This patch adds support for the JZ4740 internal codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds ASoC support for JZ4740 SoCs I2S module.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.
Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.
The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.
Solution: Mask the result so that it "wraps around", emulating
sign-extension.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* tdm slot has to be configured to get sound working on i.MX25
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.
Consequently when I2C is not set, the compilation fails [1]
This patch fixes this issues, by adding a depencdency on the related HW-
controller.
Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf
Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)
During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.
https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).
The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The header contains an extern that isn't used by anything. Remove.
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>
This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for i2s audio on Bluewater Systems Snapper CL15 module
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove break after return, it is not needed.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
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 the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
IMX_SSI_NET : enable Network Mode
IMX_SSI_SYN : enable Synchronous Mode
IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer
This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.
I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.
We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.
There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding support for openrd client platforms. It's using
the cs42l51 codec and has one mic and one speaker plugs.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables support for the i2s controller available on kirkwood
platforms
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.ul>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.
If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.
If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.
As safety measure do the register reset at remove time also.
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>
Restructure the codec power code in order to be able to hit
off when the codec is not in use.
Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.
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>
It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.
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>
Since the twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.
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>
Since the reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.
According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.
Move all chip initialization code under chip_init, and do
it when the codec is initialized.
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 means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.
Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*
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 the power down wrapper.
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>
Reset most of the codec registers to their chip reset
value.
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>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: snd-usb-caiaq: Bump version number to 1.3.21
ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
ALSA: snd-usb-caiaq: Simplify single case to an 'if'
ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
ALSA: hda: Use LPIB for a Shuttle device
ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
ALSA: hda: Use LPIB for Sony VPCS11V9E
ALSA: usb-audio: fix feature unit parser for UAC2
ALSA: asihpi - Minor code cleanup
ALSA: asihpi - Add support for new ASI8800 family
ALSA: asihpi - Fix bug preventing outstream_write preload from happening
ALSA: asihpi - Fix imbalanced lock path in hw_message
ALSA: asihpi - Remove support for old ASI8800 family
ALSA: asihpi - Add hd radio blend functions
ALSA: asihpi - Remove unused io map functions
ALSA: usb-audio: add support for UAC2 pitch control
ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
ALSA: usb-audio: fix return values
ALSA: usb-audio: parse more format descriptors with structs
sound: Add missing spin_unlock
...
Do not explicity set the default input mode. Use the hardware default
of mode 0 ('Control vinyl'), which is now available.
This reverts commit e3ca4c9.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After removing code, only one case remains. So use an 'if' instead.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This feature was undocumented on early A4DJ units. It is indicated
by lighting both the 'line' and 'phono' lamps at the same time.
Newer units document this and the newer Windows drivers enable this
for all units, so restore the functionality.
This patch simplifies the code and changes the mode mapping to match
the A8DJ, favouring simpler code and consistency over keeping the
existing mapping.
Both 'Control vinyl' and 'Phono' input modes enable the hardware
preamp. The difference is the input impedance.
This reverts commit 9a9527e.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/551949
Symptom: On the reporter's Shuttle device, using PulseAudio in Ubuntu
10.04 LTS results in "popping clicking" audio with the PA crashing
shortly thereafter.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's volume with PulseAudio.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Christian Mehlis <mehlis@inf.fu-berlin.de>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played
sound although headphones were plugged in. Using model=ideapad with
latest alsa-git kernel fixed this. So adding this quirk to use ideapad
for another Thinkpad Edge variant seems sensible.
Cc: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Andreas Herrmann <andreas.herrmann3@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/586347
Symptom: On the Sony VPCS11V9E, using GStreamer-based applications with
PulseAudio in Ubuntu 10.04 LTS results in stuttering audio. It appears
to worsen with increased I/O.
Test case: use Rhythmbox under increased I/O pressure. This symptom is
reproducible in the current daily stable alsa-driver snapshots (at least
up until 21 May 2010; later snapshots fail to build from source due to
missing preprocessor directives when compiled against 2.6.32).
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Lauri Kainulainen <lauri@sokkelo.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a small off-by-one bug which causes the feature unit to announce a
wrong number of channels. This leads to illegal requests sent to the
firmware eventually.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add hd radio blend functions. HPI version inc to 4.03.25.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This request is again handled differently in comparison to UAC1.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 devices have their information about pitch control stored in a
different field. Parse it, and emulate the bits for a v1 device.
A new struct uac2_iso_endpoint_descriptor is added.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-1 is not a good return value as it means -EPERM, "not permitted".
Choose -ENOTSUPP instead, which is what the code really wants to tell
its callers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a spin_unlock missing on the error path.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression E1;
@@
* spin_lock(E1,...);
<+... when != E1
if (...) {
... when != E1
* return ...;
}
...+>
* spin_unlock(E1,...);
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These scales should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These should be regular rather than linear scales.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
This updates the i.MX SSI driver to make it compatible with the ASoC tree
following the move of DMA parameters from the DAI to the audio substream
object.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Add a module option to allow the GPR mixer controls to have the full
resolution of the hardware, i.e., 0...2^31-1 instead of 0...100.
Because of bugs in userspace tools like alsactl and alsamixer, this is
not yet enabled by default.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the cleanup of the hw_ptr update functions in 2.6.33, the calculation
of the delta value was changed to use the modulo operator to protect
against a negative difference due to the pointer wrapping around at the
boundary.
However, the ptr variables are unsigned, so a negative difference would
result in the two complement's value which has no relation to the actual
difference relative to the boundary; the result is typically some value
near LONG_MAX-boundary. Furthermore, even if the modulo operation would
be done with signed types, the result of a negative dividend could be
negative.
The invalid delta value is then caught by the following checks, but this
means that the pointer update is ignored.
To fix this, use a range check as in the other pointer calculations.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check that the interrupt raised for a stream is actually a buffer
completion interrupt before handling it as one. Otherwise, memory
errors or FIFO xruns would be interpreted as a pointer update and could
break the stream timing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/576160
Symptom: Currently (2.6.32.12) the Dell M1730 uses the 3stack model
quirk. Unfortunately this means that capture is not functional out-
of-the-box despite ensuring that capture settings are unmuted and
raised fully.
Test case: boot from Ubuntu 10.04 LTS live cd; capture does not
work.
Resolution: Correct the model quirk for Dell M1730 to rely on the
BIOS configuration.
This patch also trivially sorts the quirk into the correct section
based on the comments.
Reported-and-Tested-By: <picdragon99@msn.com>
Tested-By: Daren Hayward
Tested-By: Tobias Krais
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
