Krzysztof Helt [Mon, 10 May 2010 07:47:32 +0000 (09:47 +0200)]
ALSA: Merge es1688 and es968 drivers
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>
Daniel Mack [Sat, 8 May 2010 09:24:56 +0000 (11:24 +0200)]
ALSA: sound/usb: fix UAC1 regression
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>
Peter Ujfalusi [Fri, 7 May 2010 11:24:11 +0000 (14:24 +0300)]
ASoC: tpa6130a2: Fix for the custom kcontrol functions
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>
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>
Peter Ujfalusi [Fri, 7 May 2010 11:05:49 +0000 (14:05 +0300)]
ASoC: core: Support for limiting the volume
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:
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>
Jassi Brar [Fri, 7 May 2010 01:21:39 +0000 (10:21 +0900)]
ARM: S3C2412: DMA: Remove I2S FIFO address
The S3C DMA API doesn't make use of hw_addr.to/from and also
the FIFO addresses are provided from the I2S drivers. So these
fields are redundant.
This patch removes the hw_addr.to/from fields for I2S and the
inclusion of header, paving way for the header to be moved closer
to the I2S controller drivers.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 6 May 2010 09:04:25 +0000 (12:04 +0300)]
ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power
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>
Peter Ujfalusi [Thu, 6 May 2010 07:37:18 +0000 (10:37 +0300)]
ASoC: tpa6130a2: Support for limiting gain
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>
Jarkko Nikula [Wed, 5 May 2010 10:02:03 +0000 (13:02 +0300)]
ASoC: tlv320aic3x: Add platform data and reset gpio handling
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>
Jarkko Nikula [Wed, 5 May 2010 08:14:22 +0000 (11:14 +0300)]
ASoC: omap: Add basic audio support for Nokia RX-51/N900
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.
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>
ASoC: S3C: I2Sv2: New field for controller feature
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 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>
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>
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>
Daniel T Chen [Wed, 28 Apr 2010 22:00:11 +0000 (18:00 -0400)]
ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
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>
Dan Carpenter [Wed, 28 Apr 2010 08:29:14 +0000 (10:29 +0200)]
ALSA: take tu->qlock with irqs disabled
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:
Daniel T Chen [Wed, 5 May 2010 02:07:58 +0000 (22:07 -0400)]
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
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>
Daniel T Chen [Tue, 4 May 2010 00:39:31 +0000 (20:39 -0400)]
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
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>
Peter Ujfalusi [Tue, 4 May 2010 08:08:18 +0000 (11:08 +0300)]
ASoC: tpa6130a2: TLV mapping for tpa6140a2
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>
Peter Ujfalusi [Fri, 30 Apr 2010 11:59:36 +0000 (14:59 +0300)]
ASoC: tlv320dac33: Support for turning off the codec
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>
Peter Ujfalusi [Fri, 30 Apr 2010 11:59:35 +0000 (14:59 +0300)]
ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure
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>
Peter Ujfalusi [Fri, 30 Apr 2010 11:59:34 +0000 (14:59 +0300)]
ASoC: tlv320dac33: Revised module loading, and DAC33 ID read
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>
Peter Ujfalusi [Fri, 30 Apr 2010 11:59:33 +0000 (14:59 +0300)]
ASoC: tlv320dac33: Optimize power up, and restore
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>
Peter Ujfalusi [Thu, 29 Apr 2010 07:58:09 +0000 (10:58 +0300)]
ASoC: TWL4030: Remove OUTL/R outputs
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>
Peter Ujfalusi [Thu, 29 Apr 2010 07:58:08 +0000 (10:58 +0300)]
ASoC: TWL4030: AIF/APLL fix in DAPM domain
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>
Mark Brown [Wed, 28 Apr 2010 17:36:10 +0000 (18:36 +0100)]
ASoC: Add WM9090 amplifier driver
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>
Liam Girdwood [Fri, 26 Mar 2010 20:05:54 +0000 (20:05 +0000)]
ASoC: tlv320dac33 - disable regulators at i2c remove()
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>
Liam Girdwood [Mon, 22 Mar 2010 19:35:06 +0000 (19:35 +0000)]
ASoC: zoom2 - update DAPM pins
Remove bogus twl4030 pins
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>
Liam Girdwood [Mon, 22 Mar 2010 19:30:54 +0000 (19:30 +0000)]
ASoC: pandora - update DAPM pins
Remove bogus TWL4030 pins.
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>
Jarkko Nikula [Mon, 26 Apr 2010 12:49:14 +0000 (15:49 +0300)]
ASoC: tlv320aic3x: Add basic regulator support
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>
Jarkko Nikula [Mon, 26 Apr 2010 12:49:12 +0000 (15:49 +0300)]
ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_level
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>
Jarkko Nikula [Mon, 26 Apr 2010 12:49:11 +0000 (15:49 +0300)]
ASoC: tlv320aic3x: Remove unused version string
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>
Mark Brown [Fri, 23 Apr 2010 16:39:23 +0000 (17:39 +0100)]
ASoC: Correct inversion of speaker mixer PCM switch
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>
Peter Ujfalusi [Fri, 23 Apr 2010 07:10:01 +0000 (10:10 +0300)]
ASoC: tlv320dac33: FIFO caused delay reporting
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>
Peter Ujfalusi [Fri, 23 Apr 2010 07:10:00 +0000 (10:10 +0300)]
ASoC: tlv320dac33: Calculate the interface speed during bursts
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>
Peter Ujfalusi [Fri, 23 Apr 2010 07:09:59 +0000 (10:09 +0300)]
ASoC: tlv320dac33: Change magic numbers used in Mode7
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>
Peter Ujfalusi [Fri, 23 Apr 2010 07:09:58 +0000 (10:09 +0300)]
ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode
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>
Peter Ujfalusi [Fri, 23 Apr 2010 07:09:57 +0000 (10:09 +0300)]
ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1
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>
Barry Song [Wed, 21 Apr 2010 09:36:49 +0000 (17:36 +0800)]
ASoC: ad193x: fix typo, delete redundant space
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>
Barry Song [Wed, 21 Apr 2010 09:36:48 +0000 (17:36 +0800)]
ASoC: ad193x: fix wrong register setting in ad193x_set_dai_fmt
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>
Hans de Goede [Fri, 23 Apr 2010 09:26:43 +0000 (05:26 -0400)]
ALSA: snd-es1968: Make hardware volume buttons an input device (rev2)
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>
Hans de Goede [Fri, 23 Apr 2010 09:26:42 +0000 (05:26 -0400)]
ALSA: snd-maestro3: Make hardware volume buttons an input device (rev2)
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>
Daniel T Chen [Thu, 22 Apr 2010 21:54:45 +0000 (17:54 -0400)]
ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558
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>
Daniel T Chen [Wed, 21 Apr 2010 23:55:43 +0000 (19:55 -0400)]
ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203
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>
Hans de Goede [Wed, 21 Apr 2010 15:04:07 +0000 (11:04 -0400)]
ALSA: snd-meastro3: Document hardware volume control a bit
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>
Hans de Goede [Wed, 21 Apr 2010 15:04:08 +0000 (11:04 -0400)]
ALSA: snd-meastro3: Ignore spurious HV interrupts during suspend / resume
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>
Hans de Goede [Wed, 21 Apr 2010 15:04:06 +0000 (11:04 -0400)]
ALSA: snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
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>
Daniel T Chen [Thu, 22 Apr 2010 11:15:26 +0000 (07:15 -0400)]
ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526
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>
Daniel T Chen [Thu, 22 Apr 2010 00:41:52 +0000 (20:41 -0400)]
ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio XPS 1645
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>
projects / karo-tx-linux.git / log