Remove the use of set_hs_extmute callback and let the codec driver to
handle the extmute GPIO.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The external mute (if it is in use) is handled by a GPIO line. Prepare to
remove the set_hs_extmute callback and replace it with:
hs_extmute_gpio: the GPIO number to use for external mute
When the users of set_hs_extmute has been converted the callback can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Original author: Russell King <rmk+kernel@arm.linux.org.uk>
Switch the omap-pcm to use dmaengine.
Certain features are not supported by after dmaengine conversion:
1. No period wakeup mode
DMA engine has no way to communicate this information through
standard channels.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the dma_data for the stream (snd_soc_dai_set_dma_data) at dai_startup
time so omap-pcm will have access to the needed information regarding to
the DMA channel earlier.
This is needed for the clean dmaengine support.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of the OMAP DMA data type definition the data_type will be used to
specify the number of bits the DMA word should be configured or 0 in case
when based on the stream's format the omap-pcm can decide the needed DMA
word size.
This feature is needed for the omap-hdmi where the sDMA need to be
configured for 32bit word type regardless of the audio format used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
omap-pcm can figure out the correct dma_type based on the stream's format.
In this way we can get rid of the plat/dma.h include from these drivers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Get the needed resources in a correct way and avoid using defines for them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the format of the stream the omap-pcm can decide alone what data
type should be used with by the sDMA.
Keep the possibility for OMAP dai drivers to tell omap-pcm if they want to
use different data type. This is needed for the omap-hdmi for example which
needs 32bit data type even if the stream format is S16_LE.
The check if (dma_data->data_type) is safe at the moment since omap-pcm
does not support 8bit samples (OMAP_DMA_DATA_TYPE_S8 == 0x00).
The next step is to redefine the meaning of dma_data->data_type to unblock
this limitation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The omap-pcm platform driver no longer needs this parameter to select
between ELEMENT and PACKET mode. The selection is based on the configured
packet_size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we only have element or packet synchronization we can use the
dma_data->packet_size to select the desired mode:
if packet_size is 0 we use ELEMENT mode
if packet_size is not 0 we use PACKET mode for sDMA synchronization.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When McBSP is configured in threshold mode we can use sDMA packet mode in
all cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mpc5200-psc-ac97 driver does not enumerate attached ac97 devices, so
register the device here.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert pcm030-audio-fabric to use the new snd_soc_register_card api
instead of the older method of registering a separate platform device.
Create the dai_link to the mpc5200_psc_ac97 platform using the device tree.
Convert the pcm030-audio-fabric driver to a platform-driver and add a
remove function.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
mpc52xx socs do not define FSL_SOC but need SND_POWERPC_SOC defined to build
ASoC drivers.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use DAPM mapping for stream events and give unique names for the streams.
This change also fixes the following warning:
twl6040-codec twl6040-codec: Failed to create Capture debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use DAPM mapping for stream events and give unique names for the streams.
This change also fixes the following warning:
twl4030-codec twl4030-codec: Failed to create Capture debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested with LPIB delay without any issues.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DMA Position in Buffer (DPIB) should be used for
ring buffer management, while LPIB register provides
information on the number of samples transfered on
the link. The difference between the two pieces of
information corresponds to hardware/DMA buffering.
This patch reports this difference in runtime->delay, and
removes the use of the COMBO mode on recent Intel hardware.
Credits to Takashi Iwai for an initial patch.
[rebased to for-next branch and replaced snd_printk() with
snd_printdd() by tiwai]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC bits are typically used to start multiple
streams synchronously. It makes sense to use them
for a single stream for a more predictable startup
sequence. The transfers only start once the DMA and
FIFOs are ready. This results in a better correlation
between timestamps and number of samples played.
Credits to Kar Leong Wang for suggesting this
improvement.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If headphone jack can't detect plug presence, and we have the jack in
the jack table, snd_hda_jack_detect will return the plug as always
present (as it'll be considered as a phantom jack). The problem is that
when this happens, line out pins will always be disabled, resulting in
no sound if there are no headphones connected.
This was reported as a no sound problem after suspend on
http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests
on first initialization before the phantom jack is added, but on resume
we reexecute the initialization code, and via_hp_automute starts
reporting HP always present with the jack now on the table.
BugLink: https://bugs.launchpad.net/bugs/1052499
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Cc: <stable@vger.kernel.org> [v3.6+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.
It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On the Thinkpad W520 - and probably several other machines with
Conexant 506x chips - the Dock Mic and Mic are connected to the
same two selector nodes. This patch will make Dock Mic take one
selector node and Mic take the other, when possible.
Without the patch, both paths would take the first selector,
leading to the normal Mic's volume being controlled by
"Dock Mic Boost".
(On other machines, this could instead fixup similar problems between
Mic and Line In, for example.)
BugLink: https://bugs.launchpad.net/bugs/1037642
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While going through Ubuntu bugs, I discovered this patch being
posted and a confirmation that the patch works as expected.
Finding out how the hw volume really works would be preferrable
to just disabling the broken one, but this would be better than
nothing.
Credit: sndfnsdfin (qawsnews)
BugLink: https://bugs.launchpad.net/bugs/559939
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Includes removal of duplicate debug print affirming entry into
the probe function, an unnecessary line break of a coding line
<80 chars and a white space change (unintentional tab).
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
We continue to allow the AB8500 CODEC to be registered via the AB8500
Multi Functional Device API, only this time we extract its configuration
from the Device Tree binary.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Register both parts of the MSP driver from Device Tree so that they
are probed when Device Tree is enabled. Also, as there is platform
data involved, we ensure that there is allocated memory to place the
configuration into and that the correct information is extracted from
the DT binary.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Here we ensure that the MOP500 audio driver will be probed during a
Device Tree boot. We also parse the sound node to link together the
codec, dma and the CPU-side Digital Audio Interface.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
In the initial submission of the MSP driver msp1 and msp3's associated
pinctrl mechanism was passed back to platform code using a plat_init()
call-back routine, but it has no place in platform code. The MSP driver
should set this up for the appropriate ports. Instead we use a use_pinctrl
identifier which is passed from platform_data/Device Tree which indicates
which ports should use pinctrl.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
The standard name (and what PulseAudio picks up) is "Dock Mic",
not "Docking Mic" or "Docking-Station".
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for USB suspend breakage moved the code to set up EP
from hw_params to prepare, but it means also the EP setup might be
called multiple times unnecessarily because the prepare callback can
be called multiple times without starting the stream (e.g. OSS
emulation).
This patch adds a new flag to struct snd_usb_substream indicating
whether the setup of EP is required, and do it only when necessary,
i.e. right after hw_params or suspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move interface and endpoint configuration from hw_params to prepare
callback. During system suspend/resume when the USB device power isn't
cycled the interface and endpoint configuration need to be set before
audio playback can continue. Resume involves another call to prepare
but not to hw_params, moving it here allows a playing stream to continue
after resume.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the interface to configure an endpoint so that it doesn't require
a hw_params struct. This will allow it to be called from prepare
instead of hw_params, configuring it after system resume.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the peiod_bytes member of snd_usb_substream. It was no longer being
set, but will be needed to resume properly in a future commit.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is more idiomatic as it means we verify that the device is there
prior to trying to do the card probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove unreferenced header files.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add missing dai_driver information to avoid these runtime errors
[ 16.433788] asoc: error - multiple DAI f0002c00.i2s registered with no name
[ 16.453551] Failed to register DAI
[ 16.461222] mpc5200-psc-i2s: probe of f0002c00.i2s failed with error -22
[ 16.475242] asoc: error - multiple DAI f0002000.ac97 registered with no name
[ 16.488087] mpc5200-psc-ac97 f0002000.ac97: Failed to register DAI
[ 16.502222] mpc5200-psc-ac97: probe of f0002000.ac97 failed with error -22
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mpc5200_psc_ac97 and mpc5200_psc_i2s modules rely on shared platform data
with mpc5200_dma.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When audmux_read_file is called, it means the driver is already
initialised successfully, so we don't need to check audmux_base.
It also fix smatch warning:
sound/soc/fsl/imx-audmux.c:78 audmux_read_file() warn: possible memory leak of 'buf'
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_imx_open should return error code returned by snd_dmaengine_pcm_open.
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PowerPC ASoC drivers frequently use the _BE variants of the SNDRV_PCM_FORMAT
macros, so we need to look for those as well.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove a call to dma_unmap_single() from the PowerPC ASoC DMA driver. The
buffer is allocated and not actually mapped, so the unmap call doesn't
make sense. It was probably left over from some early version of the driver.
This bug was unnoticed for so long because the DMA mapping functions normally
don't do anything on PowerPC.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_register_card() instead of platform_device_alloc("soc-audio")
to register the sound card from the machine drivers. The use of
platform_device_alloc is deprecated.
Although several other drivers still use platform_device_alloc(), the
Freescale drivers were not using it to pass driver data. Instead of fixing
the driver data usage, it's better to replace the deprecated code.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the input gain for the internal mic is set to its maximum level,
the background noise becomes so high - and any relevant signal clipped -
that the setting becomes unusable. It is better to limit the amplification.
BugLink: https://bugs.launchpad.net/bugs/1052460
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than including mach/iomux-mx27.h to define gpio numbers and set
up the pins, the patch moves all these into machine code and has the
gpio numbers passed to driver via platform_data. As the result, we
can remove the mach/iomux-mx27.h inclusion from driver.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()
added a new error check for input params.
this add new routine for input checks and moves buffer overflow check to this
new routine. This allows the error value to be propogated to user space
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bigger set of updates than I'm entirely comfortable with - things
backed up a bit due to travel. As ever the majority of these are small,
focused updates for specific drivers though there are a couple of core
changes. There's been good exposure in -next.
The AT91 patch fixes a build break.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.6
A bigger set of updates than I'm entirely comfortable with - things
backed up a bit due to travel. As ever the majority of these are small,
focused updates for specific drivers though there are a couple of core
changes. There's been good exposure in -next.
The AT91 patch fixes a build break.
Some of new HP laptops have a LED for microphone (or recording) mute,
and it's controlled by GPIO pin 3.
Bind this with the capture switch to turn it on/off properly by the
mixer change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These are 32 bit values that come from the user, we need to check for
integer overflows or we could end up allocating a smaller buffer than
expected.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the default value of position_fix -1, and allow user passing
position_fix=0 explicitly to set the "auto" position-fix mode.
Otherwise the auto mode may be switched to others like COMBO of
VIACOMBO when the controller prefers it, thus user can't set the auto
mode any longer.
Also updated the documentation appropriately, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the definitions of CS420X_IMAC27_122 and CS420X_APPLE as
aliases, the rest enums are set to duplicated values unexpectedly.
Move the alias definitions at the end so that the enum values are
defined in the proper order.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove useless kfree() and clean up code related to the removal.
The semantic patch that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
position p1,p2;
expression x;
@@
if (x@p1 == NULL) { ... kfree@p2(x); ... return ...; }
@unchanged exists@
position r.p1,r.p2;
expression e <= r.x,x,e1;
iterator I;
statement S;
@@
if (x@p1 == NULL) { ... when != I(x,...) S
when != e = e1
when != e += e1
when != e -= e1
when != ++e
when != --e
when != e++
when != e--
when != &e
kfree@p2(x); ... return ...; }
@ok depends on unchanged exists@
position any r.p1;
position r.p2;
expression x;
@@
... when != true x@p1 == NULL
kfree@p2(x);
@depends on !ok && unchanged@
position r.p2;
expression x;
@@
*kfree@p2(x);
// </smpl>
Signed-off-by: Peter Senna Tschudin <peter.senna@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core has for a long time had support for marking the register maps of
devices dirty when suspending so that they are resynced on resume. Also
implement this feature for CODECs using regmap.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For following the standard, define more channel map positions and
shuffle the items a bit:
- As both PulseAudio and gstreamer define MONO channel position
explicitly, we should follow that, too. The mono streams point to
this channel position unless they are explicitly assigned to certain
channel positions.
- Top-front-* and Top-rear-* positions are added, carried from
PulseAudio's definitions.
- Move NA and MONO definitions at the top of table right after
UNKNOWN, since these are more abstract in comparison with other
practical positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AK4396 DAC has a linear-scale attentuator, but
sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is
not quite right. This patch restores the correct scale, borrowing
from the ak4396 code in sound/pci/oxygen/oxygen.c.
Signed-off-by: Matteo Frigo <athena@fftw.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide channel maps for individual stereo streams of ENS1370 and
ENS1371. Note that the configuration of ENS1370 uses the secondary
PCM as the front unlike ENS1371.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is better style as we acquire resources we will need before we go into
the ASoC card probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Better style as we get all the resources we need prior to starting the
ASoC level probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MacBook Pro 10,1 needs a few adjustments to make it working:
- more COEF verbs
- some pin config overrides to disable the unused pin (0x0d, 0x12),
and fix the internal mic (0x0e)
In addition, it uses GPIO 1 and 3 like other MacBooks.
The internal digital mic on the machine is still problematic: it seems
that only the right channel is used and the left is always static.
This looks like a hardware design, so we need to cope in the software
side somehow...
The primary information and test were brought from Daniel J Blueman.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_pick_fixup() didn't check the case where the device mask bits
are set, typically used for SND_PCI_QUIRK_VENDOR() entries. Fix this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally the bogus period at BDL head was introduced as a workaround
for the mismatching position update at the period boundary, typically
seen on dmix. However, for applications like PulseAudio that don't
require period wake ups, this workaround is just superfluous. Thus
better to disable it when no_period_wakeup is given in hw_params.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c20c5a841c changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.
My hardware is Cougar Point which the commit log of
c20c5a841c mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.
Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always
false and it will never do compress capture. The test for O_WRONLY is
also slightly off. The original test would consider "->flags =
(O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid.
I've also removed the pr_err() because that could flood dmesg.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In general, mono streams have no dedicated speaker assignment, thus
they should be rather marked as UNKNOWN position.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
