Use the core to do I/O rather than directly calling the driver operations
in order to support further refactoring.
Signed-off-by: Mark Brown <broonie@linaro.org>
Move imx-audmux macro definitions to include/dt-bindings, so they can be
used for devicetree.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Gateway LT27 needs a fixup for the inverted digital mic.
Reported-by: "Nathanael D. Noblet" <nathanael@gnat.ca>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few driver specific fixes here plus one core fix for a memory
corruption issue in DAPM initialisation which could lead to crashes.
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Merge tag 'asoc-v3.11-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.11
A few driver specific fixes here plus one core fix for a memory
corruption issue in DAPM initialisation which could lead to crashes.
When users of SND_DMAENGINE_PCM are built as module, the config symbol
SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide DAPM for the device, ensuring operation with DAPM required by the
core and making it easier to hook up external hardware to it.
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide DAPM for the device, ensuring operation with DAPM required by the
core and making it easier to hook up external hardware to it.
Signed-off-by: Mark Brown <broonie@linaro.org>
The bindings do not carry any resources, as the module only registers
the ASoC platform driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
With the new dmaengine implementation, the filter_data parameter has
to be set earlier, from pxa_ssp_startup().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use snd_dmaengine_dai_dma_data for passing the dma parameters from
clients to the pxa pcm lib. This does no functional change, it's just an
intermedia step to migrate the pxa bits over to dmaengine.
The calculation of dcmd is a transition hack which will be removed again
in a later patch. It's just there to make the transition more readable.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The pxa ssp DAI acts as a user of a pxa ssp port, and needs an
appropriate 'port' phandle in DT to reference the upstream.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For the PXA DMA rework, we need the generic dmaengine implementation
that currently lives in sound/soc for standalone (non-ASoC) AC'97
support.
Move it to sound/core, and rename the Kconfig symbol.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove unneeded error handling on the result of a call to
platform_get_resource when the value is passed to devm_ioremap_resource.
A simplified version of the semantic patch that makes this change is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression pdev,res,n,e,e1;
expression ret != 0;
identifier l;
@@
- res = platform_get_resource(pdev, IORESOURCE_MEM, n);
... when != res
- if (res == NULL) { ... \(goto l;\|return ret;\) }
... when != res
+ res = platform_get_resource(pdev, IORESOURCE_MEM, n);
e = devm_ioremap_resource(e1, res);
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document
this in the DT binding. This enables the DT to include the Mic Jack in
the audio routing table, and hence enables capture of audio, in addition
to the previously-working playback.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Tegra30 I2S driver was writing the AHUB interface parameters to the
playback path register rather than the capture path register. This
caused the capture parameters not to be configured at all, so if
capturing using non-HW-default parameters (e.g. 16-bit stereo rather
than 8-bit mono) the audio would be corrupted.
With this fixed, audio capture from an analog microphone works correctly
on the Cardhu board.
Cc: stable@vger.kernel.org
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove unneeded error handling on the result of a call to
platform_get_resource when the value is passed to devm_ioremap_resource.
Move the call to platform_get_resource adjacent to the call to
devm_ioremap_resource to make the connection between them more clear.
A simplified version of the semantic patch that makes this change is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression pdev,res,n,e,e1;
expression ret != 0;
identifier l;
@@
- res = platform_get_resource(pdev, IORESOURCE_MEM, n);
... when != res
- if (res == NULL) { ... \(goto l;\|return ret;\) }
... when != res
+ res = platform_get_resource(pdev, IORESOURCE_MEM, n);
e = devm_ioremap_resource(e1, res);
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
The hardware does support synchronized start/pause/stop of pcm streams,
so there is no reason not to add that feature after more than ten years.
Some minor coding style / white space fixes in the surroundings of the
changes.
Signed-off-by: Knut Petersen <Knut_Petersen@t-online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Exynos5420 added support for I2S TDM mode. For this, there are some
register changes in the I2S controller. This patch adds the relevant
register changes to support I2S in normal mode. This patch adds a
quirk for TDM mode and if TDM mode is present all the relevent changes
will be applied.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Reviewed-by: Tomasz Figa <t.figa@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Samsung has different versions of I2S introduced in different
platforms. Each version has some new support added for multichannel,
secondary fifo, s/w reset control and internal mux for rclk src clk.
Each newly added change has a quirk. So this patch adds all the
required quirks as driver data and based on compatible string from
dtsi fetches the quirks.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Reviewed-by: Tomasz Figa <t.figa@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This does not fully map the power control available within the device
but it provides the hooks for routing signals through the device and
allows automatic management of the device low power mode.
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus
interfacing, for smartphones, tablets and other portable audio devices
based on the Arizona platform.
This patch adds the wm8997 CODEC driver.
[Fixed some interface churn from bitrot due to the patch not going via
the MFD tree as expected -- broonie]
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Ensure that the recently added path kcontrol list is initialised otherwise
we may crash trying to delete routes that don't have kcontrols.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Attempting to create the route as part of adding a mux control causes us
to attempt to add the same route twice since we loop over all sources
for the mux after creating the control. Instead do the addition in the
callers.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
The current generic parser code assumes that always a pin widget
controls the mute for an output blindly although it might be a
different widget in the middle. Instead of the fixed assumption,
check each parsed path and just pick up the right widget that has been
already defined as a mute control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute using the amp currently works only for a single amp on a
pin. Make it working also with HDA_CTL_BIND_MUTE type, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the volume control quirk for avoiding the kernel warning
for the Logitech HD Webcam C525
as in the similar commit 36691e1be6
for the Logitech HD Webcam C310.
Reported-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Tested-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Cc: <stable@vger.kernel.org> # 3.10.5+
Signed-off-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The routes linking the widget and the input mux were being added
manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds
the routes to the macro.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
Saves some code. We should also be able to manage the power up and reset
registers using DAPM but it's probably more trouble than it's worth in
mains powered systems.
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes it easer to integrate the device with other on-board components
and ensures correct operation following removal of support for non-DAPM
CODECs.
Signed-off-by: Mark Brown <broonie@linaro.org>
The GPIO manipulation done by this driver is never in atomic context so
we can use gpio_set_value_cansleep() and support GPIOs that can't be set
from atomic context.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
There is no need for the power down work to be done on a per CPU workqueue
especially considering the fairly long delay before powerdown.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Vinod Koul <vinod.koul@intel.com>
We've added a fake mute control (setting the amp volume to zero) for
CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but
this feature was overlooked in the generic parser implementation. Now
the driver lacks of mute controls on these codecs.
The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE
bits in each place checking the amp capabilities.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit aafe77cc45 (ALSA: usb-audio: add support for many Roland/Yamaha
devices) had several logic errors that prevented create_auto_midi_quirk
from enumerating any MIDI ports.
Reported-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Patch makes pcm buffers DMA-able by allocating each one separately.
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct the pins for a line-in and a headphone on LG LW25 laptop with
ALC880 codec. Other pins seem fine.
Reported-and-tested-by: Joonas Saarinen <jonskunator@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the
arity of the called function.
The semantic match that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e,f;
@@
f(...,
- ARRAY_AND_SIZE(e)
+ e,ARRAY_SIZE(e)
,...)
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The
only difference is that snd_soc_info_enum_double() is also able to handle stereo
controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext()
for the SOC_ENUM_EXT control's info callback allows us to remove
snd_soc_info_enum_ext().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info
callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The
snd_soc_info_volsw_ext() function has been unused ever since then, so remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The soc_pm_waitq waitqueue has been around as long as the ASoC framework
existed, but has never been used so far, so remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We don't actually use the "go" variable so it can be removed.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__initdata should be placed between the variable name and equal
sign for the variable to be placed in the intended section.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes it possible to hook up other devices in boards and is required
by removal of support for non-DAPM CODECs in the core.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
There is no need to pass constants via stack. The width may be explicitly
specified in the format.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver used to assume that the streaming endpoint's wMaxPacketSize
value would be an indication of how much data the endpoint expects or
sends, and compute the number of packets per URB using this value.
However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes,
while only about 88 or 44 bytes are be actually used. This discrepancy
would result in URBs with far too few packets, which would not work
correctly on the EHCI driver.
To get correct URBs, use wMaxPacketSize only as an upper limit on the
packet size.
Reported-by: James Stone <jamesmstone@gmail.com>
Tested-by: James Stone <jamesmstone@gmail.com>
Cc: <stable@vger.kernel.org> # 2.6.35+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes it easier to hook into boards and ensures the driver continues
to work with support for non-DAPM CODECs removed.
Signed-off-by: Mark Brown <broonie@linaro.org>
Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to
be DMA-able, which stack is not. Furthermore, transfer_buffer should not be
allocated as part of larger device structure because DMA coherency issues and
patch fixes this issue too.
Cc: stable@vger.kernel.org
Signed-off-by: Jussi Kivilinna <jussi.kivilinna@iki.fi>
Tested-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since non-DAPM devices are not going to be supported provide DAPM input
and output widgets and hook them up to the DAIs.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Commit 02502da45 (ASoC: imx-mc13783: Depend on ARCH_ARM) introduced 'ARCH_ARM'
as a dependency for SND_SOC_IMX_MC13783, but this is a non-existent symbol.
This makes the selection of SND_SOC_IMX_MC13783 to be impossible.
Use the correct 'ARM' symbol instead.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Beep Volume Min/Max was backwards.
Change to SOC_SONGLE_SX_TLV for correct volume representation
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@kernel.org
Description of the Asoc machine driver for an at91sam9x5 based board
with a wm8731 audio DAC. Wm8731 is clocked by a crystal and used as a
master on the SSC/I2S interface. Its connections are a headphone jack
and an Line input jack.
[Richard: this is based on an old patch from Nicolas that I forward
ported and reworked to use only device tree]
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There may be some platforms using fsl-ssi that do not have a DMA driver
with generic DMA bindings. So this patch adds support for the generic
DMA bindings, while still accepting the old "fsl,dma-events" property if
"dmas" is not found.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for non-dma pcm for imx platforms with imx-pcm-fiq support.
Instead of imx-pcm-audio, in this case imx-pcm-fiq-audio device is added
and the SIER flags are set differently.
We need imx-pcm-fiq for some boards that use an incompatible codec.
imx-pcm-fiq handles those codecs differently and allows to operate with
them. DMA is not possible because some data sent by the codecs, e.g.
wm9712, is not in the datastream. Also some data is mixed up in the
fifos, so that we need to sort them out manually.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Adds a function to parse a default port configuration from devicetree.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch cleanups empty functions on scu
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds BUSIF support for R-Car sound DMAEngine transfer.
The sound data will be transferred via FIFO which can cover blank time
which will happen when DMA channel is switching.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DMAEngine transfer on SSI.
But, it transfers sound data from memory to SSI directly
without using HPBIF at this time.
It will be updated soon
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car Sound driver will support DMA transfer in the future,
then, SSI/SRU/SRC will use it.
Current R-Car can't use soc-dmaengine-pcm.c since its DMAEngine
doesn't support dmaengine_prep_dma_cyclic(),
and SSI needs double plane transfer (which needs special submit) on DMAC.
This patch adds common DMAEngine method for it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is using struct rsnd_dai_platform_info
so that indicate sound DAI information (playback/capture SSI ID).
But, SSI settings were also required separately.
Thus, platform settings was very un-understandable.
This patch adds dai_id to SSI
settings, and removed rsnd_dai_platform_info.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The input OSR bits are specified differently for wm5110 than for current
revs of wm5102. This patch corrects support for this on wm5110.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It is unused and a leftover of the pre multi-component era.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It is unused and a leftover of the pre multi-component era.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The tlv320aic26 contains a embedded snd_soc_codec struct which is referenced in
the keyclick code. That struct is never initialized though, replace the embedded
struct with a pointer and use that in the keyclick code.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding
it) went too far and converted a DAPM control to use SOC_SINGLE_EXT()
which crashes. Revert that portion of the patch.
Signed-off-by: Mark Brown <broonie@linaro.org>
Prevent NULL dereference in snd_usb_add_endpoints(), when
alts is passed as NULL. In this case, WARN (since this is
a non-fatal bug) and return NULL ep. Call sites treat a NULL
return value as an error.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the quirks all apply to implicit feedback (the source endpoint
is always a data endpoint), there's no need to set and check
a flag for it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An implicit feedback endpoint can only be a capture source. The
consumer (sink) of the implicit feedback endpoint is therefore limited
to playback EPs.
Check if the target endpoint is a playback first and remove redundant
checks.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since implicit_fb is not changed, !implicit_fb will always
be true - it is set only after these checks.
Similarly, there's also no need to set it at the top of the function.
Change the type of implicit_fb to bool (more appropriate).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the sync endpoint currently takes up about half of set_format().
Move it to a dedicated function.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These really should be a single driver because they're fully integrated
in hardware. Make them so.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide a helper macro which includes the sum of all enable bits in
the playback control register. This simplifies the code a little.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add PCM1792A spi mode codec support. This version implements only
a subset of functionalities. Tested connect to a pandaboard ES
device and based on recently pcm1681 codec.
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC automatically creates snd_soc_dapm_dai_in and snd_soc_dapm_dai_out
widgets for DAI drivers, and adds them to the list. Later on, ASoC
creates automatic routes between these widgets and a widget with a
stream name.
We look for a snd_soc_dapm_dai_in or snd_soc_dapm_dai_out widget, and
use this to obtain the DAI structure. We then scan all widgets for
any with a stream name refering to either the capture or the playback
stream, and create routes.
If you have both a snd_soc_dapm_dai_in and a snd_soc_dapm_dai_out
referring to the same DAI structure, this ends up creating one set of
routes for the DAI for the snd_soc_dapm_dai_in widget, and a duplicated
set of routes for the snd_soc_dapm_dai_out widget.
Fix this by checking that the stream name for the widget matches the
DAI widget name.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
[Remaining patch from "ASoC: kirkwood: use devm_clk_get() for the
external clock" -- broonie]
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
We must init the delayed_work for codec-codec links
otherwise shutting down the DAI chain will fault when
calling flush_delayed_work_sync() on the linked DAI.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Powering down PLL before switching to a mode that does not use it
is a bad idea. It would cause the SGTL5000 be without internal
clock supply, especially on the I2C interface, which would make
subsequent access to it fail.
Thus, in case of not using PLL any longer, first set the mode
control, then power down PLL.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some devices have the problem that if a internal audio signal source is disabled
the output of the source becomes undefined or goes to a undesired state (E.g.
DAC output goes to ground instead of VMID). In this case it is necessary, in
order to avoid unwanted clicks and pops, to disable any mixer input the signal
feeds into or to active a mute control along the path to the output. Often it is
still desirable to expose the same mixer input control to userspace, so cerain
paths can sill be disabled manually. This means we can not use conventional DAPM
to manage the mixer input control. This patch implements a method for letting
DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable
the control if the path on which the control sits becomes inactive. Userspace
will then only see a cached copy of the controls state. Once DAPM powers the
path up again it will sync the userspace setting with the hardware and give
control back to userspace.
To implement this a new widget type is introduced. One widget of this type will
be created for each DAPM kcontrol which has the auto-disable feature enabled.
For each path that is controlled by the kcontrol the widget will be connected to
the source of that path. The new widget type behaves like a supply widget,
which means it will power up if one of its sinks are powered up and will only
power down if all of its sinks are powered down. In order to only have the mixer
input enabled when the source signal is valid the new widget type will be
disabled before all other widget types and only be enabled after all other
widget types.
E.g. consider the following simplified example. A DAC is connected to a mixer
and the mixer has a control to enable or disable the signal from the DAC.
+-------+
+-----+ | |
| DAC |-----[Ctrl]-| Mixer |
+-----+ : | |
| : +-------+
| :
+-------------+
| Ctrl widget |
+-------------+
If the control has the auto-disable feature enabled we'll create a widget for
the control. This widget is connected to the DAC as it is the source for the
mixer input. If the DAC powers up the control widget powers up and if the DAC
powers down the control widget is powered down. As long as the control widget
is powered down the hardware input control is kept disabled and if it is enabled
userspace can freely change the control's state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Merge these two structures together; nothing other than the I2S and
DMA driver makes use of struct kirkwood_dma_data, and it's not like
struct kirkwood_dma_data is really just used to convey DMA specific
data to the backend; it's more a general shared structure between the
two halves.
This will later allow kirkwood-dma.c and kirkwood-i2s.c to be merged
together.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
CODECs without any outputs now remain powered down, which means any
paths to these codecs also remain powered down.
Add an always-enabled output pin widget to the spdif transceiver codec.
This enables DAPM to correctly identify that the spdif transceiver is
in use when playback is enabled, which will then allow DAPM to power up
any links from the CPU DAI to the S/PDIF transceiver.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_hda_jack_set_gating_jack() call didn't work when
auto_{mute,mic} is suppressed because (1) am_entry is
not filled with nid of the mic pin. (2) The jacks are not
created (by snd_hda_jack_detect_enable_callback) before the
snd_hda_jack_set_gating_jack call.
Now we use the first input pin nid directly, and create the jack if it
doesn't exist yet.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fixup for Apple Mac Mini was lost during the adaption to the generic
parser because the fallback for the generic ID 8384:7680 was dropped,
and it resulted in the silence output (and maybe other problems).
Unfortunately, just adding the missing subsystem ID wasn't enough, in
this case. The subsystem ID of this machine is 0000:0100 (what Apple
thought...?), and since snd_hda_pick_fixup() doesn't take the vendor
id zero into account, the driver ignored this entry. Now it's fixed
to regard the vendor id zero as a valid value.
Reported-and-tested-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
list_first_entry() will always return a valid pointer, even if the list is
empty. So the check whether path is NULL will always be false. So we end up
calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points
right in the middle of the widget struct and by trying to modify the path the
widgets memory will become corrupted. Fix this by using list_emtpy() to check if
the widget doesn't have any paths.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
When calling krealloc for the kcontrol data the items in the path list that
point back to the head of the list will now point to freed memory, which causes
the list to become corrupted. To fix this, instead of resizing the whole data
struct, only resize the widget list.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SGTL5000 Capture Attenuate Switch (or "ADC Volume Range Reduction"
as it is called in the manual) is single bit only.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Reviewed-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When a sound capture/playback is terminated while a playback/capture
is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in
the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other
channel.
Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active
to prevent this.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Reviewed-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_cnew() can return NULL, so we should check the result before trying to
use it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
More for neatness than for any great utility. Really we shouldn't be
creating the child device at all, refactoring will follow.
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to hold on to the `codec' pointer. We can use the `dsp'
pointer and grab all the information we need from there. This
makes the parameters for the functions a bit more sane and idiomatic.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
PCM1681 can be controlled via I2C, SPI or in bootstrap mode (no control mode). This code add
support only for I2C mode.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A fix to make sure userspace knows when control writes have caused a
change in value, fixing some UIs, plus a few few driver fixes mainly
cleaning up issues from recent refactorings on less mainstream platforms.
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Merge tag 'asoc-v3.11-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.11
A fix to make sure userspace knows when control writes have caused a
change in value, fixing some UIs, plus a few few driver fixes mainly
cleaning up issues from recent refactorings on less mainstream platforms.
We should be allocating our buffers for the SPI transfers
from the DMA zone.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Just recently OMAP4 legacy boot support has been removed. No reason to keep
the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5
can boot in this mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In case of error, the function devm_ioremap_resource() returns ERR_PTR()
and never returns NULL. The NULL test in the return value check should be
replaced with IS_ERR(), and also remove the dev_err call to avoid redundant
error message.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building imx_v6_v7_defconfig with all audio drivers as modules results in
the folowing build error:
ERROR: "imx_pcm_fiq_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined!
ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined!
ERROR: "imx_pcm_fiq_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined!
ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined!
ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined!
ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined!
Fix this by allowing SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to be also
built as modules and by using 'IS_ENABLED' to cover the module case.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
If kzalloc() fails for `img' then we are going to leak the memory
for `out'. We are freeing the memory of all the tx/rx transfers
but the tx/rx buf pointers will be NULL if we drop out earlier.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Generally, S/PDIF supports 20bit and optional 24bit samples. Thus add these
two formats for the dummy codec drivers.
If one S/PDIF controller has its own limitation, its CPU DAI driver should
set the supported format by its own circumstance, since the soc-pcm driver
will use the intersection of cpu_dai's formats and codec_dai's formats.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a typo here so we end up using the old freed pointer instead of
the newly allocated one. (If the "n" is zero then the code works,
obviously).
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
d8b51c11ff [ASoC: ac97c: Use
module_platform_driver()] broke the build:
CC sound/soc/au1x/ac97c.o
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__initcall_" and "&" does not give a valid preprocessing token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__exitcall_" and "&" does not give a valid preprocessing token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token
/home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:334:31: warning: ‘au1xac97c_driver’ defined but not used [-Wunused-variable]
make[5]: *** [sound/soc/au1x/ac97c.o] Error 1
make[4]: *** [sound/soc/au1x] Error 2
make[3]: *** [sound/soc] Error 2
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The blackfin ac97 driver never defines nor uses a global ac97 struct. So remove
the extern declaration for it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver
fails with the following compile error:
sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’:
sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token
sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined
The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to
devm_gpio_request_one()").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Depending on the mclk (or crystal) selected, the wm8731 codec have some
constraints on its data sampling rates:
e.g. with a 12.288MHz or 18.432MHz crystal, the authorized rates are
8KHz, 32KHz, 48KHz and 96KHz.
Signed-off-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some reason, the DAC/ADCs are not being powered up when I try and
use the UDA1341 driver; this used to work. Looking back in the git
history, I don't see anything obvious which would cause this
regression.
However, from dumping the register writes, it seems that the codec is
powered down, and nothing calls set_bias_level to wake the codec up.
Moreover, this driver hasn't had DAPM support added to it, which
prevents platform drivers from taking advantage of DAPMs facilities.
So, let's add DAPM support to the driver.
As we move the power control for the DAC/ADC into DAPM, we no longer
need it in set_bias_level() - this function just becomes a way to
manipulate the power control and sync the register cache with the
hardware at the appropriate point.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Wait with updating the widgets power field until the changes are actually
written to the hardware in dapm_seq_run_coalesced(). This will allow us to query
the current hardware state between calling dapm_power_one_widget() and actually
writing the new power state to hardware.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_dapm_add_path() is similar to snd_soc_dapm_add_route() except that it
expects the pointer to the source and sink widgets instead of their names. This
allows us to simplify the case where we already have a pointer to widgets. (E.g.
as we have in snd_soc_dapm_link_dai_widgets()). snd_soc_dapm_add_route() will be
updated to just look up the widget and then use snd_soc_dapm_add_path() to
handle everything else.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code is limited to only setting or clearing a single bit in a
register to power a widget up or down. This patch extends the DAPM code to be
more flexible in that regard and allow widgets to use arbitrary values to be
used to put a widget in either on or off state.
Since the snd_soc_dapm_widget struct already contains a on_val and off_val field
no additional fields need to be added and in fact the invert field can even be
removed. Also the generated code is slightly smaller.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we store for each path which control (if any at all) is associated
with that control. But we are only ever interested in the reverse relationship,
i.e. we want to know all the paths a certain control is associated with. This is
currently implemented by always iterating over all paths. This patch updates the
code to keep a list for each control which contains all the paths that are
associated with that control. This improves the run time of e.g.
soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n
being the number of paths for the card) to O(1).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The 'value' field is really per control and not per widget. Currently it is only
used for virtual MUXes, which only have one control per widget. So in that case
there is not so much of a difference between whether it is stored per widget or
per control. Moving the 'value' field from the widget to the control will allow
us to use it also for cases where we have more than one control per widget. E.g.
for mixers with multiple input controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In preparation for adding additional per control data wrap all access to the
widget list in helper functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places.
This patch puts them into a common helper function. Having this encapsulated in
a helper function will also make it more easier to eventually change the data
layout of the kcontrol's private data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
DAPM operations are always performed on the card as a whole. Yet (primarily for
historic reasons) dapm_power_widgets() takes a DAPM context as its parameter.
The DAPM context is mainly used to look up a pointer to the card. The same is
true for a couple of functions that are being called from dapm_power_widgets().
This patch changes the signature of dapm_power_widgets() and a couple of related
functions to take a snd_soc_card instead of a snd_soc_dapm_context.
Some of the functions also use the DAPM's device to print error and debug
messages. This can be a bit confusing though since this means the messages for
all widgets, also those from other contexts, will be printed with that device.
The patch updates those cases to use the device of the widget's DAPM context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The update field of a DAPM context is only assigned while the card's dapm_mutex
is locked, the field is also cleared again while the mutex is stil locked. So
there will only ever be one DAPM context at a time with a non-NULL update field.
So it is safe to move the update field from the DAPM context struct to the card
struct. Doing so will allow further cleanups in this area.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make the code more generally applicable by refactoring so that the MCLK1
rate can be selected based on the compatible string provided by the device
tree, allowing use on other boards which have different rates or use other
information sources.
Signed-off-by: Mark Brown <broonie@linaro.org>
As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were
staticised meaning that the prototype (which was never needed) conflicts
with the declaration causing build failures.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
VAIO-Z laptops need to use the specific DAC for the speaker output
by some unknown reason although the codec itself supports the flexible
connection. So we implemented a workaround by a new flag,
no_primary_hp, for assigning the speaker pin first.
This worked until 3.8 kernel, but it got broken because the driver
learned for a better multi-io pin mapping, and not it can assign two
mic pins for multi-io. Since the multi-io requires to be the primary
output, the hp and two mic pins are assigned in prior to the speaker
in the end.
Although the machine has two mic pins, one of them is used as a noise-
canceling headphone, thus it's no real retaskable mic jack. Thus, at
best, we can disable the multi-io assignment and make the parser
behavior back to the state before the multi-io.
This patch adds again a new flag, no_multi_io, to indicate that the
device has no multi-io capability, and set it in the fixup for
VAIO-Z. The no_multi_io flag itself can be used generically, added
via a helper line, too.
Reported-by: Tormen <my.nl.abos@gmail.com>
Reported-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Get rid off the wm_coeff struct and the wm_coeff_add_kcontrol()
function. We are now using the snd_soc_card_kcontrol() function to
get the kcontrol pointers. No need to call into ALSA code to
register the kcontrols.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is useful for drivers who want to grab a pointer to
snd_kcontrol outside of the kcontrol callbacks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The current fixup for dell-bios model with STAC9228 codec contains the
override of pin 0x0c for analog mic. But this is actually just adding
a bogus pin and confuses the parser. Better to remove it for the
auto-mic switching.
Meanwhile, for a possible regression, keep the old configuration as
model=dell-bios-amic, so that people can test it again quickly.
Tested on Dell 1420n laptop.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With runtime power save feature enabled, Headphone hotplug
event will not be detected while controller/codec in D3. HDA has
feature WAKEEN to let codec wake up system if controller is in D3 or
system in S3.(HDA Spec 4.5.9.2/3). Codec can send out INT or wake up
controller depending on whether CIE or GIE enabled.(Figure 4, Interupt
structure).
The controller must be in RESET mode after enter runtime-suspend, otherwise
it will not be waken up even if codec send out wake-up event. And STATESTS
will be cleared after controller brought out of RESET mode.
This patch only enable WAKEEN for runtime-suspend(Controller D3) mode,
not for system S3 mode. with tool "evtest", Headphone hotplug events
could be cought and reported successfully.
[fixed an unused variable warning by tiwai]
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With jackpoll_interval != 0, it's used to poll jack event periodically
in a delayed work. if it's 0, give the caller chance to probe jack status
but will not restart the delayed work.
In the next patch which enable WAKEEN feature, HDA controller was able to wake
up system when it's in D3, it's useful to detect Jack hotplug event and notify
userspace. By default the jackpoll_interval=0, this patch let jack poll once
without starting the delayed work.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing jackpoll_interval before calling cancel_delayed_work_sync(),
otherwise the work will be triggered again and cause impact in
hda_jackpoll_work(). The next patch will poll jack once even with
jackpoll_interval=0.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test here is always true because S[i].urb is an array not a pointer.
Also it's bogus because the intent was to test:
if (S->urb[i]) {
instead of:
if (S[i].urb) {
Anyway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can
just remove this.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the
return value was never updated for this call
assign return value from put_user()
Reported-by: Haynes <hgeorge@codeaurora.org>
CC: stable@vger.kernel.org
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
As 1st protype, this patch adds SSI feature on this driver.
But, it is PIO sound playback support only at this point.
The DMA transfer, and capture feature will be supported in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds ADG feature which controls sound clock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds SCU feature on this driver.
But, it defines SCU style only, does nothing at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
The main difference between Gen1 and Gen2 are
1) register offset, 2) data path
In order to control Gen1/Gen2 by same method,
this patch adds gen.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
Gen1 series consists of SRU/SSI/ADG, and
Gen2 series consists of SCU/SSIU/SSI/ADG.
In order to control these by same method,
these are treated as "mod" on this driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuits are different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2).
(Actually, there are many difference in Generation1 chips)
Basically, for the future, Renesas R-Car series will use
Gen2 style sound circuit, but driver should care Gen1 also.
The main differences between Gen1 and Gen2 peripheral
are 1) register offset, 2) data path.
This patch adds basic (core) feature for R-Car
series sound driver as prototype
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use regmap for IO for the lm4857 driver instead of open-coding the IO read/write
functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let the ASoC core take care of registering the DAPM widget and routes as well as
the controls. This makes the code a bit shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
the compress metadata handlers were wrongly named sst_xxx
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move HW initialization to separate function to allow using the code without
the v4l parts. This is needed for use in the bttv driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and
not an unsigned long. The current tests are always false.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch modifies the i2s driver to give flexibility towards register
handling. This is a pre requirement for enabling i2s support on Exynos5420.
This patch modifies only the required registers as a pre-requirement to
support on Exynos5420.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let CPU DAI drivers set SDMA periperal type directly to support more
dma types(SPDIF, ESAI) other than only two for SSI.
This will easily allow some non-SSI drivers to use the imx-pcm-dma
as well.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_ioremap_resource does sanity checks on the given resource. No need to
duplicate this in the driver.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_ioremap_resource does sanity checks on the given resource. No need to
duplicate this in the driver.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_ioremap_resource does sanity checks on the given resource. No need to
duplicate this in the driver.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_ioremap_resource does sanity checks on the given resource. No need to
duplicate this in the driver.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_ioremap_resource does sanity checks on the given resource. No need to
duplicate this in the driver.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Register STATESTS is 16-bit length, use correct API for read/write.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use snd_soc_dapm_mixer_update_power() instead of reimplementing its
functionality.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to avoid race conditions the assignment of dapm->update should happen
while card->dapm_mutex is being held. To allow CODEC drivers to run a register
update when using snd_soc_dapm_mux_update_power() or
snd_soc_dapm_mixer_update_power() add a update parameter to these two functions.
The update parameter will be assigned to dapm->update while card->dapm_mutex is
locked.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when updating a control that is shared between multiple widgets the
whole power-up/power-down sequence is being run once for each widget. The
control register is updated during the first run, which means the CODEC internal
routing is also updated for all widgets during this first run. The input and
output paths for each widgets are only updated though during the respective run
for that widget. This leads to a slight inconsistency between the CODEC's
internal state and ASoC's state, which causes non optimal behavior in regard to
click and pop avoidance.
E.g. consider the following setup where two MUXs share the same control.
+------+
A1 ------| |
| MUX1 |----- C1
B1 ------| |
+------+
|
control ---+
|
+------+
A2 ------| |
| MUX2 |----- C2
B2 ------| |
+------+
If the control is updated to switch the MUXs from input A to input B with the
current code the power-up/power-down sequence will look like this:
Run soc_dapm_mux_update_power for MUX1
Power-down A1
Update MUXing
Power-up B1
Run soc_dapm_mux_update_power for MUX2
Power-down A2
(Update MUXing)
Power-up B2
Note that the second 'Update Muxing' is a no-op, since the register was already
updated.
While the preferred order for avoiding pops and clicks should be:
Run soc_dapm_mux_update_power for control
Power-down A1
Power-down A2
Update MUXing
Power-up B1
Power-up B2
This patch changes the behavior to the later by running the updates for all
widgets that the control is attached to at the same time.
The new code is also a bit simpler since callers of
soc_dapm_{mux,muxer}_update_power don't have to loop over each widget anymore
and neither do we need to keep track for which of the kcontrol's widgets the
current update is.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
As noticed by Lars-Peter Clausen since the move to using widgets to hook
into the DAIs we no longer directly manage the power of AIF or DAC/ADC
widgets from the stream integration so they can just use the generic power
checks instead of the custom stream integration ones they currently do.
Signed-off-by: Mark Brown <broonie@linaro.org>
The ALSA core expect the put callback of a control to return 1 if the value of
the control changed and 0 if it did not. Both snd_soc_dapm_put_volsw() and
snd_soc_dapm_put_enum_virt() currently always returns 0. For both functions we
already have a 'change' variable which either contains 1 or 0 depending on
whether the value has changed or not, so just return that.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the power state of ALC283 codec goes to D3, it gives a noise via
headphone output. This is because the driver tries to clear all pins
via snd_hda_shutup_pins(). Setting the mic pin to zero triggers such
a noise.
Define a new shutup call specific to this codec and control the pins
there more precisely. Also, add the power-save enable/disable
sequences in the resume and the new shutup calls.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_shutup_pins() unconditionally, allow it be
called in spec->shutup callback. In this way, we can avoid calling
this function if it causes a problem like we see in the next patch
following this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few small updates again, the sgtl5000 one fixes some newly triggered
issues due to some probe ordering changes which were introduced in the
last merge window.
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Merge tag 'asoc-v3.11-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.11
A few small updates again, the sgtl5000 one fixes some newly triggered
issues due to some probe ordering changes which were introduced in the
last merge window.
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of hiface, as expected by snd_pcm_update_hw_ptr0().
Caught by sparse.
Cc: Antonio Ospite <ospite@studenti.unina.it>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the adau1702 and the adau1401(a) are register compatible to the adau1701,
so add them to adau1701 driver's device id table.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9
("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due
to the dependency on the SigmaDSP firmware loader which was only available on a
limited set of platforms. This was fixed quite some time ago in commit 40216ce7
("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver
back again to SND_SOC_ALL_CODECS.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
These two functions were added two years ago in commit 4805608 ("ASoC: dapm -
Add methods to retrieve snd_card and soc_card from dapm context.") but have
remained unused so far. Considering that the dapm context actually has a direct
pointer to the card the functions also seem to be unnecessary. E.g. the
expressions 'dapm_get_soc_card(dapm)' and 'dapm->card' yield the same result.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The errors were caused by copy/paste mistake in below commit
since v3.10:
3489d50 ASoC: tegra: Use common DAI DMA data struct
It also corrects slave_id initialization in tegra20_ac97 driver.
Signed-off-by: Richard Zhao <rizhao@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lucas Stach <dev@lynxeye.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: <stable@vger.kernel.org> # 3.10
The SAIF driver is a clock provider so specifically needs the common
clock implementedation.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
There are three callers for this function, and none of them want it to
free platform for them. It leads to a double free.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
clk_prepare_enable() may fail, so let's check its return value and propagate it
in the case of error.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add SND_SOC_DAIFMT_CBM_CFS support for omap3/omap4. The patch was tested on
a pandaboard-es board connected to the pcm1792a codec. mcbspx_fsx must
configured as output and mcbspx_clkx must be configured as input.
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add MODULE_ALIAS, so that auto module loading can work.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Timur Tavi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The usage of strict_strto*() is not preferred, because
strict_strto*() is obsolete. Thus, kstrto*() should be
used.
Signed-off-by: Jingoo Han <jg1.han@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of 6fire, as expected by snd_pcm_update_hw_ptr0().
Caught by sparse.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_jack_detect() function returns a boolean value for a jack
plugged in or not, but it also returns always true when the
corresponding pin is phantom (i.e. fixed). This is OK in most cases,
but it makes the generic parser misbehaving about the auto-mute or
auto-mic switching, e.g. when one of headphone pins is a fixed.
Namely, the driver decides whether to mute the speaker or not, just
depending on the headphone plug state: if one of the headphone jacks
is seen as active, then the speaker is muted. Thus this will result
always in the muted speaker output.
So, the problem is the function returns a boolean, after all, although
we need to think of "phantom" jack. Now a new function,
snd_hda_jack_detect_state() is introduced to return these tristates.
The generic parser uses this function for checking the headphone or
mic jack states.
Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for
keeping compatibility in other driver codes.
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the build of this driver. It was broken by:
Commit 453807f300
ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params
The removed struct ep93xx_pcm_dma_params use the member 'dma_port' to
select the dma channel. The struct ep93xx_dma_data uses the member
'port'.
Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Cc: Ryan Mallon <rmallon@gmail.com>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add wm8904 based audio machine driver for Atmel EK board
The following ek board based on it
- at91sam9n12ek
- sama5d3xek (d31, d33, d34, d35)
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently after playing any audio file, we get the following error message:
$ aplay clarinet.wav
Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
$ wm8962 0-001a: Unsupported sysclk ratio 544
This error message appears about 5 seconds after the audio playback has
finished.
Quoting Mark Brown [1]:
"The issue here is triggered by the machine
switching from the FLL to direct MCLK usage where the MCLK isn't
generating a useful ratio.
I suspect we should just kill the configure_bclk() in set_sysclk(), that
one isn't safe as we can't reconfigure a live SYSCLK and it's probably
the one that generates your warnings."
Confirmed that the "Unsupported sysclk ratio" error message comes from
wm8962_set_dai_sysclk(), so get rid of wm8962_configure_bclk() inside this
function.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064241.html
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the ARM version by default as that's the more generally portable
one, it doesn't matter if they work well on random platforms when the
goal is only build coverage.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Timur Tabi <timur@tabi.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
The work used to allow the capcitors to ramp is not performance sensitive
so let the scheduler run it wherever is most efficient rather than in a
per CPU workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
None of the delayed work the driver schedules has particularly short delays
and it is not performance sensitive so let the scheduler run it wherever
is most efficient rather than in a per CPU workqueue by using the system
power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
None of the delayed work the driver schedules has particularly short delays
and it is not performance sensitive so let the scheduler run it wherever
is most efficient rather than in a per CPU workqueue by using the system
power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
The accessory detect debounce work is not performance sensitive so let
the scheduler run it wherever is most efficient rather than in a per CPU
workqueue by using the system power efficient workqueue.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
The quirk for Dell laptops with STAC9228 overrides the pin default
config of NID 0x0f to the value with AC_DEFCFG_MISC_NO_PRESENCE bit
on. I'm not quite sure why this was done so, but can guess that this
was introduced for avoiding this to be muted by another headphone
plug. Now, after transition to the generic parser, this workaround
rather causes a problem (notably as unexpected speaker mutes) because
the pin is seen as if it's always plugged in.
Since the generic parser can handle multiple headphone plugging
gracefully, we can get rid of this override now.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The EAPD GPIO is dynamically turned on/off for some machines with
Sigmatel codecs, but this didn't work as expected, and it resulted in
spontaneous lost of speaker outputs per HP plugging or power-saving.
This patch fixes the bug by simply including spec->eapd_mask into
spec->gpio_mask and spec->gpio_data bits.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the 'failed to find codec platform device' error path is executed, it should
jump to 'fail' label instead of returning an error immediately.
'fail' label will then free the ssi_np and codec_np previously acquired nodes.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no need to use a normal per-CPU workqueue for delayed power downs
as they're not timing or performance critical and waking up a core for them
would defeat some of the point.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Viresh Kumar <viresh.kumar@linaro.org>
This fixes the following by deleting dead code:
sound/soc/tegra/tegra20_ac97.c: In function ‘tegra20_ac97_platform_probe’:
sound/soc/tegra/tegra20_ac97.c:435:1: warning: label ‘err_unregister_pcm’ defined but not used [-Wunused-label]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there is no architecture dependency in the code allow it to be built
on any platform when COMPILE_TEST is enabled.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Stephen Warren <swarren@nvidia.com>
The usage of the dmaengine helpers is unconditional, especially when
doing compile testing.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
clk_prepare_enable() may fail, so let's check its return value and propagate it
in the case of error.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_ functions can make the code cleaner and smaller.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Devices like mobilephones, computers are typically used as
HDMI sources, but devices like TV, navigators will be HDMI
sinks.
for auto scenerios, In-Vehicle Infotainment(IVI) can be HDMI
sink to display movies from mobilephones.
Signed-off-by: Rongjun Ying <Rongjun.Ying@csr.com>
Signed-off-by: Barry Song <Baohua.Song@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DT platforms don't have any source dependency on any OMAP stuff so
allow them to be built when COMPILE_TEST is enabled.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
We've got bug reports that the module loading stuck on Debian system
with 3.10 kernel. The debugging session revealed that the initial
registration of OSS sequencer clients stuck at module loading time,
which involves again with request_module() at the init phase. This is
triggered only by special --install stuff Debian is using, but it's
still not good to have such loops.
As a workaround, call the registration part asynchronously. This is a
better approach irrespective of the hang fix, in anyway.
Reported-and-tested-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's not always the case that clock is already available when sgtl5000
get probed at the first time, e.g. the clock is provided by CPU DAI
which may be probed after sgtl5000. So let's defer the probe when
devm_clk_get() call fails and give it chance to try later.
It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000:
Let the codec acquire its clock).
[ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2
[ 1.934280] sgtl5000: probe of 0-000a failed with error -2
[ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered
[ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517)
[ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix to return a negative error code from the error handling
case instead of 0, as done elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since DT based boards don't have any dependency on arch/arm enable them
if COMPILE_TEST is enabled.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
The driver supports a range of devices but currently doesn't allow those
device names to be used for enumeration on DT. Add the currently listed
I2C IDs as compatible strings.
Signed-off-by: Mark Brown <broonie@linaro.org>
In the function kirkwood_set_rate, when the rate cannot be satisfied
by the internal nor by an external clock, the clock source in undefined:
warning: ‘clks_ctrl’ may be used uninitialized in this function
The ALSA subsystem should never gives such a rate because:
- the rates with the internal clock are limited to 44.1, 48 and 96 kHz
as specified by the kirkwood_i2s_dai structure,
- the other rates are proposed in the structure kirkwood_i2s_dai_extclk
only when the external clock is present.
In case of programming error (bad rate for internal clock and no
external clock), the function will simply cause a backtrace.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
This Conexant codec has a single jack that can be used as either
headphone or mic (but not headset). The existing hp_mic functionality
does not apply here, because the mic and the HP are on separate pins.
Hence make a lighter version of what has been earlier done for Realtek
codecs.
BugLink: https://bugs.launchpad.net/bugs/1198030
Tested-by: Franz Hsieh <franz.hsieh@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The addition of extclk support makes this misleading as it's only the
rates used when there is no extclk so put it in the specific DAI it
applies to.
Signed-off-by: Mark Brown <broonie@linaro.org>
Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip.
Reviewed-by: Andy Ritger <aritger@nvidia.com>
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_stop() must be called in the PCM substream lock context.
Cc: <stable@vger.kernel.org>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_stop() must be called in the PCM substream lock context.
Cc: <stable@vger.kernel.org>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_stop() must be called in the PCM substream lock context.
Cc: <stable@vger.kernel.org>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
