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If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Log the values we're getting back from the DC servo and the values we
write to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add back the zero return value (activate workqueue) when
bdl_pos_adj is nonzero for position check.
Do the position related check only for first next period
using wallclk counter.
Return -1 value (ignore interrupt) when period_bytes
variable is zero.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.
Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
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>
Use 24Mhz WALLCLK register to ignore too early interrupts and
wrong interrupt status. The bad timing confuses the higher ALSA
layer and causes audio skipping. More information about behaviour
and debugging can be found in kernel bz#15912.
https://bugzilla.kernel.org/show_bug.cgi?id=15912
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Old HP dv series seem to use the GPIO pin 0 for controlling the mute LED
although the pin is a large package, where the newer models use GPIO 3
in such a case. For fixing the regression from the previous kernels,
set spec->gpio_led statically for these model quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDA controller in US15W (Poulsbo) reports inaccurate position values
for capture streams when using the LPIB read method, resulting in
distorted recordings.
However, using the position buffer is broken for playback streams,
resulting in a fallback to the LPIB method with the current driver.
This patch works around the issue by independently detecting the read
position method for capture and playback streams.
The patch will not have any effect if the position fix method is
explicitly set.
[Code simplified by tiwai]
Signed-off-by: Shahin Ghazinouri <shahin.ghazinouri@pelagicore.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-May/027525.html
As reported on the mailing list, we also need to cap to the 0 dB offset
for Lenovo models, else the sound will be distorted.
Reported-and-Tested-by: Tim Starling <tstarling@wikimedia.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation. I did this because request more
accurately represents what it actually does.
Also, I added a string based ABI for users wanting to use a string
interface. So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface. (someone asked me for it and I don't think
it hurts anything.)
This patch updates some documentation input I got from Randy.
Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
This reverts commit 7aee67466536bbf8bb44a95712c848a61c5a0acd.
As it doesn't seem to be universally valid for all mainboard revisions of
the D945GCLF2 and breaks snd-hda-intel/ snd-hda-codec-realtek on the Intel
Corporation "D945GCLF2" (LF94510J.86A.0229.2009.0729.0209) mainboard.
00:1b.0 Audio device [0403]: Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01)
Signed-off-by: Stefan Lippers-Hollmann <s.l-h@gmx.de>
Cc: <stable@kernel.org> [2.6.33]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables the SPDIF output pin by default. It also enables
it for quirks related to Levono docking stations (x200 and 25041,
identified with the same 17aa:20f2 ID). Even though not all Lenovo
docking stations have SPDIF connectors, enabling the pin by default
shouldn't be a problem for anyone.
Other quirks remain unmodified.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows more flexible integration with subsystem features.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.
Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ideapad quirks working for my ThinkPad X100e (microphone is not tested).
Signed-off-by: Andrej Gelenberg <andrej.gelenberg@udo.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev_node_t was only used to transport some minor/major numbers
from the PCMCIA device drivers to deprecated userspace helpers.
However, only a few drivers made use of it, and the userspace
helpers are deprecated anyways. Therefore, get rid of dev_node_t .
As a first step, remove any usage of dev_node_t from drivers which
only wrote to this typedef/struct, but did not make use of it.
CC: linux-bluetooth@vger.kernel.org
CC: Harald Welte <laforge@gnumonks.org>
CC: linux-mtd@lists.infradead.org
CC: linux-wireless@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Instead of the old pcmcia_request_irq() interface, drivers may now
choose between:
- calling request_irq/free_irq directly. Use the IRQ from *p_dev->irq.
- use pcmcia_request_irq(p_dev, handler_t); the PCMCIA core will
clean up automatically on calls to pcmcia_disable_device() or
device ejection.
- drivers still not capable of IRQF_SHARED (or not telling us so) may
use the deprecated pcmcia_request_exclusive_irq() for the time
being; they might receive a shared IRQ nonetheless.
CC: linux-bluetooth@vger.kernel.org
CC: netdev@vger.kernel.org
CC: linux-wireless@vger.kernel.org
CC: linux-serial@vger.kernel.org
CC: alsa-devel@alsa-project.org
CC: linux-usb@vger.kernel.org
CC: linux-ide@vger.kernel.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
As it's only used there it makes no sense relying on pcmcia_request_irq().
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
The commit 24e4a1211f691fc671de44685430dbad757d8487
ALSA: info - Use standard types for info callbacks
introduced a wrong type to snd_opl4_mem_proc_write() for pos argument.
Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.
Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.
Also, a new PnP id is added for the card I acquired (the change
was tested on this card).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 23caaf19b ("ALSA: usb-mixer: Add support for Audio Class v2.0")
broke support for Class1 devices due to two faulty changes. This patch
fixes it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-and-Tested-by: The Source <thesourcehim@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.
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>
This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
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 support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)
If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:
snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
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>
Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control
It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.
The codec reset values are considered safe in all environmnts.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
We had a fixed external amp setup enabled for ALC888, but this seems
unnecessary. The amps are controlled rather by GPIOs.
Let's remove it now.
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: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
ALSA: take tu->qlock with irqs disabled
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
ALSA: hda - fix array indexing while creating inputs for Cirrus codecs
ALSA: es968: fix wrong PnP dma index
