There are some place in the ASoC core that expect rtd->codec to be non NULL
(mainly CODEC specific sysfs files). With componentization going forward
rtd->codec might be NULL in some cases. This patch prepares the core for this by
not registering CODEC specific sysfs files if rtd->codec is NULL. sysfs file
removal does not need to be conditionalized as it handles the removal of
non-existing files just fine.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform and CODEC probe and remove code is now largely identical. This
patch consolidates it at the component level.
The resulting code is slightly larger due to all the boiler plate code setting
up the indirection for the table based control and DAPM registration. Once all
drivers have been update to no longer use the snd_soc_codec_driver and
snd_soc_platform_driver specific fields for this the indirection can be removed
again.
This patch contains two noteworthy hacks that are only meant to be temporary to
be able to update drivers and the core in separate incremental patches.
The first hack is related to that some DPCM platforms expect that the DAPM
widgets for the DAIs of a snd_soc_component are created in the DAPM context of
the snd_soc_platform that has the same parent device. For handling this the
steal_sibling_dai_widgets attribute is introduced. It gets set for
snd_soc_platforms that register DAPM elements. When creating the DAI widgets for
a component this flag is checked and if it is found on one of the siblings the
component will not create any DAI widgets in its own DAPM context. If the
attribute is set on a platform it will look for siblings components and create
DAI widgets for them in its own context. The fix for this will be to update
the offending drivers to only register a single component rather than two.
The second hack deals with the fact that the ASoC card suspend and resume code
still needs a list of CODECs that have been registered for the card. To handle
this the generic probe and remove path have a check to see if the component is
CODEC and if yes add/remove it to the card's CODEC list. While it is possible to
clean up the suspend/resume code to not need the CODEC list anymore this is a
bit of a chicken and egg problem since it will become easier to clean up the
suspend/resume code once there is a unified component layer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The debugfs registration is mostly identical between platforms and CODECs. This
patches consolidates the two implementations at the component level.
Unfortunately there are still a couple of CODEC specific debugfs files that are
related to legacy ASoC IO that need to be registered. For this a new callback is
added to the component struct that will be initialized when a CODEC is
registered and will be used to register the CODEC specific files. Once there are
no drivers left using legacy IO this can be removed again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The imx-es8328 driver fails to build on PPC because it explicitly depends on
SND_SOC_IMX_PCM_FIQ, which itself doesn't build on PPC. Instead, rely on
the SND_SOC_FSL_SSI config option to pull in the necessary libraries.
While we're at it, remove SND_SOC_FSL_UTILS, which also is not needed.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Valleyview and Cherryview have the same behavior on display audio. So this patch
defines is_valleyview_plus() to include codecs for both Valleyview and its successor
Cherryview, and apply Valleyview fix-ups to Cherryview.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both Haswell and Broadwell need set depop_delay to 0. So apply this
setting to haswell plus.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many codes doing writes or updates COEF verbs sequentially
in a batch. Rewrite such open codes with tables for optimization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The helper functions snd_hda_check_board_config() and
snd_hda_check_board_codec_sid_config() are no longer used since the
transition to the generic parser and all quirks have been replaced
with fixups. Let's kill these dead codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some HP laptops, the mute led is controlled by codec gpio.
When some machine resume from s3/s4, the codec gpio data will be
cleared to 0 by BIOS:
Before suspend:
IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0
After resume:
IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0
To skip the AFG node to enter D3 can't fix this problem.
A workaround is to restore the gpio data when the system resume
back from s3/s4. It is safe even on the machines without this
problem.
BugLink: https://bugs.launchpad.net/bugs/1358116
Tested-by: Franz Hsieh <franz.hsieh@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/soc/generic/simple-card.c: In function simple_card_dai_link_of:
sound/soc/generic/simple-card.c:198:10: warning: passing argument 3 of
asoc_simple_card_sub_parse_of from incompatible pointer type [enabled by default]
&dai_link->cpu_dai_name);
^
sound/soc/generic/simple-card.c:112:1: note: expected const struct device_node **
but argument is of type struct device_node **
asoc_simple_card_sub_parse_of(struct device_node *np,
^
sound/soc/generic/simple-card.c:229:10: warning: passing argument 3 of
asoc_simple_card_sub_parse_of from incompatible pointer type [enabled by default]
&dai_link->codec_dai_name);
^
sound/soc/generic/simple-card.c:112:1: note: expected const struct device_node **
but argument is of type struct device_node **
asoc_simple_card_sub_parse_of(struct device_node *np,
^
Since the asoc_simple_card_sub_parse_of() is used in simple-card module only,
and the third argument is just used to get the node ponters address, so there is
no need it must to be 'const' type.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building kernel with SND_SOC_IMX_AUDMUX=n leads to the following error:
sound/built-in.o: In function `fsl_asoc_card_probe':
>> fsl-asoc-card.c:(.text+0x1467b5): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467d0): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467ed): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x146807): undefined reference to `imx_audmux_v2_configure_port'
Update Kconfig to select SND_SOC_IMX_AUDMUX when SND_SOC_FSL_ASOC_CARD=y.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The AB8500_GPIO_DIR4_REG register define has the bank for the register in the
upper 8 bits and the register itself in the lower 8 bits. When passing it to
abx500_{set,get}_register_interruptible() the upper bits get truncated which
generates the following warning from sparse:
sound/soc/codecs/ab8500-codec.c:1972:53: warning: cast truncates bits
from constant value (1013 becomes 13)
sound/soc/codecs/ab8500-codec.c:1980:53: warning: cast truncates bits
from constant value (1013 becomes 13)
The bank is passed separately to abx500_{set,get}_register_interruptible() so
the code works fine as it is. Given that all users of AB8500_GPIO_DIR4_REG
always truncate the upper 8 bits just remove them from the define.
Also remove the unnecessary casts to u8.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
It is not always possible to interchange iomem pointers with normal pointers,
which why we have annotations for iomem pointers and warn when casting them to a
normal pointer or vice versa. In this case the casting is fine and unfortunately
necessary so add the proper annotations to tell code checkers that it is
intentional. This silences the following warnings from sparse:
sound/soc/samsung/idma.c:354:20: warning: incorrect type in argument 1
(different address spaces) expected void volatile [noderef]
<asn:2>*addr got unsigned char *area
sound/soc/samsung/idma.c:372:22: warning: cast removes address space of
expression
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.
Fixes the following warnings from sparse:
sound/soc/sh/fsi.c:1307:42: warning: mixing different enum types
sound/soc/sh/fsi.c:1307:42: int enum dma_data_direction versus
sound/soc/sh/fsi.c:1307:42: int enum dma_transfer_direction
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Sparse spits out the following warning:
sound/soc/sh/rcar/gen.c:250:21: warning: dubious: x & !y
It does this because sometimes mixing boolean and bit-wise logic has not the
intended result. In this case we are fine, but replacing the bit-wise '&' with
the boolean '&&' silences the sparse warning. The generated code for both cases
is the same.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
odroidx2_drvdata and odroidu3_drvdata are not used outside this module so make
them static (and also const while we are at it).
Fixes the following warnings from sparse:
sound/soc/samsung/odroidx2_max98090.c:69:26: warning: symbol
'odroidx2_drvdata' was not declared. Should it be static?
sound/soc/samsung/odroidx2_max98090.c:74:26: warning: symbol
'odroidu3_drvdata' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
edma_pcm_platform_register() is declared in edma-pcm.h and defined in
edma-pcm.c. To make sure that the function signature matches for both
edma-pcm.c should include edma-pcm.h
Fixes the following sparse warning:
sound/soc/davinci/edma-pcm.c:48:5: warning: symbol
'edma_pcm_platform_register' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
init_name is basically a hack and should only be used for statically allocated
device structs. For dynamically allocated devices dev_set_name() should be used.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This adds an initial machine driver for the ES8328 audio codec on Freescale
boards. The driver supports headphones and an audio regulator for an onboard
speaker amp.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a codec driver for the Everest ES8328. It supports two separate audio
outputs and two separate audio inputs.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current code only allows direct routing of the WM8994 microphone
detection signal to a GPIO this change adds support to demux the
interrupt from the main interrupt line of the codec.
Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add DAPM calls to enable/disable the Class D amp.
Also add a DAPM call to turn off the PLL upon
the stream completing.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Of_node_put supports NULL as its argument, so the initial test is not
necessary.
Suggested by Uwe Kleine-König.
The semantic patch that fixes this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e;
@@
-if (e)
of_node_put(e);
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
sst_ops need to use the sst driver context. So pass sst device as argument,
which can be used to retrieve sst context.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The previous patch (ASoC: fsl_sai: Add asynchronous mode support) added
new Device Tree bindings for Asynchronous and Synchronous modes support.
However, these two shall not be present at the same time.
So this patch just simply makes them exclusive so as to avoid incorrect
Device Tree binding usage.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SAI supports these operation modes:
1) asynchronous mode
Both Tx and Rx are set to be asynchronous.
2) synchronous mode (Rx sync with Tx)
Tx is set to be asynchronous, Rx is set to be synchronous.
3) synchronous mode (Tx sync with Rx)
Rx is set to be asynchronous, Tx is set to be synchronous.
4) synchronous mode (Tx/Rx sync with another SAI's Tx)
5) synchronous mode (Tx/Rx sync with another SAI's Rx)
* 4) and 5) are beyond this patch because they are related with another SAI.
As the initial version of this SAI driver, it supported 2) as default while
the others were totally missing.
So this patch just adds supports for 1) and 3).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds software reset code in dai_probe() so as to make a true init
by clearing SAI's internal logic, including the bit clock generation, status
flags, and FIFO pointers.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that
can be used, ideally, for all Freescale CPU DAI drivers and external CODECs.
The idea of this generic sound card is a bit like ASoC Simple Card. However,
for Freescale SoCs (especially those released in recent years), most of them
have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
this is a specific feature that might be painstakingly controlled and merged
into the Simple Card driver.
So having this driver will allow all Freescale SoC users to benefit from the
simplification to support a new card and the capability of wide sample rates
support through ASRC.
The driver is initially designed for sound card using I2S or PCM DAI formats.
However, it's also possible to merge those non-I2S/PCM type sound cards, such
as S/PDIF audio and HDMI audio, into this card as long as the merge will not
break the original function and as long as there is something redundant that
can be abstracted along with I2S type sound cards.
As an initial version, it only supports three cards that I can test:
imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC
imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver
imx-audio-wm8962, just like the old imx-wm8962.c driver
The driver is also compatible with the old Device Tree bindings of WM8962 and
SGTL5000. So we may consider to remove those two drivers after this driver is
totally enabled. (It needs to be added into defconfig)
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Forgot to add a return for err_disable goto statement.
Causes compile warning of control reaching end of non-void
Signed-off-by: Brian Austin <briann.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic CS35L32 Boosted Amplifier
I2S output provides monitor data to the SOC/CODEC/DSP for speaker protection/enhancement algorithms
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALC269 & co have many vendor-specific setups with COEF verbs.
However, some verbs seem specific to some codec versions and they
result in the codec stalling. Typically, such a case can be avoided
by checking the return value from reading a COEF. If the return value
is -1, it implies that the COEF is invalid, thus it shouldn't be
written.
This patch adds the invalid COEF checks in appropriate places
accessing ALC269 and its variants. The patch actually fixes the
resume problem on Acer AO725 laptop.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Tested-by: Francesco Muzio <muziofg@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v3.17
Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.
Pull sound fixes from Takashi Iwai:
"Here is the additional fix patches that have been queued up since the
previous pull request. A few HD-audio fixes, a USB-audio quirk
addition, and a couple of trivial cleanup for the legacy OSS codes"
* tag 'sound-fix-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Set TLV_DB_SCALE_MUTE bit for cx5051 vmaster
ALSA: hda/ca0132 - Don't try loading firmware at resume when already failed
ALSA: hda - Fix pop noises on reboot for Dell XPS 13 9333
ALSA: hda - Set internal mic as default input source on Dell XPS 13 9333
ALSA: usb-audio: fix BOSS ME-25 MIDI regression
ALSA: hda - Fix parsing of CMI8888 codec
ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controller
ALSA: hda/realtek - Fixed ALC286/ALC288 recording delay for Headset Mic
sound: oss: Remove typedefs wanc_info and wavnc_port_info
sound: oss: uart401: Remove typedef uart401_devc
