
ASoC: Fix build of OMAP sound drivers
ALSA: opti93x: fix irq releasing if the irq cannot be allocated
Add dB scale for mixer controls. Fix dB scale for
Master Volume control.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin config provided by BIOS have some problems:
0x0221401f: [Jack] HP Out at Ext Front <-- other association and sequence
0x02a19020: [Jack] Mic at Ext Front <-- other association
0x01113014: [Jack] Speaker at Ext Rear <-- line out (not speaker)
0x01114010: [Jack] Speaker at Ext Rear <-- line out
0x01a19030: [Jack] Mic at Ext Rear <-- other association
0x01111012: [Jack] Speaker at Ext Rear <-- line out
0x01116011: [Jack] Speaker at Ext Rear <-- line out
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x40f000f0: [N/A] Other at Ext N/A
0x01451140: [Jack] SPDIF Out at Ext Rear
0x40f000f0: [N/A] Other at Ext N/A
just overwrite it.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move OPTi93x controls definitions to the opti93x driver
from the common wss-lib library module. These controls
are used only by the opti93x driver.
Also, fix capture source names. They are the same as
opl3sa2 names.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The als100 driver is so similar to the dt019x/als007 driver
that one driver's source can be used for both drivers with
only few changes. Merge the dt019x driver into the als100.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For codecs without EPSS support (G45/IbexPeak), the hotplug event will
be lost if the HDA is powered off during the time. After that the pin
presence detection verb returns inaccurate info.
So always power-on HDA link for !EPSS codecs.
KarL offers the fact and Takashi recommends to flag hda_bus. Thanks!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer stop callback can be called from snd_timer_interrupt(), which
is called from the hrtimer callback. Since hrtimer_cancel() waits for
the callback completion, this eventually results in a lock-up.
This patch fixes the problem by just toggling a flag at stop callback
and call hrtimer_cancel() later.
Reported-and-tested-by: Wojtek Zabolotny <W.Zabolotny@elka.pw.edu.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IbexPeak is the first Intel HDMI audio codec to support channel mapping.
Currently the outstanding problem is, the HDMI channel order do not
agree with that of ALSA. This patch presents workaround for some
typical use cases. It gives priority to the typical ALSA surround
configurations, and defines channel mapping for them.
We may need better kernel+userspace interactive channel mapping scheme.
For example, in current scheme if user plays with the surround50 device,
the kernel is unaware of this and will still select the surround41
channel allocation and channel mapping..
Thanks to Marcin for offering good tips!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA036-A specifies that the Audio Sample Packet (ASP) Channel Mapping
verbs apply to Digital Display Pin Complex instead of Converter.
With this fix, channel mapping is working as expected for IbexPeak.
Thanks to Marcin for pointing this out!
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA036 spec states:
DP (Display Port) indicates whether the Pin Complex Widget supports
connection to a Display Port sink. Supported if set to 1. Note that
it is possible for the pin widget to support more than one digital
display connection type, e.g. HDMI and DP bit are both set to 1.
Also export the DP pin cap in procfs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Note that the HBR capability only applies to HDMI pin.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes an error in processing of the HP BIOS configuration to enable
GPIO based mute LED indicator control. That error causes driver to enable
such control on all HP systems with the 92HD75 IDT codecs and results in
unnecessary toggling of the GPIO on mute control manipulation.
It also adds support of the future HP BIOS configuration extension for the
named control. New configuration string has a format HP_Mute_LED_P_G
where P can be 0 or 1 and defines mute LED GPIO control state (low/high)
that corresponds to the NOT muted state of the master volume
and G is the index of the GPIO to use (0..9)
Lastly, it adds more systems to the support of the audio implementation
as found on HP B-series systems
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are build errors when building for some of the omap2/3 boards without
enabling sound:
sound/built-in.o:(.data+0x43bc): undefined reference to `soc_codec_dev_tlv320aic23'
sound/built-in.o:(.data+0x43cc): undefined reference to `tlv320aic23_dai'
Confused me quite a bit since the drivers that had references to the
codec weren't enabled. Turns out the Makefile was using the wrong
config option to enable them. Patch below.
Reported-by: Anand Gadiyar <gadiyar@ti.com>
Signed-off-by: Olof Johansson <olof@lixom.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While Linux provided an O_SYNC flag basically since day 1, it took until
Linux 2.4.0-test12pre2 to actually get it implemented for filesystems,
since that day we had generic_osync_around with only minor changes and the
great "For now, when the user asks for O_SYNC, we'll actually give
O_DSYNC" comment. This patch intends to actually give us real O_SYNC
semantics in addition to the O_DSYNC semantics. After Jan's O_SYNC
patches which are required before this patch it's actually surprisingly
simple, we just need to figure out when to set the datasync flag to
vfs_fsync_range and when not.
This patch renames the existing O_SYNC flag to O_DSYNC while keeping it's
numerical value to keep binary compatibility, and adds a new real O_SYNC
flag. To guarantee backwards compatiblity it is defined as expanding to
both the O_DSYNC and the new additional binary flag (__O_SYNC) to make
sure we are backwards-compatible when compiled against the new headers.
This also means that all places that don't care about the differences can
just check O_DSYNC and get the right behaviour for O_SYNC, too - only
places that actuall care need to check __O_SYNC in addition. Drivers and
network filesystems have been updated in a fail safe way to always do the
full sync magic if O_DSYNC is set. The few places setting O_SYNC for
lower layers are kept that way for now to stay failsafe.
We enforce that O_DSYNC is set when __O_SYNC is set early in the open path
to make sure we always get these sane options.
Note that parisc really screwed up their headers as they already define a
O_DSYNC that has always been a no-op. We try to repair it by using it for
the new O_DSYNC and redefinining O_SYNC to send both the traditional
O_SYNC numerical value _and_ the O_DSYNC one.
Cc: Richard Henderson <rth@twiddle.net>
Cc: Ivan Kokshaysky <ink@jurassic.park.msu.ru>
Cc: Grant Grundler <grundler@parisc-linux.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: "H. Peter Anvin" <hpa@zytor.com>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Andreas Dilger <adilger@sun.com>
Acked-by: Trond Myklebust <Trond.Myklebust@netapp.com>
Acked-by: Kyle McMartin <kyle@mcmartin.ca>
Acked-by: Ulrich Drepper <drepper@redhat.com>
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Jan Kara <jack@suse.cz>
Use the chip->irq to check if the irq should be released so the irq is not released
if it has not been allocated.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The volume levels in original implementation are incorrect and does
not match the dB scale. The real range is linear (in the sense of
the dB scale) from 0dB to -100dB. Remove logaritmic table and make
all volumes from range 0dB..100dB.
The tests are in RedHat's bugzilla #540817.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quirk for the ALC662 found on the Intel D945GCLF2 (and possibly other)
mainboards.
Signed-off-by: David Santinoli <david@santinoli.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Confirmed from vendor and tests in RedHat bugzilla #536782 .
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wrong variable was returned in the case of an error
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I added the product IDs of the new revisions of the devices, so owners
can test whether this suffices to make them work. Patched against ALSA
snapshot 20091207.
Signed-off-by: Tobias Hansen <Tobias.Hansen at physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Realtek codecs, a digital mic pin is connected often only to a single
ADC. But the parser tries to set up all ADCs no matter whether the
digital mic is available, and results in non-selectable input source.
This patch adds a check of input-source availability of each ADC, and
excludes ones that don't support all input sources.
Reference: Novell bnc#561235
http://bugzilla.novell.com/show_bug.cgi?id=561235
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an updated patch for the Apple iMac 9,1 model to add sound.
Original patch posted here:
http://article.gmane.org/gmane.linux.alsa.devel/61361/match=
I have been using this patch for a while now
and have to say it works vary well, except for a few minor
things:
With the iMac 24-inch 3.06GHz Intel Core 2 Duo
everything seems to be working as it should,
although I have not looked into the microphone
(never really use one, nor have any apps to test,
my guess is it doesn't work, or I never figured out how
to get it to work).
With the iMac 24-inch 2.66GHz Intel Core 2 Duo
everything is the same as with the above machine
except I'm hearing a light scratchy/distortion noise
come out of the speakers when using headphones(above machine
does not do this).
Other than that the sound level is great(especially with good Dj headphones).
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Tested-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PnP data on the OPTI931 and OPTI933 contains io port
range for the MC indirect registers. Use the PnP range
instead of hardwired value 0xE0E.
Also, request region of MC indirect registers so it is
marked as used to other drivers (this was missing previously).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Komuro pointed out that IRQ_FIRST_SHARED is not used at all in the
PCMCIA subsystem, so remove it. Also, remove two bogus assignments.
CC: Karsten Keil <keil@b1-systems.de>
CC: netdev@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: Komuro <komurojun-mbn@nifty.com>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8904 is a high performance ultra-low power stereo CODEC
optimised for portable audio applications, with features including
a class W amplifier, FLL with free running mode, Mobile ReTune and
ground referenced headphone and line outputs.
Support for some features, most particularly the digital microphone
interface, is not yet present.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make it a bit easier to tie DAPM widgets in with the register map
without referring to the source by including the register location
controlled by the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Take the regulator framework in use for managing the power sources.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add clock enable and disable calls to resume and suspend respectively.
Also add a member to the audio device data structure which tracks the clock
status.
Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which
add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied.
[1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/
2009-November/016958.html
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This renames from a character / to : of controls. A / occurs below error
messages.
ASoC: Failed to create IN2RP/VXRP debugfs file
ASoC: Failed to create IN2LP/VXRN debugfs file
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When primary AC97 is not found, don't fail with tons of AC97 errors.
Assume that the card is SF64-PCR (tuner-only).
This makes the SF64-PCR radio card work "out of the box".
Also fixes a bug that can cause an oops here:
if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
when tea575x_tuner == 16, it passes this check and causes problems
a couple lines below:
chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
Tested with SF64-PCR, but I don't have any of those sound or sound+radio cards
to test if I didn't break anything.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix mute state reporting in tea575x-tuner.
This fixes mute function in kradio on SF64-PCR radio card.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver use global variable to access device data.
But this style will be broken
if SuperH come with multiple FSI blocks in future.
To solve this problem, this patch use cpu_dai->private_data.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
free the allocated pcm platform device in the error path.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
platform_device_unregister() frees resources for us, no need to
do it explicitly. Fixes an oops when machine code removes the
soc-audio device.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove snd_opti9xx fields which are indirect arguments to
the snd_opti9xx_configure(). Pass these values as function
arguments.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_ac97_pcm_open() is called with the exactly same arguments
for both playback and capture directions. Remove the unneeded check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Muse Pocket use brocken mixer names, so alsamixer and PA can't use it correctly
This patch add quirk to overwirte default mixers.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FSC Amilo Pi 1505 has a buggy BIOS and doesn't set up the HP and
speaker pins properly. Add the pinfix entry for that.
Reference: Novell bnc#557403
https://bugzilla.novell.com/show_bug.cgi?id=557403
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pcm->r[1].slots is the double rate slot information, not the
capture information. For capture, 'pcm' will already be the
capture ac97 pcm structure.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA's for-2.6.33 branch has a new source argument to
snd_soc_dai_set_pll().
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support runtime PM.
Driver callbacks for Runtime PM are empty because
the device registers are always re-initialized after
pm_runtime_get_sync(). The Runtime PM functions replaces the
clock framework module stop bit handling in this driver.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM12 and PCM20 can be set into the ISA PnP mode. The PCM12 PnP
was sold as the PnP device.
Add code to handle detection of these cards using ISA PnP framework.
Tested on the PCM20 in PnP mode. The PCM12 PnP has the same MS Windows
INF file except for a card name displayed for user.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Separate common probing code in order to use it
for PnP probing.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of the irq_req_t typedef'd struct can be re-worked quite
easily:
(1) IRQInfo2 was unused in any case, so drop it.
(2) IRQInfo1 was used write-only, so drop it.
(3) Instance (private data to be passed to the IRQ handler):
Most PCMCIA drivers using pcmcia_request_irq() to actually
register an IRQ handler set the "dev_id" to the same pointer
as the "priv" pointer in struct pcmcia_device. Modify the two
exceptions (ipwireless, ibmtr_cs) to also work this waym and
set the IRQ handler's "dev_id" to p_dev->priv unconditionally.
(4) Handler is to be of type irq_handler_t.
(5) Handler != NULL already tells whether an IRQ handler is present.
Therefore, we do not need the IRQ_HANDLER_PRESENT flag in
irq_req_t.Attributes.
CC: netdev@vger.kernel.org
CC: linux-bluetooth@vger.kernel.org
CC: linux-ide@vger.kernel.org
CC: linux-wireless@vger.kernel.org
CC: linux-scsi@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: Jaroslav Kysela <perex@perex.cz>
CC: Jiri Kosina <jkosina@suse.cz>
CC: Karsten Keil <isdn@linux-pingi.de>
for the Bluetooth parts: Acked-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes two issues:
a) Infinite loop in resume function
b) Writes to non-existing registers in resume function
Cc: stable@kernel.org
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit fe3e78e073
ASoC: Factor out snd_soc_init_card()
removed the error paths that are still valid for wm97* codecs, causing
the compile errors like
sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined
sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined
sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined
Revert the removed error path codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT codecs seem to behave weird when SET_PIN_SENSE verb is issued
before reading the jack-detection although the TRIG_REQ pin capability
is given by the hardware.
Since snd_hda_jack_detect() issues the SET_PIN_SENSE verb simply judging
from the pincap, we have to revert the change in the commit
d56757abc1
ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect()
to plain GET_PIN_SENSE verb without triggering.
Reported-by: Jiri Slaby <jirislaby@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a hardware is not detected there is a kernel crash
due to not initialized snd_miro->aci pointer. This pointer
is initialized after detection of the opti (miro) chip.
This bug was introduced by patches to expose
ACI mikser outside the snd-miro driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The non-cohernet PPC arch doesn't give the correct address by a simple
virt_to_page() for pages allocated via dma_alloc_coherent().
This patch adds a hack to fix the conversion similarly like MIPS.
Note that this doesn't fix perfectly: the pages should be marked with
proper pgprot value. This will be done in a future implementation like
the conversion to dma_mmap_coherent().
Acked-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The non-coherent MIPS arch doesn't give the correct address by a simple
virt_to_page() for pages allocated via dma_alloc_coherent().
Original patch by Wu Zhangjin <wuzj@lemote.com>.
[Ralf mentioned: "The origins of this patch go back far further.
The oldest patch I could find which is a superset of this was written
by Atsushi Nemoto and various incarnations of it have been sumitted
to and reject by me a number of times through the years."]
A proper check of the buffer allocation type was added to avoid the
wrong conversion.
Note that this doesn't fix perfectly: the pages should be marked with
proper pgprot value. This will be done in a future implementation like
the conversion to dma_mmap_coherent().
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
RT workqueue is going away in the near future, replace it with
singlethread wq for now, which is still supported.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a helper (inline) function as the default page ops. Any hacks wrt
the page address conversion will be applied in this function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dma_mmap_coherent() for mmapping the buffers allocated via
dma_alloc_coherent() if available. Currently, only ARM has this function,
so we do temporarily have an ifdef pcm_native.c. This should be handled
better globally in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4792
Cristian reported that these models have really bad sound above 6 dB
and proposed the original patch. I've updated the comment to reflect
this change.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Reported-by: Cristian Klein
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/487884
This Gateway model needs External Amplifier muted for audible playback,
so set the inv_eapd quirk for it.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chip field is no longer needed. Move those of its fields that are
actually used to the device structure itself.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the dependecy from the USB MIDI code on the snd_usb_audio
structure. This allows using the USB MIDI module from another driver
without having to pretend to be the generic USB audio driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to select between the two altsettings on Roland USB
MIDI devices where the input endpoint is either bulk or interrupt.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Vostro 1015n uses Conexant CX20583-10Z (0x14f1:5067). Patch is
based on "olpc-xo-1_5" branch. Dell uses digital mic.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit f2624791a0.
Łukasz Wojniłowicz reported that the change causes both internal and
external mics not working any more. The headphone jacking issue was
fixed by his previous patch, it's better to revert to acer-aspire-4930g
model.
Reported-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we get a stream suspend event force the power down since otherwise
the stream would remain marked as active. In future we'll probably want
to make this stream-specific and add an interface to make the power down
of other widgets optional in order to support leaving bypass paths
active while suspending the processor.
Cc: stable@kernel.org
Reported-by: Joonyoung Shim <jy0922.shim@samsung.com>
Tested-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the buffer size calculation to use the size which ALSA is expecting.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the snd_card pointer from the snd_miro structure and
do some small code improvements.
Also, move Opti chipset detection before detection of the
ACI mixer, so the mci_base value is set in one place only.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_JACK needs to be selected explicitly only when INPUT=y or
INPUT_SND. The current way, INPUT=SND_HDA_INTEL isn't strict enough.
Reported-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use hweight16 instead of Brian Kernighan's/Peter Wegner's method
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use db scale for all volume controls according to Crystal's datasheets.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the quirk for Acer Aspire 5930G from model=acer-aspire-4930g to
model=acer-aspre-6530g. The tuba bass gets muted along with the other
built-in speakers upon headphones insertion, the internal mic works
perfectly etc.
Reported-by: Claudio Viano <claudio.viano@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use edma_pause and edma_resume to make missing dma_events
less likely. This may not be needed, but it looks better.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix underruns by using dma to copy 1st to sram
in a ping/pong buffer style and then copying from
the sram to the ASP. This also has the advantage
of tolerating very long interrupt latency on dma
completion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename variable master_lch to asp_channel
Rename variable slave_lch to asp_link[0]
Rename local variables:
lch to link
count to asp_count
src to asp_src
dst to asp_dst
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the left and right 16 bit samples to be shifted out as 1
32 bit sample.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: tlv320aic23 fix rate selection
ASoC: OMAP3 Pandora: update for TWL4030 codec changes
ASoC: Modifying the license string GPLv2 for OMAP3 EVM
ALSA: hda - Fix quirk for VAIO type G
ALSA: usb - Quirk to disable master volume control in PCM2702
The mute-LED isn't synchronized with the actual mute state on some
HP laptops with IDT 92HD83xxx codecs. A similar hack using
check_power_status callback is added for this codec, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the ordering of sr_valid_mask array.
The lower bit of the index represents USB
not bosr.
Reported-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A while ago TWL4030 had it's playback stream name changed, but
pandora needs it for it's playback path. Update to correct stream
name so that playback works again.
Also mark VIBRA output as not connected.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Correcting the license string from GPLv2 -> GPL v2.
Found the problem while building OMAP3 ASoC driver as
module.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove requirement that dma_params is 1st in the structures
davinci_audio_dev and davinci_mcbsp_dev.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Making room for namespace for the PCM Controller driver
the platform driver(s3c24xx-pcm) has been renamed to SoC
agnostic name 's3c-dma'.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The s3c24xx_pcm prefix for the soc_platform is inappropriate when
some Samsung SoCs have PCM controllers which will eventually have
drivers and hence namespace ambiguities.
To resolve naming ambiguities in future the following have been
renamed in order
1) s3c24xx_pcm_dma_params -> s3c_dma_params
2) s3c24xx_pcm_preallocate_dma_buffer -> s3c_preallocate_dma_buffer
3) s3c24xx_pcm_dmamask -> s3c_dma_mask
4) s3c24xx_pcm_XXX -> s3c_dma_XXX
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dual-headphone mode with STAC/IDT codecs is useful only for machines
that have two (or more) built-in headphones.
But, some HP laptops give multiple headphone pin configs, one for the
built-in and another for the separate (likely a docking station) one.
This results in a missing speaker volume control.
This patch adds more check for the dual-headphone mode to avoid this
problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function is only called from snd_ctl_ioctl() and the file parameter
can never be null so there is no need to check it here.
We dereference file at the start of the function:
struct snd_card *card = file->card;
and it confuses static checkers to dereference a pointer before
checking it.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We tracked down the first-0.5s-hdmi-audio-samples-lost problem to the
AC_VERB_SET_CHANNEL_STREAMID command. It is suspected that many HDMI
sinks need some time to adapt to the new state.
The workaround is to avoid changing stream id/format whenever possible.
Proposed by David.
Signed-off-by: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remember the active infoframe, so as to avoid stop/restart infoframe
transmission when switching between audio clips of the same format.
Proposed by Shang and David.
CC: Shane W <shane-alsa@csy.ca>
CC: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
And make it right when called for more than one times.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This avoids lost of presence info on module reloading.
The presence info used to be only updated at the (rare) hotplug events.
Proposed by David, thanks!
CC: David Härdeman <david@hardeman.nu>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the fm_port and mpu_port variables directly in a probe function.
This completely eliminates a need to copy the fm_port value to
the snd_miro structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.
Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.
Also, delete one misnamed cs4231 register define.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ice1724 - make some bitfields unsigned
ALSA: hda - Dell Studio 1557 hd-audio quirk
ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume()
ALSA: hda - Add another Nvidia HDMI codec id (10de:0005)
ALSA: hda: Use model=mb5 for MacBookPro 5,2
* 'hostprogs-wmissing-prototypes' of git://git.kernel.org/pub/scm/linux/kernel/git/josh/linux-misc:
Makefile: Add -Wmising-prototypes to HOSTCFLAGS
oss: Mark loadhex static in hex2hex.c
dtc: Mark various internal functions static
dtc: Set "noinput" in the lexer to avoid an unused function
drm: radeon: Mark several functions static in mkregtable
arch/sparc/boot/*.c: Mark various internal functions static
arch/powerpc/boot/addRamDisk.c: Mark several internal functions static
arch/alpha/boot/tools/objstrip.c: Mark "usage" static
Documentation/vm/page-types.c: Declare checked_open static
genksyms: Mark is_reserved_word static
kconfig: Mark various internal functions static
kconfig: Make zconf.y work with current bison
The ALC262 has a quirk entry matching with all Sony Vaio laptops
to use model=sony-assamd as default. But, model=auto works much better
for new models in the recent driver versions, thus it's safer to disable
that default quirk entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vaio type G laptop doesn't work with the current quirk setup.
After some tests, it turned out that it should be model=auto as default.
Reported-by: Mattia Dongili <malattia@linux.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the master volume control in the PCM2702 chipset.
The datasheet documents two independent channel volume controls, one
master mute control and one master volume control. All controls are
fully functional except for the master volume control, which returns
USB stalls on all GET requests.
Signed-off-by: Javier Kohen <jkohen@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the {orig,midi}_dev equals num_midis, that's one too
large already.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The beep_mode option value was wrongly defined: it must be 0 = off and
1 = on.
Also, evaluate the beep_mode value at snd_hda_attach_beep_device()
properly so that no device is created when beep_mode=0 is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for dynamically created controls to proc codec file
(Control: lines).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an initial patch to show universal control<->NID assigment in
proc codec file. The change helps to debug codec related problems.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep_mode parameter for snd-hda-intel module allows to choose among
different digital beep device registation to the input layer.
0 = do not register to the input layer
1 = register to the input layer all time
2 = use "Beep Switch" control exported to user space mixer applications
Also, introduce CONFIG_SND_HDA_INPUT_BEEP_MODE for default value.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unregister work should be also canceled in snd_hda_detach_beep_device()
function.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The massive register/unregister calls for input device layer might be
overkill. Delay unregister call by one HZ as workaround.
Also, as benefit, beep->enabled variable is changed immediately now
(not from workqueue).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original implementation was keeping registered input device for SND_BEEP
and SND_TONE events all time. This patch changes this behaviour:
If digital PC Beep is turned off using universal control switch,
the input device is unregistered.
Explanation: The kd_mksound() send SND_BEEP and SND_TONE only to last
registered device acceping those events. It means that the HDA Intel
audio driver blocks also the internal PC Speaker device (pcspkr.c
driver) even if the HDA Beep is muted. The user can easy disable
all beeps using 'setterm -blength 0' or 'xset b off' command.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a clean up and doesn't change the behavior.
Bit fields should always be unsigned. Otherwise pm_suspend_enabled will
be -1 when you want it to be 1. The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.
The other bitfields in that struct are unsigned already.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper suspend/resume code for Juli@ cards. Based on ice1724
suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
Tested on linux-2.6.31.6
Signed-off-by: Aleksey Kunitskiy <alexey.kv@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_via.c: In function 'via_hp_bind_automute':
sound/pci/hda/patch_via.c:2074: internal compiler error: in do_SUBST, at combine.c:462
Please submit a full bug report,
with preprocessed source if appropriate.
See <URL:http://gcc.gnu.org/bugs.html> for instructions.
[added a comment by tiwai]
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAPM widgets are now insntantiated by the core when creating the card
so there is no need for the individual CODEC drivers to do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While probing, AC97 codec drivers and soc-core generically execute the
following sequence:
snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID
to detect ->... -> set platform_data to ac97 by soc-core
commit 474828a40f adds platform_data to
snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97
before actual ac97 operations. Then while ac97_read access platform_data
of snd_ac97 for detecting, NULL pointer oops will fire. That means old
platform_data patch doesn't work in real-life cases.
This patch moves the operation of setting ac97 platform_data earlier
than ac97 reading/writing operations. Then it makes platform_data of
AC97 become practically useful.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Corrected the order of 'source' and 'pll_id' arguments.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
CONFIG_SND_HDA_POWER_SAVE is independent from CONFIG_SND_HDA_HWDEP.
Thus snd_hda_hwdep_add_power_sysfs() needs the check of both kconfigs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit e330323520
"ALSA: hda - proc - show which I/O NID is associated to PCM device"
introduces the access to substream pointer. But, PCMs may have no
substreams in one or both directions, and this results in NULL
dereference. Also, print the first substream number doesn't make
sense.
This patch removes the access to the substream pointer, and reformat
to fit to the standard coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/462098
Until we can look closer at the verbs, let's use ALC885_MB5 for
codec SSID 0x106b4600 to enable playback and capture for MacBookPro
5,2s.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the power on/off counter and expose via sysfs files.
The sysfs files, power_on_acct and power_off_acct, are created under
each codec hwdep sysfs directory (e.g. /sys/class/sound/hwC0D0).
The files show the msec length of the codec power-on and power-off,
respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test of index `i' is after the read - too late - and
unsafe: if snd_hda_get_connections() fails in the last
iteration a read beyond the array is possible.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a RawMIDI substream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The substream_opened field is to count the number of opened substreams,
not the number of times that any substreams have been opened.
Furthermore, all substreams being opened does not imply that the next
open would fail, due to the possibility of O_APPEND. With this wrong
check, opening a substream multiple times would succeed only if the
device had more unopened substreams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 broke the
error handling code in rawmidi_open_priv().
If only the output substream of a RawMIDI device has been opened and
if this device is then opened with O_RDWR | O_APPEND and if the
initialization of the input substream fails (either because of low
memory or because the device driver's open callback fails), then the
runtime structure of the already open output substream will be freed
and all following writes through the first handle will cause
snd_rawmidi_write() to use the NULL runtime pointer.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 dropped the
check that a substream must already have been opened with O_APPEND to be
able to open it a second time.
This would make it possible for a substream to be switched to append
mode, which would mean that non-atomic writes would fail unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 moved the
substream initialization code to where it would be executed every time
the substream is opened.
This had the consequence that any further opening would drop and leak
the data in the existing buffer, and that the device driver's open
callback would be called multiple times, unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The present quirk for HP dc5750 seems broken and maps the pins wrongly.
Since the auto-parser works well for this device, set the default entry
to use model=auto.
Reference: Novell bnc#552154
https://bugzilla.novell.com/show_bug.cgi?id=552154
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add reboot notifier to each codec so that it can do some workarounds
needed for reboot.
So far, patch_sigmatel.c calls its shutup routine for avoiding noises
at reboot on some HP machines.
References: Novell bnc#544779
http://bugzilla.novell.com/show_bug.cgi?id=544779
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TX and RX irq handlers are identical. Merge them
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
APLL_CTL register is configured by the twl4030-codec MFD
driver.
Remove code, which makes changes in the APLL_CTL register,
and replace those with checks against the configured
audio_mclk configuration done in the MFD driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch increases the number of supported audio channels from 4
to 16 and has been sponsored by Shotspotter Inc. It also fixes a
FSYNC rate calculation bug when McBSP is FSYNC master.
Signed-off-by: Graeme Gregory <gg@slimlogic.co.uk>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Our contacts at Conexant suggested that we reduce the external
microphone bias to 50% in order to center the input signal with
the DC input range of the codec. This is because the microphone
port is DC coupled for potential use with sensors.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/478309
The internal microphone on this VAIO model does not work unless the
"auto" quirk is used.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Upcoming change to omap-mcbsp.c require that machine drivers using OMAP
as a DAI master to pass sample rate generator input clock frequency to
the omap-mcbsp.c DAI driver.
Pandora is using 256*Fs output from the TWL4030 codec as an input clock to
the McBSP sample rate generator.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch was generated by
git grep -E -i -l 's(le|el)ct' | xargs -r perl -p -i -e 's/([Ss])(le|el)ct/$1elect/
with only skipping net/netfilter/xt_SECMARK.c and
include/linux/netfilter/xt_SECMARK.h which have a struct member called
selctx.
Signed-off-by: Uwe Kleine-Knig <u.kleine-koenig@pengutronix.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
something-bility is spelled as something-blity
so a grep for 'blit' would find these lines
this is so trivial that I didn't split it by subsystem / copy
additional maintainers - all changes are to comments
The only purpose is to get fewer false positives when grepping
around the kernel sources.
Signed-off-by: Dirk Hohndel <hohndel@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of
requiring manual settings of PCMCIA_DEBUG.
Also, remove all usages of the CS_CHECK macro and replace them with proper
Linux style calling and return value checking. The extra error reporting may
be dropped, as the PCMCIA core already complains about any (non-driver-author)
errors.
CC: Jaroslav Kysela <perex@perex.cz>
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
1. Set the third argument of the snd_device_new to not NULL, so there is
no warning about bug during chip detection. The third argument is not
used in this driver. It was changed in my previous patch.
2. Remove the fm_port and mpu_port fields from the snd_es18xx structure.
They can be converted to function arguments.
3. Remove the dmaN_size fields from the snd_es18xx structure. These
values are used only in pointer functions and can be easily calculated.
4. Remove the ctrl_lock spinlock which is used only in one read function
which is called once during chip initialization. There are many
writes to the same register and they are not protected on purpose
(see the comment ina the snd_es18xx_config_write()).
5. Use the first part of the text5Sources string table as the text4Soruces
table (they are the same).
6. Merge the same cases for the ES1887 and ES1888 when setting chip's caps.
7. Move the snd_es18xx_reset() to __devinit section.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MPC5200 AC97 driver is disabling the slots when a stop
trigger is received, but not reenabling them if the stream
is started again without processing the hw_params again.
This patch fixes the problem by caching the slot enable bit
settings calculated at hw_params time so that they can be
reapplied every time the start trigger is received.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the resolving of the psc_dma_stream pointer to a helper function
to reduce duplicate code
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sound drivers PCM DMA is supposed to free-run until told to stop
by the trigger callback. The current code tries to track appl_ptr,
to avoid stale buffer data getting played out at the end of the
data stream. Unfortunately it also results in race conditions
which can cause the audio to stall.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All DMA blocks are lined up to period boundaries, but the DMA
handling code tracks bytes instead. This patch reworks the code
to track the period index into the DMA buffer instead of the
physical address pointer. Doing so makes the code simpler and
easier to understand.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, CORB/RIRB still remains even if the driver is switched to the
single_cmd mode. The specification says that this should be disabled,
but I hoped this isn't the case; indeed most devices worked together with
CORB/RIRB.
However, Poulsbo (US15W) seems problematic with this setup, and it
requires to disable CORB/RIRB when single_cmd is used.
Now this patch disables CORB/RIRB initialization when the single_cmd
mode is used. Also the unsolicited event is disabled because it can't
work without RIRB.
Reported-and-tested-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix combine_word problem where first octet is not
read properly. The only affected place seems to be the
INPUT_TERMINAL type. Before now, sound controls can be created
with the output terminal's name which is a fallback mechanism
used only for unknown input terminal types. For example,
Line can wrongly appear as Speaker. After the change it
should appear as Line.
The side effect of this change can be that users
can expect the wrong control name in their scripts or
programs while now we return the correct one.
Probably, these defines should use get_unaligned_le16 and
friends.
Signed-off-by: Julian Anastasov <ja@ssi.bg>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPIN_LOCK_UNLOCKED is deprecated. Use __SPIN_LOCK_UNLOCKED instead.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops cause annoying clicks or noises at shutdown/reboot since
the speaker pin is set still high. Apply the same procedure used for
the suspend to avoid such clicks/noises for freeing the codec, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing clk_enable after acquiring the 'audio-bus' clock.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:
omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.
The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.
Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of storing the PID number, take a reference to the task's pid
structure. This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Function hp_bseries_system() is always used, outside of
CONFIG_ boundaries/controls, so move it.
sound/pci/hda/patch_sigmatel.c:5458: error: implicit declaration of function 'hp_bseries_system'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.
Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To unify control names, rename "PC Speaker" to "Speaker" for PPC ALSA drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion in control names for the standard analog PC Beep generator
using a small Internal PC Speaker, rename all related "PC Speaker" and "PC
Beep" controls to "Beep" only. This name is more universal and can be also
used on more platforms without confusion.
Introduce also "Internal Speaker" in ControlNames.txt for systems with
full-featured build-in internal speaker.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/474972
This Sony model needs External Amplifier muted for audible playback, so
make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the TI ADS117x family of multichannel ADCs
and was sponsored by Shotspotter Inc.
Signed-off-by: Graeme Gregory <gg@slimlogic.co.uk>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The XO-1.5 laptop now has a unique subvendor/subproduct ID, which can
be used to automatically select the correct CXT5066 configuration.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).
Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.
Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables GPIO to control mute LED indicator on the HP systems
with the special string in BIOS and applies it with the correct polarity on
HP B-series systems.
It also restores configuration of the pin intended as the second Headphone
on HP B-series systems but configured as something else in the BIOS to
pass MS DTM.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A fairly hefty change in diff terms but no actual code changes, will be
used by the next commit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fixes-s3c-2632-rc5' of git://git.fluff.org/bjdooks/linux:
ARM: S3C2410: Fix sparse warnings in arch/arm/mach-s3c2410/gpio.c
ARM: S3C2440: mini2440: Fix spare warnings
ARM: S3C24XX: Fix warnings in arch/arm/plat-s3c24xx/gpio.c
ARM: S3C2440: mini2440: Fix missing CONFIG_S3C_DEV_USB_HOST
ARM: S3C24XX: arch/arm/plat-s3c24xx: Move dereference after NULL test
ARM: S3C: Fix adc function exports
ARM: S3C2410: Fix link if CONFIG_S3C2410_IOTIMING is not set
ARM: S3C24XX: Introduce S3C2442B CPU
ARM: S3C24XX: Define a macro to avoid compilation error
ARM: S3C: Add info for supporting circular DMA buffers
ARM: S3C64XX: Set rate of crystal mux
ARM: S3C64XX: Fix S3C64XX_CLKDIV0_ARM_MASK value
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't check invalid HP pin
ALSA: dummy - Fix descriptions of pcm_substreams parameter
ALSA: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound)
ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using ALC268
sound: via82xx: deactivate DXS controls of inactive streams
ALSA: snd-usb-caiaq: Bump version number to 1.3.20
ALSA: snd-usb-caiaq: Lock on stream start/unpause
ALSA: snd-usb-caiaq: Missing lock around use of buffer positions
ALSA: sound/parisc: Move dereference after NULL test
ALSA: sound: Move dereference after NULL test and drop unnecessary NULL tests
ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller
ALSA: pcsp - Fix nforce workaround
ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency
ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text
ASoC: Fix possible codec_dai->ops NULL pointer problems
ALSA: hda - Fix capture source checks for ALC662/663 codecs
ASoC: Serialize access to dapm_power_widgets()
Set the codec->bias_level to SND_SOC_BIAS_OFF before changing
the initial bias level to STANDBY.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Wolfson Microelectronics WM8727 DAC, this is a simple
non-configurable DAC.
Signed-off-by: Neil Jones <neil.jones@imgtec.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
alc_automute_pin() might be called even if any HP pin is defined, and
it will result in verbs with NID=0.
This patch adds a check for the validity of HP widget before issuing
any verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert psc-ac97,i2s to platform drivers similar to the davinci ones.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of
requiring manual settings of PCMCIA_DEBUG.
Also, remove all usages of the CS_CHECK macro and replace them with proper
Linux style calling and return value checking. The extra error reporting may
be dropped, as the PCMCIA core already complains about any (non-driver-author)
errors.
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/368629
We should use a quirk mask for these Dell Inspiron Mini9s and Vostro
A90s, as the model=dell quirk appears to enable audio on them.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, if the high-res timers are unavailable, snd-pcsp does not
initialize. People who choose it over pcspkr, loose their console beeps
in that case and get annoyed.
With this patch, the console beeps remain regardless of the high-res
timers. Additionally, the "nopcm" modparam is added to forcibly
disable the PCM capabilities of the driver.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a wrapper call snd_soc_update_bits_locked()
that will take the codec mutex. This call is used
when the codec mutex is not already taken.
Drivers calling snd_soc_update_bits() may wish to
make sure the codec mutex is taken from the driver.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the io_mutex. It has a drawback of serializing
all accesses to snd_soc_update_bits() even when multiple
codecs are in use. In addition, it fails to actually do
its task - during snd_soc_update_bits(), dapm_update_bits()
may also be accessing the same register which may result in
an outdated register value.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When any codec communication error happens, try to switch to the polling
mode first before turning off MSI. MSI gets more stable nowadays, thus
we should keep it on as much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove intermediate snd_audiodrive structure between
snd_card structure and snd_es18xx. This reduces size of
source code and binary driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_card pointer is redundant and code can be easily
changed to work without it.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The OSS driver for Ensoniq SoundScape cards is broken after conversion
to mutexes and a new ALSA snd-sscape driver handles all devices handled
by the OSS one.
The ALSA driver was tested with these cards:
Spea V7 MediaFX
Ensoniq Soundscape Elite
Ensoniq Soundscape VIVO (this card is not handled by the OSS driver)
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Activate the DXS volume controls only when the corresponding stream is
being used. This makes the behaviour consistent with the other drivers
that have per-stream volume controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a proper ifdef CONFIG_SND_DEBUG_VERBOSE to avoid a compile warning:
sound/pci/hda/patch_intelhdmi.c:406: warning: ‘hdmi_get_channel_count’ defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a bug which can result in white noise from the driver after stream
start or unpause.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a race which causes snd_pcm_update_hw_ptr_pos() to report a bug.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variables are unsigned so the test `>= 0' is always true,
the `< 0' test always fails. In these cases the other part of
the test catches wrapped values.
In dac_audio_write() there does not occur a test for wrapped
values, but the test appears redundant.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for VT1818S codec, which is similiar with VT1708S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the NULL test on h is needed in snd_harmony_mixer_init, then the
dereference should be after the NULL test.
Actually, there is a sequence of calls: snd_harmony_create, then
snd_harmony_pcm_init, and then snd_harmony_mixer_init. snd_harmony_create
initializes h, but may indeed leave it as NULL. There was no NULL test at
the beginning of snd_harmony_pcm_init, so I have added one. The NULL test
in snd_harmony_mixer_init is then not necessary, but in case the ordering
of the calls changes, I have left it, and moved the dereference after it.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In pcm.c, if the NULL test on pcm is needed, then the dereference should be
after the NULL test.
In dummy.c and ali5451.c, the context of the calls to
snd_card_dummy_new_mixer and snd_ali_free_voice show that dummy and pvoice,
respectively cannot be NULL.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The attached patch fixes the problems introduced in this commit:
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=eea0579fc85e64e9f05361d5aacf496fe7a151aa
- Fix nForce workaround by honouring the pointer_update var
- Revert "ns" to u64, as per the hrtimer API
- Revert to the zero-delay timer startup, since I can't reproduce any
problem with it (please, give me the hint!)
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Intel IbexPeak HDMI codec supports 2 converters and 3 pins,
which requires converting the cvt_nid/pin_nid to arrays.
The active pin number (the one connected with a live HDMI monitor/sink)
will be dynamically identified on hotplug events.
It exports two HDMI devices, so that user space can choose the A/V pipe
for sending the audio samples.
It's still undefined behavior when there are two active monitors
connected and routed to the same audio converter.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new Intel HDMI codec supports 2 independant HDMI/DisplayPort pipes.
We'll be exporting them as 2 pcm devices. So bump up the allowed number
of HDMI devices.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This unifies the code and data structure,
and makes it easy to add more HDMI devices.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SuperH FSI device have the hardware limitation to use DMA.
If DMA is used, LCD output will be broken.
Maybe there are some solution. But I don't know how to do it now.
This patch remove DMA support and use soft transfer.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An earlier patch merely adds id for 0x80862804.
It has 2/3 cvt/pin nodes and shall be tied to the IbexPeak handler.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.
Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modifying the Kconfig and Makefile in sound/soc/omap folder
to add support for OMAP3517 / AM3517 EVM in Alsa SoC.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adding support for OMAP3517 / AM3517 EVM in Alsa SoC.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The pop-removal specific values are configured for TWL4030 codec
for OMAP3EVM through this patch.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.
If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.
Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.
Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix up some comments, remove all enable_pin() calls (edge widgets
are all enabled by default) and mark the microphone as disabled by
default since it requires a resistor fit to connect it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C64XX DMA implementation will work a lot better with the ability
to enqueue circular buffers as the hardware can do it's own linked-list
management.
Add a function s3c_dma_has_circular() to show that the system can do this
and a flag for the channel.
Update the s3c24xx/s3c64xx I2S DMA code to deal with this.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Mark Brown <broonie@@opensource.wolfsonmicro.com>
codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.
Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.
Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can replace PPC32 || PPC64 as a dependancy with just PPC as all
powerpc platforms (32-bit and 64-bit) define PPC now.
Signed-off-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I thought it could be usefull to add some information on how to get the device
fully supported by loading a line discipline on the modem line.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:
omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.
The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.
Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Hi Mark,
Here is a patch that corrects small omissions I have found in my code.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the remaining headers under plat-omap/include/mach
to plat-omap/include/plat. Also search and replace the
files using these headers to include using the right path.
This was done with:
#!/bin/bash
mach_dir_old="arch/arm/plat-omap/include/mach"
plat_dir_new="arch/arm/plat-omap/include/plat"
headers=$(cd $mach_dir_old && ls *.h)
omap_dirs="arch/arm/*omap*/ \
drivers/video/omap \
sound/soc/omap"
other_files="drivers/leds/leds-ams-delta.c \
drivers/mfd/menelaus.c \
drivers/mfd/twl4030-core.c \
drivers/mtd/nand/ams-delta.c"
for header in $headers; do
old="#include <mach\/$header"
new="#include <plat\/$header"
for dir in $omap_dirs; do
find $dir -type f -name \*.[chS] | \
xargs sed -i "s/$old/$new/"
done
find drivers/ -type f -name \*omap*.[chS] | \
xargs sed -i "s/$old/$new/"
for file in $other_files; do
sed -i "s/$old/$new/" $file
done
done
for header in $(ls $mach_dir_old/*.h); do
git mv $header $plat_dir_new/
done
Signed-off-by: Tony Lindgren <tony@atomide.com>
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the NULL test on jack is needed, then the derefernce should be after the
NULL test.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Codec read/write functions: wait 21us between the pokings of hardware.
Add timeouts to unbounded loops waiting for bits to change.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Verify that the correct register has been received from the codec.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls()
to check the capture source selections. This should be alc882, instead.
Reference: Novell bnc#546918
http://bugzilla.novell.com/show_bug.cgi?id=546918
Signed-off-by: Takashi Iwai <tiwai@suse.de>
48 kHz limit is for slightly better stability, and sample rates other
than 48k (like 96k/192k) are for better sound quality.
We choose better quality, so remove the 48k limit.
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit f0613d5752
ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.
Let's enable all formats/rates as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a reboot on an ARM1176 which amounts to a softreset, it has been
noted that the ALSA driver does not get registered and the probe fails
with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process
of reading from a register the SL1TxBusy bit is set indicating that the
transceiver is busy and remains so until the default timeout occurs.
Set the Power down register 0x26 to an arbitrary value as specified in
the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take
their default state.
Signed-off-by: Philby John <pjohn@in.mvista.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The volume-knob widget needs to be set with 0x7f instead of 0xff
for Dell laptops with STAC9228 codec, too, like the previous commit.
Reference: Novell bnc#545013
http://bugzilla.novell.com/show_bug.cgi?id=545013
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup. The delta bit (bit 7)
shouldn't be set for these devices.
This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.
Reference: Novell bnc#546006
http://bugzilla.novell.com/show_bug.cgi?id=546006
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Simtec Linux Team <linux@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.
Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd
This is the correct error number for telling the USB system that this
driver is not for the device.
Signed-off-by: Tobias Hansen <Tobias.Hansen@physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to
the recent changes. Simply increase the array size to avoid the overflow.
Reported-by: Luca Tettamanti <kronos.it@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power for the amplifier should be handled internally
by the tpa6130a2 driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.
Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes:
sound/pci/hda/patch_via.c: In function 'patch_vt1718S':
sound/pci/hda/patch_via.c:4951: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1716S':
sound/pci/hda/patch_via.c:5441: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt2002P':
sound/pci/hda/patch_via.c:5794: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1812':
sound/pci/hda/patch_via.c:6148: error: expected expression before 'return'
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.
Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chipsets with the isa_dma_bridge_buggy set do not stop DMA during
DMA counter reads. The DMA counter is read in two 8-bit read steps
on x86 platform. Sometimes, such reads happen during higher byte
change so the lower byte is already decremented (rolled over) but
the higher byte is not. It introduces an error that position is
moved 256 bytes ahead of the true position. Thus, the next DMA
position read can return a lower value then the previous read.
If the DMA position is decreased (reversed) the ALSA subsystem is
tricked into the playback underrun error and resets the playback.
It results in a "pop" during a playback.
Work around the issue by reading the counter twice and choosing a higher
value.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The C4231 control set is a superset of the AD1848 control
set so reuse the CS4231 controls definitions for the AD1848.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to customer request, hp should be default to redirected mode,
i.e. PW4 connect select default to to MW0.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To via_control_templates[].
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As init verbs, vt17xx_volume_init_verb is a better place to hold them.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With snd_hda_override_amp_caps.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replaced with via_playback_multi_pcm_prepare/cleanup to support
multi-stream operations
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port
Connectivity field into 'AC_JACK_PORT_COMPLEX'
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1708 does not support unsolicited response, but we need hp detect to
automute speaker. Implemented in workqueue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smart 5.1 is for 3-jacks model, to reuse input pins as outputs.
While off, they act as "line out" / "line in" / "mic in".
While on, they acts as "line out" / "back left/right" / "center/lfe".
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use hp_independent_mode_index to store hp index, and simplify function
via_independent_hp_put with it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1708S and VT1702, deactivate "Headphone Playback Volume" and
"Headphone Playback Mute" control if "Independent HP" mode is OFF.
and rename VT1702 "Independent HP" text.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1708B, VT1708S and VT1702, enter low current mode if no analog
stream is opened and all aa path mute.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enter low power state if AA-Path volume is muted.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
according to customer request, VT1702 AA-Path max volume (12 dB) is too
high, so limit to 0 dB.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IS_VT17*_VENDORID macros are used nowhere, so clean them up.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert CS4231 mixer to dB scale after AD1848 mixer.
Also, add missing microphone boost control for the AD1848
and correct wrong bits for loopback volume on the AD1848.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix coding style errors in the driver.
Also, add missing argument for CMD_XXX_MIDI_VOL command.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow Nvidia HDMI to support more possible sample rates and formats.
At best, the really supported rates and formats should be determined
together with the negotiation with the HDMI receiver, but it's currently
not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3
standard in this regard). As a compromise, we enable all bits, assuming
that all recent devices do support such rates/formats.
Tested-by: Alan Alan <alanwww1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* PLEASE NOTE - this change requires the corresponding update of
envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
in regular mixers. E.g. alsamixer ignores its read-only status
and allows changing the levels with keys which makes no sense.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w
design different from other siblings), it needs to call fixup_automic_adc()
explicitly to set up the auto-mic routing. Otherwise the indices for
int/ext mics aren't set properly.
Reference: Novell bnc#544899
http://bugzilla.novell.com/show_bug.cgi?id=544899
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The "VIA DXS" controls are actually volume controls that apply to the
four PCM substreams, so we better indicate this connection by moving the
controls to the PCM interface.
Commit b452e08e73 in 2.6.30 broke the
restoring of these volumes by "alsactl restore" that most distributions
use; the renaming in this patch cures that regression by preventing
alsactl from applying the old, wrong volume levels to the new controls.
http://bugzilla.kernel.org/show_bug.cgi?id=14151http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
alc_subsystem_id() tries to pick up a headphone pin if not configured,
but this caused side-effects as the problem in commit
15870f05e9.
This patch fixes the driver behavior to pick up invalid HP pins; at least,
the pins that are listed as the primary outputs aren't taken any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS A7K needs additional GPIO1 bit setup; it has to be cleared.
Added a new fixup hook for this laptop so that it works as is.
Refernece: Novell bnc#494309
http://bugzilla.novell.com/show_bug.cgi?id=494309
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Module parameters shouldn't be marked as __devinitdata since they can be
referred via sysfs even after probing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Old Soundscape cards (pre PnP) work only with AD1848 codecs.
If the CS4231 codec is installed it must be used in AD1848
compatible mode.
Also, add gameport support and remove an unused mpu field.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent auto-parser doesn't work for machines with a single output
with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets
the hp_pins[0] while it's listed in line_outs[0].
This ends up with the doubled initialization of the same mixer widget,
and it mutes the DAC route because hp_pins has no DAC assigned.
To fix this problem, just check spec->autocfg.hp_outs and speaker_outs
so that they are really detected pins.
Reference: Novell bnc#544161
http://bugzilla.novell.com/show_bug.cgi?id=544161
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no sense to limit open MIDI connections with limit
as high as ULONG_MAX.
Also, convert more messages to use the snd_printk.
Correct few old and misleading comments as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix for typo in commit 8d50e447d1
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't pass the advanced position to strlcat() but just gives the buffer
head position so that the max size limit can be checked correctly.
Introduced a new helper function to standaralize strlcat() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/
if the codec->dev is NULL:
debugfs/asoc/{codec->name}/
as root for the debugfs entries.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The remove callback has to be marked as __devexit, as the dynamic unbind
is possible.
Reported-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-parser for ALC662/663/272 codecs doesn't work properly when
a speaker is connected to mono NID 0x17, and doesn't handle the dynamic
DAC assignment properly.
This patch fixes the issues and also improves the assignment of DACs
so that HP and speakers can have independent volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Soundblaster X-FI Titenium with emu20k2 the SIDE and SURROUND mute
functions are swapped.
It was checked with 'speaker-test -c 8 -s 3' and (un)mute surround or
'speaker-test -c 8 -s 7' and (un)mute side. The volume seems not
to be affected and works as expected.
Signed-off-by: Sven Eckelmann <sven.eckelmann@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I can't see any reason for struct i2c_driver keywest_driver to not be
static.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model also needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the auto-mic switching between an analog and a digital mic is
needed with IDT codecs, the current driver doesn't reset the connection
of the digital mux.
This patch fixes the behavior by checking both mux connections properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree. So sort
the options such they expand/collapse properly.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch was tested on Toshiba NB200 and is found to enable sound.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037bb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.
An additional firmware initialization code has been moved from the OSS
driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The client->driver pointer can be NULL when i2c-device probing fails
in i2c_new_device(). This patch adds the NULL checks for client->driver
and return the error instead of blind assumption of driver availability.
Reported-by: Tim Shepard <shep@alum.mit.edu>
Cc: Jean Delvare <khali@linux-fr.org>
Cc: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around. Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mia has an undocumented line-out control, and it has to be exposed.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit fdbc66266c, I mistakenly
replaced the capture mixer array for ALC268_ACER to nosrc version
although this should be kept to alt_mixer. Now fixed back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: ALSA bug #0004614https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614
port-A (0x11) - front hp-out
port-D (0x12) - rear line out
port-E (0x1c) - front mic-in
port-F (0x16) - Internal speakers
digital-mic (0x17) - Internal mic
init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware
Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent kernel can handle MSI properly on non-Intel platforms,
let's enable MSI as default.
If any borken device is found, we can add the quirk entry to the list,
which is currently empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Insert "Playback" into the input monitor control names to prevent
alsa-lib from treating these controls as global controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control that allows disabling the high-pass filter of the WM8785 ADC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control to select between sharp and slow roll-of filter responses
of the DACs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control to increase the oversampling factor to 128x on cards with
PCM1796 or PCM1792A DACs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a callback that allows model drivers to modify the default I2S MCLK
rate.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to adjust the headphone amplifier output for
headphones with different impedances.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep a cache of codec registers to avoid unnecessary writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the possibility to route a mix of the two channels of stereo data to
the center and LFE outputs. This is implemented only for models where
the DACs support this, i.e., for the Xonar D1 and DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On card models with two-channel outputs, the base driver can
automatically disable the upmixing control so that the model
drivers do not need to do this.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Essence ST uses the CS2000 chip to generate the DAC master clock, so
we better initialize and program it appropriately.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The virtuoso.c file has become rather big. This patch splits it up so
that only code for very similar card models is in one file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the card is used with a Pericom PI7C9X110 PCI-E/PCI bridge,
reconfigure the latter's PCI buffering to fix an unknown problem.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On cards where the EEPROM was deliberately omitted, we do not need to
try to restore the EEPROM's contents.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move code from the OSS sscape driver in order to support old Soundscape OEM models.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* mark struct vm_area_struct::vm_ops as const
* mark vm_ops in AGP code
But leave TTM code alone, something is fishy there with global vm_ops
being used.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.
Signed-off-by: Graeme Gregory <dp@xora.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After puting a cd-audio inside my laptop there was no sound out here,
so I decided to install alsa-driver with debug level and setup a
model=test, it didn't help, but then I look at source code and added
this few lines, now cd-audio is working both when playback/recording.
[Additional minor fixes of mixer element/item names by tiwai]
Signed-off-by: Lukasz Marcinowski <nowymarluk@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.
It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds support for US-144 when attached on USB1.1.
Unlike the US-122L it uses both USB interfaces 0 and 1.
Signed-off-by: Tobias Hansen <Tobias.Hansen@physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Support for customization of the external clock names
* Adding hooks to playback_pro_open and capture_pro_open, allowing e.g.
limiting available stream rates to a single value when the external
clock rate is detected
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* pro-rate-locking applies to internal clock mode only
* required rate and current rate are compared for internal clock mode only
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* complete support for ak4113
* based on code for ak4114 and ak4117
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* complete support for ak4620
* codec regs listed in proc for all codecs/chips
* adding total regs for each codec
* fixing nb. of steps in input attenuation controls
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Get rid of that commented usage of the now defunct MODULE_PARM macro.
Signed-off-by: Robert P. J. Day <rpjday@crashcourse.ca>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
we cannot set the sampling rate of the device, but can only read it
from the board, so we don't need the member for it.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the rmh bus is not used asynchronously, so it is safe to remove the
specific code pieces.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify snd_pcm_drain() implementation and avoid unneeded array-
allocation for waitqueues. Instead, one waitqueue is used for the
first draining stream, and wait until all streams finished.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows subsytems to provide devtmpfs with non-default permissions
for the device node. Instead of the default mode of 0600, null, zero,
random, urandom, full, tty, ptmx now have a mode of 0666, which allows
non-privileged processes to access standard device nodes in case no
other userspace process applies the expected permissions.
This also fixes a wrong assignment in pktcdvd and a checkpatch.pl complain.
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
New machine driver for WM8580 I2S i/f on SMDK64XX.
By default SoC-Slave is set and WM8580 is configured to use it's
PLLA to generate clocks from a 12MHz crystal attached to WM8580.
[Added dependency on BROKEN since the IISv4 interface hasn't been merged
yet, fixed the PLL API usage and removed the disabling of the PLL in the
hw_free function since that'll break simultaneous playback and record
-- broonie.]
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci: (62 commits)
DaVinci: DM646x - platform changes for vpif capture and display drivers
davinci: DM355 - platform changes for vpfe capture
davinci: DM644x platform changes for vpfe capture
davinci: audio: move tlv320aic33 i2c setup into board files
DaVinci: EDMA: Adding 2 new APIs for allocating/freeing PARAMs
DaVinci: DM365: Adding entries for DM365 IRQ's
DaVinci: DM355: Adding PINMUX entries for DM355 Display
davinci: Handle pinmux conflict between mmc/sd and nor flash
davinci: Add NOR flash support for da850/omap-l138
davinci: Add NAND flash support for DA850/OMAP-L138
davinci: Add MMC/SD support for da850/omap-l138
davinci: Add platform support for da850/omap-l138 GLCD
davinci: Macro to convert GPIO signal to GPIO pin number
davinci: Audio support for DA850/OMAP-L138 EVM
davinci: Audio support for DA830 EVM
davinci: Correct the number of GPIO pins for da850/omap-l138
davinci: Configure MDIO pins for EMAC
DaVinci: DM365: Add Support for new Revision of silicon
DaVinci: DM365: Fix Compilation issue due to PINMUX entry
DaVinci: EDMA: Updating default queue handling
...
Instead of always returnig pointer to the 'audio-bus' clock,
check which clock is used to generate internal clocks and
then return it's pointer.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.
Tested on DA830/OMAP-L137 EVM, DM6467 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the sound system hasn't been utilized yet and we suspend, then we
attempt to save/restore using state that doesn't exist. So use a global
handle instead to reconfigure properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/asoc:
ASoC: remove unused #include <linux/version.h>
ASoC: S3C lrsync function made to work with IRQs disabled.
ASoC: Fix display of stream name in DAPM debugfs
ASoC: Clean up error handling in MPC5200 DMA setup
The headphone and speaker mixer elements aren't properly set for
MSI GX620 with targa-8ch-dig quirk.
Also fixed the speaker volume control for other ALC883-targa quirks,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dc7d7b830e trimmed the platform_device parameter from all of the
suspend functions, but it also accidentally removed it from the resume
function in the Blackfin I2S driver. So restore it.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was
merged, but two places were missed (the probe/resume functions). Restore
handling of this option so it gets initialized properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CDCLK can either be an output generated by the CPU, intended for use
as the CODEC master clock, or an input (probably from the CODEC)
providing a master clock for the IIS block.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables tlv320aic3101 support on DM365 EVM and
it was tested on DM365 EVM rev c.
Note: this patch was created based on temp/asoc branch.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes.
2) Convert from numerical to bit-field values for BCLK selection.
3) Use proper error checking for return value from clk_get
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK
is dead due to improper initialization of CPU or CODEC, the system
gets stuck in the loop because jiffies may never get updated.
Implemented counter based wait mechanism for atleast the same
timeout period.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The pin setup for Dell S14 quirk is rather wrong for the latest driver.
Fixed pin 0x0a, 0x0b, 0x0d and 0x0f.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unnecessary (and buggy) init sequences left for IDT92HD83*
codecs in the previous fixes. The DACs are now dynamically connected,
thus shouldn't be set statically in init verbs. Also, the mono_nid
is detected dynamically, thus shouldn't be set staticaly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
[ARM] Update mach-types
ARM: 5636/1: Move vendor enum to AMBA include
ARM: Fix pfn_valid() for sparse memory
[ARM] orion5x: Add LaCie NAS 2Big Network support
[ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
ARM: 5689/1: Update default config of HP Jornada 700-series machines
ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
ARM: 5688/1: ks8695_serial: disable_irq() lockup
ARM: 5687/1: fix an oops with highmem
ARM: 5684/1: Add nuc960 platform to w90x900
ARM: 5683/1: Add nuc950 platform to w90x900
ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
MMC: MMCI: convert realview MMC to use gpiolib
ARM: 5685/1: Make MMCI driver compile without gpiolib
ARM: implement highpte
ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
...
Fix up trivial conflict in arch/arm/kernel/signal.c.
It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
Add the quirk entry for HP dv6. Also add a workaround for the headphone
detection by setting hp_detect=1 beforehand. Without this, the driver
won't do auto-muting because BIOS doesn't give any HP pin but only a
line-out pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that hp_detect is set even though no headphone pin is
detected. The driver issues, however, an unsol event only to hp_pins[0],
which can be invalid.
This patch adds the check of the valid pin to send an unsol event
at initialization and resume callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines.
However, currently we don't set it unless the model is specified just
for safety reason. But, most machines do need this bit, so this safety
handling is rather annoying.
This patch enables GPIO0 setup as default for them. Many HP / Dell
laptops should work even without model override with this change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.
The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
when != if (...) { <+...x...+> }
(
x->f1 = E
|
(x->f1 == NULL || ...)
|
f(...,x->f1,...)
)
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/usb-audio:
ALSA: usb-audio - Fix types taken in min()
sound: usb-audio: do not make URBs longer than sync packet interval
sound: usb-audio: add MIDI drain callback
sound: usb-audio: use multiple output URBs
sound: usb-audio: use multiple input URBs
sound: usb-audio: Xonar U1 digital output support
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
* topic/soundcore-preclaim:
sound: make OSS device number claiming optional and schedule its removal
sound: request char-major-* module aliases for missing OSS devices
chrdev: implement __[un]register_chrdev()
* topic/oss:
ALSA: allocation may fail in snd_pcm_oss_change_params()
sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
sound: fix OSS MIDI output data loss
* topic/hda: (92 commits)
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ALSA: hda - Add support of Alienware M17x laptop
ALSA: hda - Remove dead codes from patch_sigmatel.c
ALSA: hda - Fix input source selection of IDT92HD73xx
ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
ALSA: hda - Unmute docking line-out as default with AD1984A codec
ALSA: hda - Add another entry for Nvidia HDMI device
ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
ALSA: hda - Fix ALC268/ALC269 headphone pin routing
ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
ALSA: hda - Add more quirk for HP laptops with AD1984A
ALSA: hda - Add / fix model entries for HD-audio driver
ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
ALSA: hda - Improve auto-cfg mixer name for ALC662
ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
ALSA: hda - Improve auto-cfg mixer name for ALC262
ALSA: hda - Improve auto-cfg mixer name for ALC260
ALSA: hda - Improve auto-cfg mixer name for ALC880
...
* topic/asoc: (226 commits)
ASoC: au1x: PSC-AC97 bugfixes
ASoC: Fix WM835x Out4 capture enumeration
ASoC: Remove unuused hw_read_t
ASoC: fix pxa2xx-ac97.c breakage
ASoC: Fully specify DC servo bits to update in wm_hubs
ASoC: Debugged improper setting of PLL fields in WM8580 driver
ASoC: new board driver to connect bfin-5xx with ad1836 codec
ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
ASoC: davinci: i2c device creation moved into board files
ASoC: Don't reconfigure WM8350 FLL if not needed
ASoC: Fix s3c-i2s-v2 build
ASoC: Make platform data optional for TLV320AIC3x
ASoC: Add S3C24xx dependencies for Simtec machines
ASoC: SDP3430: Fix TWL GPIO6 pin mux request
ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
OMAP: McBSP: Use textual values in DMA operating mode sysfs files
ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
ASoC: Select core DMA when building for S3C64xx
...
The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
Receiver-Amplifier and Headphone-Amplifier.
The datasheet for the ak4671 can find at the following url:
http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.
Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the following bugs:
- only reprogram bitdepth if it has changed since last call to hw_params.
- add locking inside ac97_read/write functions:
When reprogramming sample depth, the ac97 unit has to be disabled,
which should not be done in the middle of codec register accesses.
- retry timed-out codec register accesses.
- wait for status bits to set/clear when starting/stopping various
functional blocks; very important after reenabling AC97 unit else
sound may be distorted (e.g. high-pitch noise in 1kHz sine wave).
- clear fifos before/after starting/stopping RX/TX.
- longer timeouts waiting for PSC/AC97 ready after cold reset
with certain codecs this can take ridiculous amounts of time.
Run-tested on various Au1200 platforms with various codecs.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Increase the limit of PCM substreams to 128. The default value is
unchanged; only the max accept value is increased.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the debug proc file to see or change the snd_pcm_hardware fields
to emulate. The parameters can be changed by writing to a proc file like:
# echo periods_min 4 > /proc/asound/card1/dummy_pcm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.
Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix minimum period size for cs46xx cards. This fixes a problem in the
case where neither a period size nor a buffer size is passed to ALSA;
this is the case in Audacious, OpenAL, and others.
Signed-off-by: Sophie Hamilton <kernel@theblob.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it. On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.
To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is for the AK4671 codec driver using this format.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.
When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time. This will get back to the old behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:
[lspci extract]
Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
Subsystem: CLEVO/KAPOK Computer Device [1558:5409]
[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002
[Added a comment about HP mute and the model description by tiwai]
Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Today's linux-next fails to build with
sound/arm/pxa2xx-ac97.c: In function 'pxa2xx_ac97_probe':
sound/arm/pxa2xx-ac97.c:211: error: 'pxa2xx_audio_ops_t' has no member named 'codec_data'
make[2]: *** [sound/arm/pxa2xx-ac97.o] Error 1
It looks like commit e2365bf313 has
introduced this; patch below.
Signed-off-by: Robert Schwebel <r.schwebel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the expire-time calculation in the systimer mode when the buffer
size isn't aligned to the period size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the system-timer mode, snd-dummy driver issues each tick to update
the position. This is highly inefficient and even inaccurate if the
timer can't be triggered at each tick.
Now rewritten to wake up only at the period boundary. The position
is calculated from the current jiffies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow snd-dummy driver to use high-res timer as its timing source
instead of the system timer. The new module option "hrtimer" is added
to turn on/off the high-res timer support. It can be switched even
dynamically via sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card model detection code introduced in 2.6.30 that tries to work
around partially broken EEPROM contents by reading the EEPROM directly
does not handle cards where the EEPROM has been omitted. In this case,
we have to use the default ID to allow the driver to load.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Ozan Çağlayan <ozan@pardus.org.tr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the previous fix of input source for IDT92HD73xx, the amp mux
and amp vol stuff became unused. Let's rip off dead codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mux_nids to select directly the input source instead of mux
mixers so that it works with the current mux enum handler for IDT
92HD73xx codecs.
Also, clean up useless / unnecessary mixer controls and init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As discussed, the patch uses the original TDM order without rewriting.
For the match between TDM slot number and audio channel number, a new
API need be added.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute the docking-station line-out as default on machines with
AD1984A codec chip. It can be still muted via "Dock" mixer switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not forget to program the MCLK ratio for the I2S output.
Otherwise, the master clock frequency can be too high for
the DACs at sample frequencies above 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocation may fail, show if it did.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of automatic mute and mic-switching of the docking
station HP and mic plugs for AD1984A laptop model for some HP machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the headphone pin routing of ALC268/ALC269 codecs. Using alc882
routine doesn't work because alc268/alc269 parser assumes the
independent DACs for both HP and speaker outputs. Need to assign the
DAC depending on the pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1
quirk with ALC889A codec, which caused the silent headphone output.
Also, this gives the individual Headphone and Speaker volume controls.
Reference: kernel bug#14078
http://bugzilla.kernel.org/show_bug.cgi?id=14078
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element. Simpliy
adding the call of get_mux_nids() fixes the problem.
Reference: Novell bnc#534904
https://bugzilla.novell.com/show_bug.cgi?id=534904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the digital mic capture volume wasn't created. This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins. Thus the automatic
capture volume creation should check both directions for digital mics.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.
Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.
Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, the codec setup data structure has to remain for successful
probe.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Julia Lawall