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 6f3991152f.
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
Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.
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>
In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c
While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.
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>
Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.
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>
Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock
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>
No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.
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>
Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.
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 IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.
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>
Define more bit definitions in the order of mainline
support for the SoC.
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 header for I2Sv2
linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/
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>
BugLink: https://launchpad.net/bugs/541802
The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_cxt5045() for all Packard Bell models.
Reported-by: Valombre
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper
HP and Mic support.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should disable irqs when we take the tu->qlock because it is used in
the irq handler. The only place that doesn't is
snd_timer_user_ccallback(). Most of the time snd_timer_user_ccallback()
is called with interrupts disabled but the the first ti->ccallback()
call in snd_timer_notify1() has interrupts enabled.
This was caught by lockdep which generates the following message:
> =================================
> [ INFO: inconsistent lock state ]
> 2.6.34-rc5 #5
> ---------------------------------
> inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage.
> dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes:
> (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer]
> {HARDIRQ-ON-W} state was registered at:
> [<c1048de9>] __lock_acquire+0x654/0x1482
> [<c1049c73>] lock_acquire+0x5c/0x73
> [<c125ac3e>] _raw_spin_lock+0x25/0x34
> [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer]
> [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer]
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/549267
The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.
Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/573284
The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.
Reported-by: Andy Couldrake <acouldrake@googlemail.com>
Tested-by: Andy Couldrake <acouldrake@googlemail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes a problem where cards show up as only having a single mixer
element, suppressing all sound output.
Signed-off-by: Brian J. Tarricone <brian@tarricone.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.
Provide different mixer control for the chips with correct
TLV mapping.
User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140
The way machine drivers are using this amplifier remained
the same.
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>
Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).
After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).
The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.
There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, but we does not
need to execute the playback related configuration
2. Playback caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, and also we need
to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
still on.
Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.
Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.
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>
As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.
The substream must be easily available in other places than
pcm_* callbacks.
Manage a pointer in _startup, and _shutdown for this.
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>
Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.
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>
On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.
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>
OUTL/R are leftovers from the original driver, and they
are no longer needed.
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>
This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.
We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */
/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */
/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */
The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).
The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.
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>
This enables autoloading of the TXx9 sound driver on RBTX4927.
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
To: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Cc: Linux MIPS Mailing List <linux-mips@linux-mips.org>
Patchwork: http://patchwork.linux-mips.org/patch/1101/
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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>
The SYSCLK source is automatically managed when configuring the PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.
Bypass mode: FIFO is bypassed, report 0 as delay
Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.
Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.
Phase1: when we received the alarm threshold, but our workqueue has
not been executed (safeguard phase). Just count the played out
samples since ts1 and subtract it from the alarm threshold
value.
Phase2: During nSample burst (after writing to nSample register), count
the played out samples since ts1, count the samples received
since ts2 (in a burst). Estimate the FIFO depth using these and
alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
samples since ts1. Estimate the FIFO depth using the nSample
configuration and the alarm threshold value.
Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received
Interrupts are coming when DAC33 FIFO depth reaches upper threshold.
Phase1: Draining phase (after the burst), counting the played out
samples since ts1, and subtract it from the upper threshold
value.
Phase2: During burst operation. Using the pre calculated time needed to
play out samples from the buffer during the drain period (from
upper to lower threshold), move the time window to cover the
estimated time from the burst start to the current time.
Calculate the samples played out since lower threshold and also
the samples received during the same time.
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>
When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.
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>
Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.
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>
There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.
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>
Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.
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>
There is only one dma for the ESS ES968 based board.
Its index is 0 and not 1.
This make the es968 card working.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the core jack implementation and allow reporting on the status
of NULL jacks, avoiding the need to check in detection implementations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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>
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>
The hardware volume handling code in essence just detects key presses, and
then does some hardcoded modification of the master volume based on which key
is pressed.
Clearly the right thing to do here is just report these keypresses to
userspace and let userspace decide what to with them.
This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While working on the sound suspend / resume problems with my laptop
I noticed that the hardware volume handling code in essence just detects
key presses, and then does some hardcoded modification of the master volume
based on which key is pressed.
This made me think that clearly the right thing to do here is just report
these keypresses to userspace and let userspace decide what to with them.
This patch adds a Kconfig option which when enabled reports the volume
buttons as keypresses using an input device. When enabled this option
also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock
and the need for using a tasklet in general.
As an added bonus the keys now work identical to volume keys on a (usb)
keyboard with multimedia keys, providing visual feedback of the volume
level change, and a better range of the volume control (with a properly
configured desktop environment).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/568600
The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.
This change is necessary for 2.6.32.11 and 2.6.33.2 alike.
Reported-by: Andy Ross <andy@plausible.org>
Tested-by: Andy Ross <andy@plausible.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/459083
The OR has verified with 2.6.32.11 and the latest alsa-driver stable
daily snapshot that position_fix=1 is necessary for the external mic
to work and for PulseAudio not to crash constantly.
This patch is necessary also for 2.6.32.11 and 2.6.33.2.
Reported-by: <imwithid@yahoo.com>
Tested-by: <imwithid@yahoo.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While working on a fix for the volume being muted on the allegro in my
Compaq EVO N600C after suspend, I've learned a few things about the hardware
volume control worth documenting. The actual fix for the suspend / resume
issue is in the next patch in this set.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore spurious HV interrupts during suspend / resume, this avoids
mistaking them for a mute button press. This is not very pretty but
it seems the only way to fix the master volume control gets muted
after suspend issue I'm seeing. Note that the es1968 driver is doing
exactly the same.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this quirk sound stops working after suspend resume. With this quirk,
one still needs to manually unmute the master volume control after a suspend /
/ resume cycle. That is fixed in another patch in this set.
Note that this patch was submitted to the alsa bug tracker a long time ago:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/567494
The OR has verified that the existing model quirk, ALC880_UNIWILL,
is insufficient for audible playback and capture by default. Instead,
the ALC880_F1734 model quirk needs to be used.
This change is necessary for both 2.6.32.11 and 2.6.33.2.
Reported-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Tested-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/553002
The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.
This change is necessary for 2.6.32.11 and 2.6.33.2 alike.
Reported-by: Robert Chambers
Tested-by: Robert Chambers
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of AudioScience ASI boards.
The driver has been tested for years on alsa-driver external tree,
now finally got merged to the kernel.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An index equal to the array size may not be accessed.
Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When HP laptop with AD1981 codec is suspended and the docking-station
is connected before the resume, the outputs get confused, and wrongly
routed still to the speaker. This is because of a change in 2.6.34-rc1
ea52bf260e
ALSA: hda: Add powerdown for Analog Devices HDA codecs
The problem was the added resume callback that doesn't consider the
modified init hook. The fix is simply remove the resume callback here
and make the resume normally. This doesn't change any behavior intended
in the commit above (for shutting down the sound at suspend) but only
fixes the resume.
Reported-and-tested-by: Frans Pop <elendil@planet.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Doing so causes a deadlock, so just signal the timer to stop
using an atomic variable.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed 8 checkpatch errors (ERROR: do not use assignment in if condition)
in sound/i2c/i2c.c.
Signed-off-by: Brian Waters <brianmwaters@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
attached please find a patch that adds support for at least the T410s
HDA codec. Most likely it will also add support for the T410 and T510
based models.
The patch was derived from Ideapad support. Support for the laptop's and
docking-station output connectors as well as the docking-station microphone
connector and the laptops internal devices has been tested. Since it has been
developed without a data-sheet available, support for digital outputs and the
laptop's microphone input may well be incorrect.
Microphone mute functionality is not included:
The microphone mute button seems to be reported through thinkpad_acpi key
0000101b. The mute button LED seems to be wired to thinkpad_acpi led
number 15.
Signed-off-by: Jens Taprogge <jens.taprogge@taprogge.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Biostar mobo seems to give a wrong DMA position, resulting in
stuttering or skipping sounds on 2.6.34. Since the commit
7b3a177b0d, "ALSA: pcm_lib: fix "something
must be really wrong" condition", makes the position check more strictly,
the DMA position problem is revealed more clearly now.
The fix is to use only LPIB for obtaining the position, i.e. passing
position_fix=1. This patch adds a static quirk to achieve it as default.
Reported-by: Frank Griffin <ftg@roadrunner.com>
Cc: Eric Piel <Eric.Piel@tremplin-utc.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the matching model for Clevo laptop M570U.
Signed-off-by: Joerg Schirottke <master@kanotix.com>
Tested-by: Maximilian Gerhard <maxbox@directbox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the notification of elapsed periods is not very exact.
Increase minimum periods to 4 as suggested by Liam Girdwood.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some VIA codecs have no multiple source selection for headphone pins,
thus it's useless (and wrong) to create "Independent HP" control on them.
This patch adds the check of connections to skip the control in such a
case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 5b0cb1d850
ALSA: hda - add more NID->Control mapping
breaks the control element allocation by returning a wrong value.
Let's fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set no_llseek to llseek file ops of each sound component (but for hwdep).
This avoids the implicit BKL invocation via generic_file_llseek() used
as default when fops.llseek is NULL.
Also call nonseekable_open() at each open ops to ensure the file flags
have no seek bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The llseek implementation is identical for existing driver implementations,
so let's merge to the common layer. The same code for the text proc file
can be used even for the binary proc file.
The driver can provide its own llseek method if needed. Then the common
code will be skipped.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the validity of the file position in the common info layer before
calling read or write callbacks in assumption that entry->size is set up
properly to indicate the max file size.
Removed the redundant checks from the callbacks as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 29a4f2d3 used writel() at offset 0x26 which is
half-word aligned causing unaligned exceptions on a
Cortex-A8. The original patch solved the "aaci-pl041 fpga:04:
ac97 read back fail" issue on a soft reset. Reading from any
arbitrary aaci register seems to solve this issue.
Signed-off-by: Philby John <pjohn@mvista.com>
Acked-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for sound through the WM8750 codec on Zipit Z2.
Also, this patch incorporates support for detecting headset jack
insertion through the jack detection API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Bill Gatliff <bgat@billgatliff.com>
Acked-by: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF
ground or Hi-Z to make the headphone working. Other than that, model=auto
works fine, so let's use model=auto with a specific fix-up table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few enhancement / fixes for fix-up table of some Realtek codecs:
- Apply fix-ups only for the auto model
- Apply additional verbs after normal init verbs
- Add a debug print to show the fix-up application
This is basically a preliminary work for the next fix for Sony VAIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after
disconnection. This is due to the access to the endpoints which have
been already released at disconnection while the files are still alive.
This patch fixes the problem by checking disconnection state at
snd_usbmidi_output_drain() and by releasing urbs but keeping the
endpoint instances until really all freed.
Tested-by: Tvrtko Ursulin <tvrtko@ursulin.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin. This can be a problem when mode=auto
and there is only one input pin. Then user can't change the connection
at all.
This patch adds the code to initialize the input pin connection of these
codecs.
Reference: Novell bnc#594363
https://bugzilla.novell.com/show_bug.cgi?id=594363
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.
Also, this patch fixes the Jive and Spitz machine.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Corrected HP and mic pins for ALC269vb amic and dmic models.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269vb has an alternative HP pin 0x21 in addition.
Fix the parser to recognize it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a local mutex instead of BKL. This should suffice since each device
type has also its open_mutex.
Also, a bit of clean-up of the legacy device auto-loading code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a regular timer results in poll times < 1 jiffie with small
buffers, so we loaded the timer with the actual jiffie value. We can
be more accurate using a hrtimer. Also, we have to call
snd_pcm_period_elapsed after playing period_bytes and not
runtime->period_size (which is in samples and not in bytes).
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking if we are DMA capable we have to check for the
IMX_SSI_DMA flag which is already set from platform_data instead
of setting it again when we want to do DMA.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
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:
ALSA: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
This adds support for the Medion WIM2160 soundcard.
There's no PCI quirk added because it has the same PCI id as the
Medion MD2.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assign DACs properly to each output. Currently, the front output is bound
to HP/speaker outputs blindly, but they should be assigned to individual
DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".
Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Cc: <stable@kernel.org>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 6a4f2ccb46 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections. Fixed now.
Reference: bko#15689
https://bugzilla.kernel.org/show_bug.cgi?id=15689
Reported-by: Adrian Ulrich <kernel@blinkenlights.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.
Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we may try to power down multiple times when the using
idle bias off and the driver is removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo
operations has been deprecated and with some more recente revisions
may perform incorrectly, especially when only analogue bypass paths
are in use. Switch to using readback from the DC servo command
register instead, which is supported for all devices. Without this
unacceptably long timeouts may be observed in some circumstances.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If we need to offset correct the DC servo then don't use runtime
recalibration since that is likely to introduce further offsets
which will be evident on powerdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
