The Teclast X98+ II is a Cherrytrail tablet, which require two quirks:
- it has stereo speakers,
- its jack detection mechanism is inverted.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk to support boards whose jack detection mechanism is
inverted.
It will set the 'everest,jack-detect-inverted' boolean device property
for the es8316 codec driver.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes the sai driver structure overwriting which results in
a cpu dai name equal NULL.
Fixes: 3e086ed ("ASoC: stm32: add SAI driver")
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The control values and texts of the enum kcontrol associated
with a widget need to be freed when the widget is removed.
However, both struct snd_soc_dapm_widget and struct soc_enum
contain a dobj member, which resulted in a confusion.
The existing code generates a null pointer dereference by
attempting to free the values and texts from the dobj which
belongs to the widget instead of the dobj belonging to the
enum kcontrol.
The suggested fix is to use the correct dobj member (se->dobj)
of the enum kcontrol.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When ioctl calls are made with non-null-terminated userspace strings,
strlcpy causes an OOB-read from within strlen. Fix by changing to use
strscpy instead.
Signed-off-by: Zubin Mithra <zsm@chromium.org>
Reviewed-by: Guenter Roeck <groeck@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use SND_SOC_DAPM_SUPPLY for mic bias DAPM
instead of deprecated SND_SOC_DAPM_MICBIAS.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In ESAI synchronous mode, the clock is generated by Tx, So
we should always set registers of Tx which relate with the
bit clock and frame clock generation (TCCR, TCR, ECR), even
there is only Rx is working.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use an explicit define to avoid Sparse issues coming from the use of
cpu_to_be32
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use le16/32/64_to_cpu() as needed to make Sparse happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The axg tdmout driver just need a different skew offset to operate
correctly on the g12a SoC family.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The g12a tdmout requires a different signal skew offset than the axg.
With this change, the skew offset is added as a parameter of the tdm
formatters to prepare the addition of the g12a support.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the axg, frddr could only be connected to 1 downstream element, so the
playback was possible on 1 interface only at a time.
On the g12a, the frddr may connect and wait for the request of up to 3
downstream elements. With this, it possible for single playback to be
played on several interfaces at the same time.
Like the toddr fifo, the g12a frddr also need to take care of resetting
the read pointer to the initial fifo address when preparing a playback.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the g12a SoC fifo can set the fifo initial start address, we must
make sure to actually reset the write pointer to this address when
starting a capture.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The g12a fifos gained the ability to set the initial address of the
pointer within the buffer, instead of defaulting to the buffer start
address.
It is not very useful to us (yet) but we need to put a copy the buffer
start address in the related register for the fifo to work properly on the
g12a SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the card is registered by the machine driver,
dai link components are probed after the snd_card is
created. This is done in snd_soc_bind_card() which calls
snd_soc_instantiate_card() to first create the snd_card
and then probes the link components by calling
soc_probe_link_components(). The snd_card is used by the
component driver to add the kcontrols associated
with dapm widgets to the card.
When the machine driver is unregistered, the snd_card
is freed when the card resources are cleaned up.
But the snd_card needs to be valid while unloading the
topology dapm widgets in order to remove the kcontrols
from the card.
Since, unloading topology is done when the component
driver is removed, the link components should be removed
in snd_soc_unbind_card(). This will ensure that the kcontrols
are removed before the card resources are cleaned up and
the snd_card itself is freed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology is not unloaded in the core during unregister_component()
anymore. So, add the remove() callback that will unload the
topology.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The chips main power supplies VA and VP are enabled during probe but
then never disabled, this will cause warnings from the regulator
framework on driver removal. Fix this by adding a remove callback and
disabling the supplies, whilst doing so follow best practice and put the
chip back into reset as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sleep PM callbacks to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support of master mode for cs42l51 cirrus audio codec.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add cs42l51 audio codec power supply management
through regulator framework.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is unsafe to call snd_compr_stop_error from outside of the
compressed ops. Firstly the compressed device lock needs to be held
and secondly it queues error work to issue a trigger stop which
should not happen after the stream has been freed. To avoid these
issues use the same trick used for the IRQ handling, simply send a
snd_compr_fragment_elapsed to cause user-space to wake on the poll,
then report the error when user-space issues the pointer request
after it wakes.
Fixes: a2bcbc1b9a ("ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@kernel.org
According the publicly available datasheet (and some test) the max98357a
also supports 32, 44.1 and 88.2 kHz sample rate.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PowerTune controls the power level of the chip. On playback this
indirectly controls things like the gain of the various output
amplifiers. This can allow for the decrease of output levels
from the codec. This adds controls for those power levels to
the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a switch for setting common mode voltage. This can allow
for higher drive levels on the amplifier outputs.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch marks RXFIFO_DATA as precious to avoid being read
outside a call from the driver, such as regmap debugfs
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two identical spelling mistakes in dev_err messages. Fix them.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Reviewed-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fix the wrong reg value for rk322x/rk322xh,
cuz there is no STORE JUSTIFIED MODE on it.
on rk322x/rk322xh, the same bit means PDM_MODE/RESERVED,
if the bit is set to RESERVED, the controller will not work.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch make the waterlevel more reasonable, because the pdm
controller share the single FIFO(128 entries) with each channel.
adjust waterlevel in frame to meet the vad or dma frames request.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for rk1808, the pdm controller
is the same as rk3308.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support fractional div for rk3308.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to reset controller every time, do this
once in pdm_probe.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add default regs value for controller.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch set left justified store mode default.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch decreases the transfer bursts to avoid the fifo overrun.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is because set_fmt ops maybe called when PD is off,
and in such case, regmap_ops will lead system hang.
enale PD before doing regmap_ops.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit da215354eb ("ASoC: simple-card: merge simple-scu-card")
merged simple-scu-audio-card which can handle DPCM into
simple-audio-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its CPU/Codec DAI count.
But, because of it, existing "simple-audio-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "simple-audio-card" user
can select "normal sound card", and "simple-scu-audio-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Fixes: da215354eb ("ASoC: simple-card: merge simple-scu-card")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit ae3cb57909 ("ASoC: audio-graph-card: merge
audio-graph-scu-card") merged audio-graph-scu-card which can
handle DPCM into audio-graph-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its OF-graph endpoint connection.
But, because of it, existing "audio-graph-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "audio-graph-card" user
can select "normal sound card", and "audio-graph-scu-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Fixes: ae3cb57909 ("ASoC: audio-graph-card: merge audio-graph-scu-card")
Reported-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the copyright dates and use the SPDX identifier instead
of reciting the license.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the unnecessary validation of the 'cstream' variable to fix
below smatch warning:
sprd_platform_compr_drain_notify() warn: variable dereferenced
before check 'cstream' (see line 105)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The author of these files has changed her name. Update
instances in the code of her dead name to current legal
name.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509.
The Tuxedo XC 1509 and the System76 oryp5 are the same barebone
notebooks manufactured by Clevo. To name the fixups both use after the
actual underlying hardware, this patch also changes System76_orpy5
to clevo_pb51ed in 2 enum symbols and one function name,
matching the other pci_quirk entries which are also named after the
device ODM.
Fixes: 7f665b1c32 ("ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5")
Signed-off-by: Richard Sailer <rs@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Just Implement of reset combo jack JD verb for ACT_PRE_PROBE state.
Intel test result was also failed.
It test passed until changed the initial state to ACT_INIT.
Mic JD will show every time.
This patch also changed the model name as 'alc-chrome-book' for
application of Chrome OS.
Fixes: 10f5b1b85e ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Suggested-by: Baolin Wang <baolin.wang@linaro.org>
Tested-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Users have been seeing sound stability issues with max98090 codecs since:
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
At first that commit broke sound for Chromebook Swanky and Clapper models,
the problem was that the machine-driver has been controlling the wrong
clock on those models since support for them was added. This was hidden by
clk-pmc-atom.c keeping the actual clk on unconditionally.
With the machine-driver controlling the proper clock, sound works again
but we are seeing bug reports describing it as: low volume,
"sounds like played at 10x speed" and instable.
When these issues are hit the following message is seen in dmesg:
"max98090 i2c-193C9890:00: PLL unlocked".
Attempts have been made to fix this by inserting a delay between enabling
the clk and enabling and checking the pll, but this has not helped.
It seems that at least on boards which use pmc_plt_clk_0 as clock,
if we ever disable the clk, the pll looses its lock and after that we get
various issues.
This commit fixes this by enabling the clock once at probe time on
these boards. In essence this restores the old behavior of clk-pmc-atom.c
always keeping the clk on on these boards.
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-by: Mogens Jensen <mogens-jensen@protonmail.com>
Reported-by: Dean Wallace <duffydack73@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m
gcc warn this:
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function mt8183_da7219_max98357_dev_probe:
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms?
dai_link->platform = NULL;
^~~~~~~~
platforms
use 'dai_link->platforms' instead of 'dai_link->platform'.
Fixes: 11c0269017 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=m the
following error pops up:
../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function ‘mt8183_mt6358_ts3a227_max98357_dev_probe’:
../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: ‘struct snd_soc_dai_link’ has no member named ‘platform’; did you mean ‘platforms’?
dai_link->platform = NULL;
^~~~~~~~
platforms
Rework to use 'dai_link->platforms' instead of 'dai_link->platform'.
Fixes: 11c0269017 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227")
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tidy up some instances of dereferencing to obtain things that are
already stored in local variables.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm_adsp_compr_detach is NULL aware so there is no need to check for NULL
before calling it, remove the redundant check.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trigger stop can be called in situations where trigger start failed
and as such it can't be assumed the buffer is already attached to
the compressed stream or a NULL pointer may be dereferenced.
Fixes: 639e5eb3c7 ("ASoC: wm_adsp: Correct handling of compressed streams that restart")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer TravelMate B114-21 laptop cannot detect and record sound from
headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb
quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix
this issue.
[ fixed the missing brace and reordered the entry -- tiwai ]
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Cc: Dan Carpenter <dan.carpenter@oracle.com> (commit_signer:1/11=9%,authored:1/11=9%)
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This aligns all kcontrol tplg pointer increments to be consistent
in the respective create methods and ensures that the position is
pointing to the next widget rather the current invalid widget.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the same machine driver is reused between platforms but with a
different alias, using the driver name is not enough. Add additional
fallback case to use the card device name.
Tested on GeminiLake with bxt_da7219_max98357a machine driver
Suggested-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We use 2-stage DMA mode to support Spreadtrum audio compress offload,
which means we use one DMA source channel to transfer data from IRAM
buffer to the DSP fifo to do decoding/encoding, once IRAM buffer is
empty by transferring done, another DMA destination channel will be
triggered automatically to start to transfer data from DDR buffer to
the IRAM buffer. This can reduce the AP subsystem wakeup times to save
power.
Co-developed-by: Yintang Ren <yintang.ren@unisoc.com>
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, buffers, schedulers, src's, encoders, decoders
and effect type dapm widgets remain always on as their
power_check method is not set. Setting this callback allows these
widgets in the audio path to be powered managed properly.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can optionally detect headset or microphone button presses.
Add support for this by passing this event to the jack layer.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can detect the insertion/removal of headphones and headsets.
Enable reporting this status by enabling this interrupt and forwarding
this to upper-layers if a jack has been defined.
This jack definition and the resulting operation from a jack detection
event must currently be defined by sound card platform code until CODEC
outputs to jack mappings can be defined generically.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If for any reason, the backend does not have the requested substream
(like capture on a playback only backend), the BE will be skipped in
dpcm_be_dai_startup().
However, dpcm_apply_symmetry() does not skip those BE and will
dereference the be_substream (NULL) pointer anyway.
Like in dpcm_be_dai_startup(), just skip those BE.
Fixes: 906c7d690c ("ASoC: dpcm: Apply symmetry for DPCM")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk':
sound/soc/generic/simple-card-utils.c:164:18: warning:
parameter 'dai_name' set but not used [-Wunused-but-set-parameter]
It's not used since commit 0580dde594 ("ASoC: simple-card-utils: add
asoc_simple_debug_info()"), so can be removed.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On some devices (Teclast X98+ II tablet, maybe others), the jack
detection has been wired backwards, so when the ES8316 reports
headphones being present it means they are actually not plugged.
Use a quirk around this incorrect behaviour, which can be enabled
through the 'everest,jack-detect-inverted' boolean device property.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Acked-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the machine board with
mt6358, da7219 and max98357 codecs.
Signed-off-by: Shunli Wang <shunli.wang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the machine board with TS3A227.
Signed-off-by: Shunli Wang <shunli.wang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the ACPI ID for the product "chromebook pixel 2015" to match the
coreboot settings.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Support multiple endpoints on cs42L51 codec port
when used in of_graph context.
This patch allows to share the codec port between two CPU DAIs.
Example:
STM32MP157C-DK2 board uses CS42L51 audio codec.
This codec is connected to two serial audio interfaces,
which are configured either as rx or tx.
From AsoC point of view the topolgy is the following:
// 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51)
Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0
Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0
In the DT two endpoints have to be associated to the codec port:
cs42l51_port: port {
cs42l51_tx_endpoint: endpoint@0 {
remote-endpoint = <&sai2a_endpoint>;
};
cs42l51_rx_endpoint: endpoint@1 {
remote-endpoint = <&sai2b_endpoint>;
};
};
However, when the audio graph card parses the codec nodes, it expects
to find DAI interface indexes matching the endpoints indexes.
The current patch forces the use of DAI id 0 for both endpoints,
which allows to share the codec DAI between the two CPU DAIs
for playback and capture streams respectively.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The common pins were mistakenly not added to the DAPM graph.
Adding these pins will allow valid graphs to be created.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The important fixes at this time are a couple fixes in ALSA core:
a fix for PCM is about the OOB access in PCM OSS plugins that has
been for long time, but hasn't hit so often until now just because
we allocated a large buffer via vmalloc(), and surfaced more often
after switching to kvmalloc(). Another fix is for a long-standing
PCM problem wrt racy PM resume. Others are trivial nospec coverage
and usual HD-audio quirks.
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Merge tag 'sound-5.1-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The important fixes at this time are a couple fixes in ALSA core: a
fix for PCM is about the OOB access in PCM OSS plugins that has been
for long time, but hasn't hit so often until now just because we
allocated a large buffer via vmalloc(), and surfaced more often after
switching to kvmalloc(). Another fix is for a long-standing PCM
problem wrt racy PM resume.
Others are trivial nospec coverage and usual HD-audio quirks"
* tag 'sound-5.1-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix speakers on Acer Predator Helios 500 Ryzen laptops
ALSA: pcm: Don't suspend stream in unrecoverable PCM state
ALSA: hda/ca0132 - Simplify alt firmware loading code
ALSA: pcm: Fix possible OOB access in PCM oss plugins
ALSA: hda/realtek: Enable headset MIC of ASUS X430UN and X512DK with ALC256
ALSA: hda/realtek: Enable headset mic of ASUS P5440FF with ALC256
ALSA: hda/realtek: Enable ASUS X441MB and X705FD headset MIC with ALC256
ALSA: hda/realtek - Add support for Acer Aspire E5-523G/ES1-432 headset mic
ALSA: hda/realtek: Enable headset MIC of Acer Aspire Z24-890 with ALC286
ALSA: seq: oss: Fix Spectre v1 vulnerability
ALSA: rawmidi: Fix potential Spectre v1 vulnerability
alloc_pages_exact() is more suitable choice for allocating the sound
buffers, as it doesn't need to align with power-of-two. Along with
the conversion, we can drop __GFP_COMP as well.
The patch also replace the error messages to be more explicit.
Acked-by: Michal Hocko <mhocko@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_malloc_pages() and snd_free_pages() are merely thin wrappers of
the standard page allocator / free functions. Even the arguments are
compatible with some standard helpers, so there is little merit of
keeping these wrappers.
This patch replaces the all existing callers of snd_malloc_pages() and
snd_free_pages() with the direct calls of the standard helper
functions. In this version, we use a recently introduced one,
alloc_pages_exact(), which suits better than the old
snd_malloc_pages() implementation for our purposes. Then we can avoid
the waste of pages by alignment to power-of-two.
Since alloc_pages_exact() does split pages, we need no longer
__GFP_COMP flag; or better to say, we must not pass __GFP_COMP to
alloc_pages_exact(). So the former unconditional addition of
__GFP_COMP flag in snd_malloc_pages() is dropped, as well as in most
other places.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Michal Hocko <mhocko@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_timer_close() is supposed to close the timer instance and sync
with the deactivation of pending actions. However, there are still
some overlooked cases:
- It calls snd_timer_stop() at the beginning, but some other might
re-trigger the timer right after that.
- snd_timer_stop() calls del_timer_sync() only when all belonging
instances are closed. If multiple instances were assigned to a
timer object and one is closed, the timer is still running. Then
the pending action assigned to this timer might be left.
Actually either of the above is the likely cause of the reported
syzkaller UAF.
This patch plug these holes by introducing SNDRV_TIMER_IFLG_DEAD
flag. This is set at the beginning of snd_timer_close(), and the flag
is checked at snd_timer_start*() and else, so that no longer new
action is left after snd_timer_close().
Reported-by: syzbot+d5136d4d3240cbe45a2a@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For checking the pending timer instance that is still left on the
timer object that is being closed, we set/clear a bit flag
SNDRV_TIMER_IFLG_CALLBACK around the call of callbacks. This can be
simplified by replace with the list_empty() call for ti->ack_list.
This covers the existence more comprehensively and safely.
A gratis bonus is that we can get rid of SNDRV_TIMER_IFLG_CALLBACK bit
flag definition as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a card is under disconnection, we bail out immediately at each
timer interrupt or tasklet. This might leave some items left in ack
list. For a better integration of the upcoming change to check
ack_list emptiness, clear out the whole list upon the emergency exit
route.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer core has two almost identical code for processing callbacks:
once in snd_timer_interrupt() for fast callbacks and another in
snd_timer_tasklet() for delayed callbacks. Let's unify them.
In the new version, the resolution is read from ti->resolution at each
call, and this must be fine; ti->resolution is set in the preparation
step in snd_timer_interrupt().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a feature request for the ancient sutff, but it's still valid;
the loading of a GUS-patch isn't available via hwdep device although
it's supported over OSS sequencer. The only missing piece is the call
of snd_soundfont_load_guspatch() in synth emux hwdep code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers
don't work out of the box.
The problem can be worked around with hdajackretask, remapping the
"Black Headphone, Right side" pin (0x21) to the Internal speaker.
This patch adds a quirk to change this mapping by default.
[ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the
latest tree by tiwai ]
Signed-off-by: Bernhard Rosenkraenzer <bero@lindev.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements Audio Mixer machine driver for NXP iMX8 SOCs.
It connects together Audio Mixer and related SAI instances.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
The Audio Mixer is a on-chip functional module that allows mixing of
two audio streams into a single audio stream.
Audio Mixer datasheet is available here:
https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To enable S24_LE format, sample_type in topology fw has to be set to 1.
But sample_type defined in topology firmware configuration is not
getting reflected in the dsp param. This patch sets sample_type in base
config so that the sample type defined in the topology firmware is reflected
in the dsp params. This issues was uncovered while debugging the S24_LE format
which require the MSB byte in 32 bit word to be skipped. Setting sample_type
in topology firmware to 1 helps to skip MSB byte word.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some architectures do not yet support the common clock API at all but
the tlv320aic32x4 driver now requires it.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
UAPI Changes:
- Report an error early instead of SIGBUS later when mmap beyond BO size
Core Changes:
- This includes backmerge of drm-next and two merges of Maarten's
topic/hdr-formats
Driver Changes:
- Add Comet Lake (Gen9) PCI IDs to Coffee Lake ID list (Anusha)
- Add missing ICL PCI ID (Jose)
- Fix legacy gamma mode for ICL (Ville)
- Assume eDP is present on port A when there is no VBT (Thomas)
- Corrections to eDP training patterns (Jose)
- Fix PSR2 selective update corruption after PSR1 setup (Jose)
- Fix CRC mismatch error for DP link layer compliance (Aditya)
- Fix CNL DPLL readout and clean up code (Ville)
- Turn off the CUS when turning off a HDR plane (Ville)
- Avoid a race with execlist tasklet during race (Chris)
- Add missing CSC readout and clean up code (Ville)
- Avoid unnecessary wakeref during debugfs/drop_caches/set (Chris, Caz)
- Hold references to ring/HW context/context explicitly when used (Chris)
- Assume next platforms inherit old platform (Rodrigo)
- Use HWS indices rather than addresses for breadcrumbs (Chris)
- Add REG_BIT/REG_GENMASK and REG_FIELD_PREP macros (Jani)
- Convert crept in C99 types to kernel fixed size types (Jani)
- Avoid passing full dev_priv in forcewake functions (Daniele)
- Reset GuC on GPU reset (Sujaritha)
- Rework MG and Combo PLLs to vfuncs (Lucas)
- Explicitly track ppGTT size (Chris, Bob)
- Coding style improvements and code modularization (Ville)
- Selftest and debugging improvements (Chris)
Signed-off-by: Dave Airlie <airlied@redhat.com>
# Conflicts:
# drivers/gpu/drm/i915/intel_hdmi.c
From: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20190325124925.GA12726@jlahtine-desk.ger.corp.intel.com
The clocking and processing blocks are now properly set up to
support 192000 sample rates. Allow drivers to ask for that.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
mclk is not used by anything anymore. Remove support for it.
All that information now comes from the clock tree.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sysclk is now managed by the CCF. Change this function
to merely find the system clock and set it using
clk_set_rate.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code uses a static lookup table to determine the
settings of the various clock devices on board the chip. This is
limiting in a couple of ways. First, this doesn't allow for any
master clock rates other than the three that have been
precalculated. Additionally, new sample rates are difficult to
add to the table. Witness that the chip is capable of 192000 Hz
sampling, but it is not provided by this driver. Last, if the
driver is clocked by something that isn't a crystal, the
upstream clock may not be able to achieve exactly the rate
requested in the driver. This will mean that clocking will be
slightly off for the sampling clock or that it won't work at all.
This patch determines the settings for all of the clocks at
runtime considering the real conditions of the clocks in the
system. The rules for the clocks are in TI's SLAA557 application
guide on pages 37, 51 and 77.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move these to separate helper functions. This looks cleaner and fits
better with the new clock setting in CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Control the clock gating to the various clock components to use
the CCF. This allows us to prepare_enalbe only 3 clocks and the
relationships assigned to them will cause upstream clockss to
enable automatically. Additionally we can do this in a single
call to the CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage BDIV divider as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage DAC/ADC dividers as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage codec clock input as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage the on-board PLL as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of single config, private_value is left uninitialized.
The private_value does need to be initialized or in
snd_soc_dapm_new_control_unlocked() call failure case, it leads to a
bogus free in snd_soc_dapm_free_kcontrol()
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently snd_aloop supports only S16 and S32 audio sample formats. With
this patch the S24 formats are also supported.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently PCM core sets each opened stream forcibly to SUSPENDED state
via snd_pcm_suspend_all() call, and the user-space is responsible for
re-triggering the resume manually either via snd_pcm_resume() or
prepare call. The scheme works fine usually, but there are corner
cases where the stream can't be resumed by that call: the streams
still in OPEN state before finishing hw_params. When they are
suspended, user-space cannot perform resume or prepare because they
haven't been set up yet. The only possible recovery is to re-open the
device, which isn't nice at all. Similarly, when a stream is in
DISCONNECTED state, it makes no sense to change it to SUSPENDED
state. Ditto for in SETUP state; which you can re-prepare directly.
So, this patch addresses these issues by filtering the PCM streams to
be suspended by checking the PCM state. When a stream is in either
OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend
action is skipped.
To be noted, this problem was originally reported for the PCM runtime
PM on HD-audio. And, the runtime PM problem itself was already
addressed (although not intended) by the code refactoring commits
3d21ef0b49 ("ALSA: pcm: Suspend streams globally via device type PM
ops") and 17bc4815de ("ALSA: pci: Remove superfluous
snd_pcm_suspend*() calls"). These commits eliminated the
snd_pcm_suspend*() calls from the runtime PM suspend callback code
path, hence the racy OPEN state won't appear while runtime PM.
(FWIW, the race window is between snd_pcm_open_substream() and the
first power up in azx_pcm_open().)
Although the runtime PM issue was already "fixed", the same problem is
still present for the system PM, hence this patch is still needed.
And for stable trees, this patch alone should suffice for fixing the
runtime PM problem, too.
Reported-and-tested-by: Jon Hunter <jonathanh@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
w_text_param can be NULL and it is being dereferenced without checking.
Add the missing sanity check to prevent NULL pointer dereference.
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unlike other drivers probe method, of_match_node return value
is not used or checked. This patch removes the redundant code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Reviewed-by: Steven Price <steven.price@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support of low power modes to STM32 SAI driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using bare externs outside include files that types should
at least match. This fixes a type confusion between bool
and int.
Cc: broonie@kernel.org
Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do division with div_u64 for the PLL calculation.
These errors are fixed and list as follows:
1."__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
2."__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
3. nau8810.c:(.text.nau8810_calc_pll+0xd8): undefined reference to
`__udivdi3'
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The only significant change is the regression fixes for the jack
detection at resume on HD-audio, while others are all small or
trivial fixes like the coverage of missing error code or usual
HD-audio quirk.
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Merge tag 'sound-5.1-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The only significant change is the regression fixes for the jack
detection at resume on HD-audio, while others are all small or trivial
fixes like the coverage of missing error code or usual HD-audio quirk"
* tag 'sound-5.1-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286
ALSA: hda - Enforces runtime_resume after S3 and S4 for each codec
ALSA: hda - Don't trigger jackpoll_work in azx_resume
ALSA: opl3: fix mismatch between snd_opl3_drum_switch definition and declaration
ALSA: hda - add Lenovo IdeaCentre B550 to the power_save_blacklist
ALSA: firewire-motu: use 'version' field of unit directory to identify model
ALSA: sb8: add a check for request_region
ALSA: echoaudio: add a check for ioremap_nocache
ca0132 codec driver loads the firmware selectively depending on the
model in addition to the fallback of the default firmware. The code
works good, but a minor problem is that the current code seems
confusing for Clang where it spews a warning about uninitialized
variable.
This patch simplifies the code flow for such a false-positive
warning. After this refactoring, the ca0132_spec.alt_firmware_present
field is no longer used, hence it's eliminated as well.
Reported-and-tested-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM OSS emulation converts and transfers the data on the fly via
"plugins". The data is converted over the dynamically allocated
buffer for each plugin, and recently syzkaller caught OOB in this
flow.
Although the bisection by syzbot pointed out to the commit
65766ee0bf ("ALSA: oss: Use kvzalloc() for local buffer
allocations"), this is merely a commit to replace vmalloc() with
kvmalloc(), hence it can't be the cause. The further debug action
revealed that this happens in the case where a slave PCM doesn't
support only the stereo channels while the OSS stream is set up for a
mono channel. Below is a brief explanation:
At each OSS parameter change, the driver sets up the PCM hw_params
again in snd_pcm_oss_change_params_lock(). This is also the place
where plugins are created and local buffers are allocated. The
problem is that the plugins are created before the final hw_params is
determined. Namely, two snd_pcm_hw_param_near() calls for setting the
period size and periods may influence on the final result of channels,
rates, etc, too, while the current code has already created plugins
beforehand with the premature values. So, the plugin believes that
channels=1, while the actual I/O is with channels=2, which makes the
driver reading/writing over the allocated buffer size.
The fix is simply to move the plugin allocation code after the final
hw_params call.
Reported-by: syzbot+d4503ae45b65c5bc1194@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS laptop P5440FF with ALC256 can't detect the headset microphone
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset
MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
li->conf will be 0 if it was not DPCM case.
Then, 1) we shouldn't call devm_kcalloc() with size 0,
2) we need NULL pointer check if li->conf was not 0.
This patch fixed above issues.
Special thanks to Pierre-Louis Bossart
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lochnagar is an evaluation and development board for Cirrus
Logic Smart CODEC and Amp devices. It allows the connection of
most Cirrus Logic devices on mini-cards, as well as allowing
connection of various application processor systems to provide a
full evaluation platform.
Lochnagar 2 provides a set of line inputs/outputs, and a USB audio
device. This driver adds support for these analog line connections and
the Lochnagar side of the USB audio link.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Different processing blocks are required for different sampling
rates and power parameters. Set the processing blocks based
on this information.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Inter-IC Sound Controller (I2SMCC) provides a 5-wire, bidirectional,
synchronous, digital audio link to external audio devices: I2SMCC_DIN,
I2SMCC_DOUT, I2SMCC_WS, I2SMCC_CK, and I2SMCC_MCK pins.
The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and
supports a Time Division Multiplexed (TDM) interface with external
multi-channel audio codecs.
The I2SMCC consists of a receiver, a transmitter and a common clock
generator that can be enabled separately to provide Master, Slave or
Controller modes with receiver and/or transmitter active.
DMA Controller channels, separate for the receiver and for the transmitter,
allow a continuous high bit rate data transfer without processor
intervention to the following:
- Audio CODECs in Master, Slave, or Controller mode
- Stereo DAC or ADC through a dedicated I2S serial interface
- Multi-channel or multiple stereo DACs or ADCs, using the TDM format
This IP is embedded in Microchip's new sam9x60 SoC.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils is using asoc_simple_card_xxx() for each
function naming, but it is very verbose.
Thus it is easy to be over 80 char.
This patch renames it to asoc_simple_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph are using
asoc_simple_card_parse_dai() which is different implementation.
But, these are implemanted at simple-card-utils.
It should be implemanted at each files.
This patch separate these into each files.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph are initializing each priv,
but it is same operation.
This patch adds new asoc_simple_card_init_priv() and initialize
priv by same operation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_be_hw_params_fixup() between in these
2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_dai_init() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_hw_param() between in these 2 drivers.
One note is that only simple-card supports simple_set_clk_rate()
at hw_param from commit e9be4ffd4f ("ASoC: simple-card: set cpu
dai clk in hw_params").
By this patch, audio-graph has same feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_shutdown() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_startup() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Historically, simple-card/simple-scu-card/audio-graph/audio-graph-scu
are similar but different generic sound card.
simple-scu-card which was for DPCM was merged into simple-card, and
audio-graph-scu which was for DPCM was merged into audio-graph.
simple-card is for non OF graph sound card, and
audio-graph is for OF graph sound card.
And, small detail difference (= function parameter, naming, etc)
between simple-card/audio-graph has been unified.
So today, the difference between simple-card/audio-graph are
just using OF graph style, or not.
In other words, there should no difference other than OF graph sytle.
simple-card/audio-graph are using own priv today , but we can merge it.
This patch merge it at simple_card_utils.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils has dev_dbg(), but people want to
add #define DEBUG at simple-card/audio-graph, not simple-card-utils.
And, people want to get all information.
This patch adds new asoc_simple_debug_info() to indicates information.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the DAI clocks for DA7219 have now been split into BCLK and WCLK,
the clock lookup name needs to be udpated here to select BCLK to
achieve the same functionality as before with regards to DAI clock
gating.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the purposes of platforms which use the codec as DAI clock
master for the CPU and other codec devices, there is the need to
not only expose the clock gating of BCLK and WCLK but also the
ability to set those rates without going through the ASoC APIs.
To make this possible, the previous CCF implementation in the
driver has been extended to separate BCLK and WCLK out. WCLK is
the parent clock to BCLK, and is also the clock gate for both.
BCLK in HW is a factor/multiplier of WCLK so derives from whatever
SR is chosen for WCLK, hence the need to make it a child of WCLK
for the purposes of CCF. Enabling/disabling either BCLK or WCLK
will result in clocks being ungated/gated accordingly. To simplify
matters, these clocks can only be configured if the codec is set
as master, otherwise CCF control is disallowed.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is very low possibility ( < 0.1% ) that channel swap happened
in beginning when multi output/input pin is enabled. The issue is
that hardware can't send data to correct pin in the beginning with
the normal enable flow.
This is hardware issue, but there is no errata, the workaround flow
is that: Each time playback/recording, firstly clear the xSMA/xSMB,
then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled
before xSMA). Which is to use the xSMA as the trigger start register,
previously the xCR_TE or xCR_RE is the bit for starting.
Fixes commit 43d24e76b6 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a constraint for the channel number setting on the
asrc of older version (e.g. imx35), the channel number should
be even, odd number isn't valid.
So add this constraint when the asrc of older version is used.
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
To fix this, set the INCR bit in all cases.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect
the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk
applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Aspire Z24-890 cannot detect the headset MIC until
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap)
Fix this by sanitizing dev before using it to index dp->synths.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->stream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap)
Fix this by sanitizing info->stream before using it to index
rmidi->streams.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed96 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G")
Fixes: b72f936f6b ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register platform component with a prefix, to avoid warnings
on debugfs entries creation, due to component name
redundancy.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DFSDM must be stopped when a new setting is applied.
restart systematically DFSDM on multiple prepare calls,
to apply changes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm_adsp_ops structures should be static and correct two printf
specifiers.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Fixes: 4e08d50d1f ("ASoC: wm_adsp: Factor out DSP specific operations")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can simplify the code by caching the CPU DAI master/slave
information rather than reading previously set register bit.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Break the clock setting logic out from the main hw_params. It's
rather large and unweildy and makes for a large function. This
also better enables some of the following changes to the clock
tree access in the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on MACH_JZ4740 prevent us from creating a generic kernel that
works on more than one MIPS board. Instead, we just depend on MIPS being
set.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A Halo Core DSP has a memory protection unit that can trap and signal
memory access faults. This patch adds a function that dumps the fault
information.
The interrupt reaches the host via the parent codec interrupt controller
so this fault function is exported to be called by the codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Halo core is a new generation of audio DSP architecture from
Cirrus Logic. A new iteration of the WMFW file format (v3) is also
added, for this new architecture. Currently this format is not
supported on the old ADSP2 architecture however support may be
added for it in the future.
Signed-off-by: Wen Shi <wenshi@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the signature of mtk_regmap_update_bits to also take a shift, and
warn when reg >= 0 but shift < 0. This reduce the code repetition
on the calling side, and prevent future UBSAN warning when some of the
xxx_shift and xxx_reg are both set to -1.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sound/soc/mediatek/common/mtk-afe-fe-dai.c, when xxx_reg is -1, it's
a no-op to call mtk_regmap_update_bits, but since both xxx_reg and
xxx_shift are set to -1, the (1 << xxx_shift) in the argument would
trigger a UBSAN warning.
Fix the warning by setting those xxx_shift to 0 instead.
Note that since the code explicitly checks .mono_shift >= 0 and
.fs_shift >= 0 before using them in '<<' operator, those two members are
not set to 0.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for the addition of more types of DSP core refactor the
handling of DSP specific operations such as starting the memory or
enabling the core into a set of callbacks. This should make it easier to
add new core types and allow for more code reuse between them.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to duplicate this code for both ADSP1 and 2 as the
handling is exactly the same.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for further additions refactor the reading of the
firmware status.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The original wm_adsp2_early_event took an additional frequency
argument for clocking control so could not be used directly as a
DAPM callback. But this setup could equally be done by the codec
driver function wrapping wm_adsp2_early event. In preparation
for adding support for new core types wm_adsp2_set_dspclk has
been exported, and the freq argument removed so that it can
be used directly as a DAPM callback.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This function is not presently called from outside the adsp code and nor
should it be, as such stop exporting it.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a watchdog timeout is received from the DSP it is safe to assume the
DSP is not functioning anymore and as such any active compressed streams
should be put into an error state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Best to lock across handling the bus error to ensure the DSP doesn't
change power state as we are reading the status registers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During recent logging improvements it seems two error messages lost
their updates during patch application/rebasing. Add these back in.
Fixes: 0d3fba3e7a ("ASoC: wm_adsp: Improve logging messages")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some strings are allocated by kstrdup, but not freed when error
happened.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stream_name is allocated by kstrdup. We have to free it when the
dai is removed or return from error.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3baffc4a84 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is parsing mclk_fs at many places, but it should be
same operation. This patch adds graph_parse_mclk_fs()
and parse it.
This patch also renames similar function graph_get_conversion()
to graph_parse_convert().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
use same naming rule, and this patch add missing of_node_put() on it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is parsing mclk_fs at many places, but it should be
same operation. This patch adds simple_parse_mclk_fs()
and parse it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The jack type detection needs the main bias power of analog.
The modification makes sure the main bias power on/off while jack plug/unplug.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skip for i2s5 in mck_disable which is also bypassed in mck_enable.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In kernel API of Linux FireWire subsystem, handlers of isochronous
receive (IR) context can get context headers as an argument of
callback. When 4 byte header is used, the context header includes
isochronous packet header for each packet. When 8 byte header is
used, it includes isochronous cycle as well.
ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous
cycle from the cycle of interrupt. The usage of 8 byte header can
obsolete the computation.
Furthermore, this change works well for a case that a series of
packet in one interrupt includes skipped isochronous cycle,
This commit uses 8 byte header to handle isochronous cycle.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds support for MOTU 8pre FireWire, which was shipped 2007
and nowadays already discontinued. Userspace applications can transmit
and receive PCM frames and MIDI messages for this model via ALSA PCM
interface and RawMidi/Sequencer interfaces.
Like the other models of MOTU FireWire series, this model has many
quirks in its CIP.
At first, data channels for two pairs of optical interfaces. At lower
sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is
available for ADAT data, thus 8 data chunks are transferred by CIP.
At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz,
two pairs are available to keep 8 chunks for ADAT data, thus CIP
still includes 8 data chunks.
Apart from data chunks for optical interface, CIP includes fixed number
of data chunks. In tx stream, two chunks for status message, eight
chunks for samples from analog 1-8 input, two chunks for mix-return.
In rx stream, two chunks for control message, two chunks for main 1-2
output, two chunks for phone 1-2 output, two chunks for dummy 1-2.
CIP header in tx stream includes quirks for its dbs and dbc fields.
The value of dbs field is fixed to 0x13, against its actual size.
The value of dbc field is firstly updated to 0x07 from zero, then
it's incremented continuously according to actual number of data h
blocks.
Finally, the model has own bits to disable frame fetch.
This commit uses several options to absorb the above quirks.
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 0410b57d bus_info_length 4, crc_length 16, crc 46461
404 31333934 bus_name "1394"
408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4)
40c 0001f200 company_id 0001f2 |
410 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 0003991c directory_length 3, crc 39196
42c 120001f2 specifier id
430 1300000f version
434 17103800 model
eui-64 leaf at 438
-----------------------------------------------------------------
438 00022681 leaf_length 2, crc 9857
43c 0001f200 company_id 0001f2 |
440 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_opl3_drum_switch declaration in the header file
has the order of the two arguments on_off and vel swapped when
compared to the definition arguments of vel and on_off. Fix this
by swapping them around to match the definition.
This error predates the git history, so no idea when this error
was introduced.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another machine which does not like the power saving (noise):
https://bugzilla.redhat.com/show_bug.cgi?id=1689623
Also, reorder the Lenovo C50 entry to keep the table sorted.
Reported-by: hs.guimaraes@outlook.com
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add BYT_RT5651_JD_NOT_INV quirk for devices with an inverted
(active-high instead of the normal active-low) jack-detect switch.
And add a quirk for the Complet Electro Serv MY8307 tablet which has
an inverted jack-detect switch (and a mono-speaker).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards use a jack-receptacle with a switch which reports the
jack-inserted status as active-high, rather then the standard active-low
reporting most jacks use.
This commit adds support for it. This is activated by a boolean
"realtek,jack-detect-not-inverted" device-property. The not-inverted
in the device-property name, rather then active-high, was chosen to keep
the device-property naming consistent with the rt5640 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suspend and resume sleep callbacks to STM32 SPDIFRX driver,
to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dpcm get from fe_clients/be_clients
may be free before use
Add a spin lock at snd_soc_card level,
to protect the dpcm instance.
The lock may be used in atomic context, so use spin lock.
possible race condition between
void dpcm_be_disconnect(
...
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
kfree(dpcm);
...
and
for_each_dpcm_fe()
for_each_dpcm_be*()
race condition example
Thread 1:
snd_soc_dapm_mixer_update_power()
-> soc_dpcm_runtime_update()
-> dpcm_be_disconnect()
-> kfree(dpcm);
Thread 2:
dpcm_fe_dai_trigger()
-> dpcm_be_dai_trigger()
-> snd_soc_dpcm_can_be_free_stop()
-> if (dpcm->fe == fe)
Excpetion Scenario:
two FE link to same BE
FE1 -> BE
FE2 ->
Thread 1: switch of mixer between FE2 -> BE
Thread 2: pcm_stop FE1
Exception:
Unable to handle kernel paging request at virtual address dead0000000000e0
pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
sound/soc/soc-pcm.c:3226
if (dpcm->fe == fe)
lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
Backtrace:
[<ffffff89602dba80>] notify_die+0x68/0xb8
[<ffffff896028c7dc>] die+0x118/0x2a8
[<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c
[<ffffff89602a27f4>] do_translation_fault+0x64/0xa0
[<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0
[<ffffff8960282ad0>] el1_da+0x24/0x40
[<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
[<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
[<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44
[<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c
[<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c
[<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128
[<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0
[<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14
[<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244
[<ffffff8960283740>] el0_svc_naked+0x34/0x38
[<ffffffffffffffff>] 0xffffffffffffffff
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some amplifier may not have a GPIO to control the power, but instead simply
rely on the regulator to power up and down the amplifier.
In order to support those setups, let's make the GPIO optional.
Signed-off-by: Mylène Josserand <mylene.josserand@bootlin.com>
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ALSA firewire-motu driver uses the value of 'model' field
of unit directory in configuration ROM for modalias for MOTU
FireWire models. However, as long as I checked, Pre8 and
828mk3(Hybrid) have the same value for the field (=0x100800).
unit | version | model
--------------- | --------- | ----------
828mkII | 0x000003 | 0x101800
Traveler | 0x000009 | 0x107800
Pre8 | 0x00000f | 0x100800 <-
828mk3(FW) | 0x000015 | 0x106800
AudioExpress | 0x000033 | 0x104800
828mk3(Hybrid) | 0x000035 | 0x100800 <-
When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3,
I got change of the value from 0x100800 to 0x103800. On the other
hand, the value of 'version' field is fixed to 0x00000f. As a quick
glance, the higher 12 bits of the value of 'version' field represent
firmware version, while the lower 12 bits is unknown.
By induction, the value of 'version' field represents actual model.
This commit changes modalias to match the value of 'version' field,
instead of 'model' field. For degug, long name of added sound card
includes hexadecimal value of 'model' field.
Fixes: 6c5e1ac0e1 ("ALSA: firewire-motu: add support for Motu Traveler")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v4.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case request_region fails, the fix returns an error code to
avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case ioremap_nocache fails, the fix releases chip and returns
an error code upstream to avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some gleaning after the first batch; mostly about HD-audio quirks but
also some NULL dereference fixes in corner cases and a random build
error fix, too.
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Merge tag 'sound-fix-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Some cleaning after the first batch; mostly about HD-audio quirks but
also some NULL dereference fixes in corner cases and a random build
error fix, too"
* tag 'sound-fix-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Add support headset mode for New DELL WYSE NB
ALSA: hda/realtek - Add support headset mode for DELL WYSE AIO
ALSA: hda/realtek: merge alc_fixup_headset_jack to alc295_fixup_chromebook
ALSA: pcm: Fix function name in kernel-doc comment
ALSA: hda: hdmi - add Icelake support
ALSA: hda - add more quirks for HP Z2 G4 and HP Z240
ALSA: hda/realtek - Fixed Headset Mic JD not stable
ALSA: hda/realtek: Enable headset MIC of Acer TravelMate X514-51T with ALC255
ALSA: hda/tegra: avoid build error without CONFIG_PM
ALSA: usx2y: Fix potential NULL pointer dereference
ALSA: hda: Avoid NULL pointer dereference at snd_hdac_stream_start()
Component driver may want to use tlv data. Create tlv before
soc_tplg_init_kcontrol so component driver can use the tlv data
in the control_load ops.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook.
There are no other users for ALC225_FIXUP_HEADSET_JACK other than
the chromebook hardware.
Fixes: 10f5b1b85e ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a port of the ASoC Icelake HDMI codec code to the legacy
HDA driver with some cleanups.
ASoC commit 019033c854a20e10f691f6cc0e897df8817d9521:
"ASoC: Intel: hdac_hdmi: add Icelake support"
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Bard liao <bard.liao@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver will select correct BCLK automatically according to
BCLK and FS information in I2S master mode.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After commit fbeec965b8 ("ASoC: samsung: odroid: Fix 32000 sample rate
handling") the audio root clock frequency is configured improperly for
44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead
of 22579000 Hz. This results in a too low value of the PSR clock divider
in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g.
1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2).
Fix this by increasing the correction passed to clk_set_rate() to take
into account inaccuracy of the EPLL frequency properly.
Fixes: fbeec965b8 ("ASoC: samsung: odroid: Fix 32000 sample rate handling")
Reported-by: JaeChul Lee <jcsing.lee@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver changes the stream name of DAC and ADC to avoid the issue of
widget with prefixed name. When the machine adds prefixed name for codec,
the stream name of DAI may not find the widgets.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and
HP Z240 models.
Reported-by: Jeff Burrell <jeff.burrell@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Implement of reset combo jack JD. It will show normally.
Fixes: e854747d75 ("ALSA: hda/realtek - Enable headset button support for new codec")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC
until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the
internal DMIC uses another module - snd_soc_skl as the driver. We still
need the NID 0x1a in the quirk to enable the headset MIC.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The #ifdef protection around the PM functions is wrong, leading to
a failed reference in some configurations:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend':
sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration]
Better remove the #ifdefs entirely and rely on the compiler silently
dropping unused functions marked __maybe_unused.
Fixes: 707e0759f2 ("ALSA: hda/tegra: implement runtime suspend/resume")
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_alloc_urb() can fail due to kmalloc failure and push the error
upstream. Further this can cause a NULL pointer dereference in
init_pipe_urbs(). This patch avoids such a scenario.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the capture and playback channels are optional in the axi_i2s IP
block. Reflect this in the driver by enabling only the channel(s) that
have a DMA.
Signed-off-by: Luca Ceresoli <luca@lucaceresoli.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
compiler complains about following declarations
sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern1[] = {
^~~~~
sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern2[] = {
^~~~~
sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsisr_table[] = {
^~~~~
sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan288888[] = {
^~~~~
sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan244888[] = {
^~~~~
sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan222222[] = {
^~~~~
This patch moves the 'static' keyword to the front of the
declaration to fix the compiler warnings
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
lockdep warns us that priv->lock and k->k_lock can cause a
deadlock when after acquire of k->k_lock, process is interrupted
by src, while in another routine of src .init, k->k_lock is
acquired with priv->lock held.
This patch avoids a potential deadlock by not calling soc_device_match()
in SRC .init callback, instead it adds new soc fields in priv->flags to
differentiate SoCs.
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
- Declare SR as volatile, as it is changed by hardware.
- Remove TXDR from readable and volatile register list,
as it is intended for write accesses only.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has two issues when machine add prefix name for codec.
(1)The stream name of DAI can't find the AIF widgets.
(2)The drivr can enable/disalbe the MICBIAS and SAR widgets.
The patch will fix these issues caused by prefixed name added.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dpcm get from fe_clients/be_clients
may be free before use
Add a spin lock at snd_soc_card level,
to protect the dpcm instance.
The lock may be used in atomic context, so use spin lock.
Use irq spin lock version,
since the lock may be used in interrupts.
possible race condition between
void dpcm_be_disconnect(
...
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
kfree(dpcm);
...
and
for_each_dpcm_fe()
for_each_dpcm_be*()
race condition example
Thread 1:
snd_soc_dapm_mixer_update_power()
-> soc_dpcm_runtime_update()
-> dpcm_be_disconnect()
-> kfree(dpcm);
Thread 2:
dpcm_fe_dai_trigger()
-> dpcm_be_dai_trigger()
-> snd_soc_dpcm_can_be_free_stop()
-> if (dpcm->fe == fe)
Excpetion Scenario:
two FE link to same BE
FE1 -> BE
FE2 ->
Thread 1: switch of mixer between FE2 -> BE
Thread 2: pcm_stop FE1
Exception:
Unable to handle kernel paging request at virtual address dead0000000000e0
pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
sound/soc/soc-pcm.c:3226
if (dpcm->fe == fe)
lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
Backtrace:
[<ffffff89602dba80>] notify_die+0x68/0xb8
[<ffffff896028c7dc>] die+0x118/0x2a8
[<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c
[<ffffff89602a27f4>] do_translation_fault+0x64/0xa0
[<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0
[<ffffff8960282ad0>] el1_da+0x24/0x40
[<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
[<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
[<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44
[<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c
[<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c
[<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128
[<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0
[<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14
[<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244
[<ffffff8960283740>] el0_svc_naked+0x34/0x38
[<ffffffffffffffff>] 0xffffffffffffffff
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If playback and capture are enabled concurrently, when the capture stops
the output becomes inaudile. The playback application will become stuck
and underrun after a timeout.
This is caused by mistaken use of the stream_id, which should only be
set for playback and not for capture
Tested on Apollolake and Kabylake with SST driver.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current implementation of the hdac_hda codec results in zero-valued
samples on capture and noise with headset playback when SOF is used on
platforms with an on-board HDaudio codec. This is root-caused to SOF
using be_hw_params_fixup, and the prepare() call using invalid runtime
fields to determine the format.
This patch moves the format handling to the hw_params() callback, as
done already for hdac_hdmi, to make sure the fixed-up information is
taken into account but keeps the codec initialization in prepare() as
the stream_tag is only available at that time. Moving everything in the
prepare() callback is possible but the code is less elegant so this
two-step solution was chosen.
The solution was tested with the SST driver with no regressions, and all
the issues with SOF playback and capture are solved.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On HDaudio platforms, if playback is started when capture is working,
there is no audible output.
This can be root-caused to the use of the rx|tx_mask to store an HDaudio
stream tag.
If capture is stared before playback, rx_mask would be non-zero on HDaudio
platform, then the channel number of playback, which is in the same codec
dai with the capture, would be changed by soc_pcm_codec_params_fixup based
on the tx_mask at first, then overwritten by this function based on rx_mask
at last.
According to the author of tx|rx_mask, tx_mask is for playback and rx_mask
is for capture. And stream direction is checked at all other references of
tx|rx_mask in ASoC, so here should be an error. This patch checks stream
direction for tx|rx_mask for fixup function.
This issue would affect not only HDaudio+ASoC, but also I2S codecs if the
channel number based on rx_mask is not equal to the one for tx_mask. It could
be rarely reproduecd because most drivers in kernel set the same channel number
to tx|rx_mask or rx_mask is zero.
Tested on all platforms using stream_tag & HDaudio and intel I2S platforms.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_DAVINCI_MCASP driver can use either edma or sdma as
a back-end, and it takes the presence of the respective dma engine
drivers in the configuration as an indication to which ones should be
built. However, this is flawed in multiple ways:
- With CONFIG_TI_EDMA=m and CONFIG_SND_SOC_DAVINCI_MCASP=y,
is enabled as =m, and we get a link error:
sound/soc/ti/davinci-mcasp.o: In function `davinci_mcasp_probe':
davinci-mcasp.c:(.text+0x930): undefined reference to `edma_pcm_platform_register'
- When CONFIG_SND_SOC_DAVINCI_MCASP=m has already been selected by
another driver, the same link error appears even if CONFIG_TI_EDMA
is disabled
There are possibly other issues here, but it seems that the only reasonable
solution is to always build both SND_SOC_TI_EDMA_PCM and
SND_SOC_TI_SDMA_PCM as a dependency here. Both are fairly small and
do not have any other compile-time dependencies, so the cost is
very small, and makes the configuration stage much more consistent.
Fixes: f2055e145f ("ASoC: ti: Merge davinci and omap directories")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that SOC_ENUM_SINGLE_EXT_DECL() contains a 'const'
modifier already, so adding another one does not make it more const:
sound/soc/ti/ams-delta.c:203:14: error: duplicate 'const' declaration specifier [-Werror,-Wduplicate-decl-specifier]
static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
^
include/sound/soc.h:351:2: note: expanded from macro 'SOC_ENUM_SINGLE_EXT_DECL'
const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After running into a link error:
sound/soc/ti/edma-pcm.o:(.rodata+0x18): undefined reference to `edma_filter_fn'
I checked all users of this, and they have new-style 'dma_slave_map' tables,
so none of them should still need it. Removing the associated lines
simplifies the code and avoids the build-time dependency on the
respective dmaengine drivers.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use alsa snd_pcm_hw_constraint_single service to manage
channels restriction. This provides better status on driver
limitations, to the application.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update traces to log capture/playback stream start/stop.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Building with clang shows a variable that is only used by the
suspend/resume functions but defined outside of their #ifdef block:
sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted
We commonly fix these by marking the PM functions as __maybe_unused,
but here that would grow the davinci_mcasp structure, so instead
add another #ifdef here.
Fixes: 1cc0c054f3 ("ASoC: davinci-mcasp: Convert the context save/restore to use array")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
compiler complains about following declarations
sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern1[] = {
^~~~~
sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern2[] = {
^~~~~
sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsisr_table[] = {
^~~~~
sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan288888[] = {
^~~~~
sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan244888[] = {
^~~~~
sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan222222[] = {
^~~~~
This patch moves the 'static' keyword to the front of the
declaration to fix the compiler warnings
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch sets missing stream_name of capture part of the DAI driver
so we can define DAPM routing properly also for the capture stream.
While at it "Playback" suffix is added to the playback stream names
to clearly identify playback/capture.
Together with related dts patch this fixes NULL pointer dereference
when opening ALSA device for recording on Odroid XU3.
Fixes: 64aba9bca5 ("ASoC: samsung: i2s: Add widgets and routes for DPCM support")
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Here is the big char/misc driver patch pull request for 5.1-rc1.
The largest thing by far is the new habanalabs driver for their AI
accelerator chip. For now it is in the drivers/misc directory but will
probably move to a new directory soon along with other drivers of this
type.
Other than that, just the usual set of individual driver updates and
fixes. There's an "odd" merge in here from the DRM tree that they asked
me to do as the MEI driver is starting to interact with the i915 driver,
and it needed some coordination. All of those patches have been
properly acked by the relevant subsystem maintainers.
All of these have been in linux-next with no reported issues, most for
quite some time.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'char-misc-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc
Pull char/misc driver updates from Greg KH:
"Here is the big char/misc driver patch pull request for 5.1-rc1.
The largest thing by far is the new habanalabs driver for their AI
accelerator chip. For now it is in the drivers/misc directory but will
probably move to a new directory soon along with other drivers of this
type.
Other than that, just the usual set of individual driver updates and
fixes. There's an "odd" merge in here from the DRM tree that they
asked me to do as the MEI driver is starting to interact with the i915
driver, and it needed some coordination. All of those patches have
been properly acked by the relevant subsystem maintainers.
All of these have been in linux-next with no reported issues, most for
quite some time"
* tag 'char-misc-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc: (219 commits)
habanalabs: adjust Kconfig to fix build errors
habanalabs: use %px instead of %p in error print
habanalabs: use do_div for 64-bit divisions
intel_th: gth: Fix an off-by-one in output unassigning
habanalabs: fix little-endian<->cpu conversion warnings
habanalabs: use NULL to initialize array of pointers
habanalabs: fix little-endian<->cpu conversion warnings
habanalabs: soft-reset device if context-switch fails
habanalabs: print pointer using %p
habanalabs: fix memory leak with CBs with unaligned size
habanalabs: return correct error code on MMU mapping failure
habanalabs: add comments in uapi/misc/habanalabs.h
habanalabs: extend QMAN0 job timeout
habanalabs: set DMA0 completion to SOB 1007
habanalabs: fix validation of WREG32 to DMA completion
habanalabs: fix mmu cache registers init
habanalabs: disable CPU access on timeouts
habanalabs: add MMU DRAM default page mapping
habanalabs: Dissociate RAZWI info from event types
misc/habanalabs: adjust Kconfig to fix build errors
...
Commit 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
re-worked the clean-up of any platform pointers that may have been
initialised by the function snd_soc_init_platform(). This commit missed
one error path where if any of the prelinks for a soundcard failed to
initialise, then these platform pointers would not be cleaned-up. This
then prevents the soundcard from being initialised following a probe
deferral when any of the soundcard prelinks cannot be found.
Fix this by ensuring that soc_cleanup_platform() is called when
initialising the soundcard prelinks fails.
Fixes: 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
Signed-off-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Limiting the value of the passed in params->msbits in the hw_params()
callback is redundant on three counts:
1. We already specify in the DAI driver that we can only handle up to
24 bits. This means msbits will be limited to 24 via the ALSA
constraints imposed by the ASoC core, unless we have multiple codecs
that can handle more bits.
2. Nothing in our hw_params() implementation uses this value.
3. The copy of the params that we are passed by the ASoC core never
reads back the msbits value.
Consequently, this code is unnecessary and does nothing useful. Remove
it.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error check on set_sync function return.
Add of_node_put() as of_get_parent() takes a reference
which has to be released.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change capabilities exposed in SAI S/PDIF mode, to match
actually supported formats.
In S/PDIF mode only 32 bits stereo is supported.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow indexation of sai iec958 controls according
to device id.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch
cases where we are expecting to fall through.
This patch fixes the following warning:
In file included from sound/soc/codecs/ab8500-codec.c:24:
sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’:
./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=]
_dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’
dev_err(dai->component->dev,
^~~~~~~
sound/soc/codecs/ab8500-codec.c:2132:2: note: here
default:
^~~~~~~
Warning level 3 was used: -Wimplicit-fallthrough=3
This patch is part of the ongoing efforts to enable
-Wimplicit-fallthrough.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using the S/PDIF DAI, there is no requirement to call
snd_soc_dai_set_fmt() as there is no DAI format definition that defines
S/PDIF. In any case, S/PDIF does not have separate clocks, this is
embedded into the data stream.
Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt
to configure TDA998x via the hw_params callback fails as the
hdmi_codec_daifmt is left initialised to zero.
Since the S/PDIF DAI will only be used by S/PDIF, prepare the
hdmi_codec_daifmt structure for this format.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add an entry to the quirks-table to for usb-audio to recognize the
Microbook II (although it only exposes vendor interfaces). A simple boot
quirk is also implemented to set up the sample rate and make sure that
no audio urbs are sent before the device is ready.
This patch only provides audio playback and capture at 96kHz sample
rate. Notice the following shortcomings:
- The sample rate is currently hardcoded to 96k although the device also
supports 48k and 44.1k.
- The various mixer controls of the MicroBook are not made available.
- The keep-iface control should be on by default because the device
shuts down whenever the altsetting is reset which is usually unwanted.
(I don't know the best way to do this)
- The communication format used by the MicroBook for sample rate setting
and also other setup has been reverse engineered by looking at the
usbmon output while running the windows driver through virtualbox. In
this patch the first byte of every message is set to \0 while in the
observed communications the first byte acts as a "message-counter"
increasing its value with every message sent. Leaving it at \0 does
not seem to affect the device.
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a bug that prevents freeing the reset gpio on unloading
the module.
aic3x_i2c_probe is called when loading the module and it calls list_add
with a probably uninitialized list entry aic3x->list (next = prev = NULL)).
So even if list_del is called it does nothing and in the end the gpio_reset
is not freed. Then a repeated module probing fails silently because
gpio_request fails.
When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move
list_del to aic3x_i2c_remove because aic3x_remove may be called
multiple times without aic3x_i2c_remove being called which leads to
a NULL pointer dereference.
Signed-off-by: Philipp Puschmann <philipp.puschmann@emlix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
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Merge tag 'asoc-v5.1-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More changes for v5.1
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
Dummy write in capture master mode is used to gate
bus clocks. This write is useless in slave mode
as the clocks are not managed by slave.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clocks do not need to be released on driver removal,
as this is already managed before.
Remove useless remove callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA configuration is not balanced on start/stop.
Move DMA configuration to trigger callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move counter handling to trigger start section
to manage multiple start/stop events.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S supports 16 bits data in 32 channel length.
However the expected driver behavior, is to
set channel length to 16 bits when data format is 16 bits.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because of regmap cache, interrupts may not be cleared
as expected.
Declare IFCR register as write only and make writings
to IFCR register unconditional.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_CROS_EC_CODEC depends on MFD_CROS_EC.
Add that dependency to SND_SOC_SDM845 to fix unmet direct dependencies
warning.
Fixes: 74c6ecf419 (ASoC: qcom: Kconfig: select dmic for sdm845)
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Tested-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch enables the reuse of kbl_da7219_max98927 machine driver to
support max98373. The same machine driver is modified for cases where one
amplifier is swapped out with another. Most of the changes are about
renaming the codec and codec_dai names, with minor differences due to
support for 24 bits in one case and 16 in the other.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently each SSI unit 's busif mode/adinr/dalign address is
registered by: (in busif4 case)
RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80)
But according to user manual 41.1.4 Register Configuration
ssi9 4/5/6/7 busif mode/adinr/dalign register address
( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] )
are out of this rule.
This patch registers ssi9 4/5/6/7 mode/adinr/dalign register
as single register, and access these registers in case of
SSI9 BUSIF 4/5/6/7.
Fixes: commit 8c9d750333 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In data blocks of common isochronous packet for MOTU devices, PCM
frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described
in IEC 61883-6. The frames are not aligned to quadlet.
For capture PCM substream, ALSA firewire-motu driver constructs PCM
frames by reading data blocks byte-by-byte. However this operation
includes bug for lower byte of the PCM sample. This brings invalid
content of the PCM samples.
This commit fixes the bug.
Reported-by: Peter Sjöberg <autopeter@gmail.com>
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 4641c93940 ("ALSA: firewire-motu: add MOTU specific protocol layer")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I set 10 seconds for the timeout of the i915 audio component binding
with a hope that recent machines are fast enough to handle all probe
tasks in that period, but I was too optimistic. The binding may take
longer than that, and this caused a problem on the machine with both
audio and graphics driver modules loaded in parallel, as Paul Menzel
experienced. This problem haven't hit so often just because the KMS
driver is loaded in initrd on most machines.
As a simple workaround, extend the timeout to 60 seconds.
Fixes: f9b54e1961 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the compressed stream implementation has acquired support for
multiple DAI links and compressed streams it has become harder to
interpret messages in the kernel log. Add additional macros to include
the compressed DAI name in the log messages, allowing different streams
to be easily disambiguated.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, only a single compressed stream is supported per firmware.
Add support for multiple compressed streams on a single firmware, this
allows additional features like completely independent trigger words or
separate debug capture streams to be implemented.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the code slightly clearer and prepare things for the addition of
multiple compressed streams on a single DSP core.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for more refactoring add a helper function to strip the
padding from ADSP data.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq_get_irq_data() function doesn't return error pointers, it
returns NULL.
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A platform can have multiple sound cards for different audio paths.
Following is the print seen duirng device boot for jetson-xavier,
ALSA device list:
#0: nvidia,p2972-0000 at 0x3518000 irq 17
By looking at above, it is not very clear if the sound card is for
HDA. It becomes confusing when platform has registered multiple cards,
and platform model name is used for card.
This patch uses "nvidia,model" property mentioned in hda device tree
to get the card name. Since property is optional, legacy boards will
continue to use "tegra-hda". Custom name can be passed wherever needed.
This naming convention is conistent with the way sound cards are named
in general.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This addresses an issue pointed out by compiler warning:
sound/soc/samsung/odroid.c: In function ‘odroid_audio_probe’:
sound/soc/samsung/odroid.c:298:22: warning: ‘cpu_dai’ may be used
uninitialized in this function [-Wmaybe-uninitialized]
priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis");
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need 32ea33a044 ("mei: bus: export to_mei_cl_device for mei
client devices drivers") for the mei-hdcp patches.
References: https://lkml.org/lkml/2019/2/19/356
Signed-off-by: Daniel Vetter <daniel.vetter@intel.com>
Here are a few last-minute fixes for 5.0. The most significant one
is the OF-node refcount fix for ASoC simple-card, which could be
triggered on many boards. Another fix for ASoC core is for the
error handling in topology, while others are device-specific fixes
for Samsung and HD-audio.
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Merge tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few last-minute fixes for 5.0.
The most significant one is the OF-node refcount fix for ASoC
simple-card, which could be triggered on many boards. Another fix for
ASoC core is for the error handling in topology, while others are
device-specific fixes for Samsung and HD-audio"
* tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: simple-card: fixup refcount_t underflow
ASoC: topology: free created components in tplg load error
ALSA: hda/realtek: Disable PC beep in passthrough on alc285
ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5
ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI
Although qcom_snd_parse_of() tries to manage the of-node refcount,
there are still a few places that lead to the unblanced refcount in
the error code path. Namely,
- for_each_child_of_node() needs to unreference the iterator node if
aborting the loop in the middle,
- cpu, codec and platform node objects have to be unreferenced at each
iteration,
- platform and codec node objects have to be referred before jumping
to the error handling code that unreference them unconditionally.
This patch tries to address these by moving the assignment of platform
and codec node objects to the beginning of the loop and adding the
of_node_put() calls adequately.
Fixes: c25e295cd7 ("ASoC: qcom: Add support to parse common audio device nodes")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The apq8016 driver leaves the of-node refcount at aborting from the
loop of for_each_child_of_node() in the error path. Not only the
iterator node of for_each_child_of_node(), the children nodes referred
from it for codec and cpu have to be properly unreferenced.
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
In odroid_audio_probe() some OF nodes are left without reference count
decrease after use. Fix it by ensuring required of_node_calls() are done
before exiting probe.
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function simple_for_each_link() has a few missing places that
forgot unrefereing of-nodes after the use. The main do-while loop
may abort when loop=0, and this leaves the node object still
referenced. A similar leak is found in the error handling of NULL
codec that aborts the loop as well. Last but not least, the inner
for_each_child_of_node() loop may abort in the middle, and this leaks
the refcount of the iterator node.
This patch addresses these missing refcount issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We forgot to unreference the platform node object obtained from
of_get_child_by_name(). This leads to the unbalance of node
refcount.
Fixes: e0ae225b7e ("ASoC: simple-card: support platform in dts parse")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The node obtained from of_find_node_by_path() has to be unreferenced
after the use, but we forgot it for the root node.
Fixes: f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support")
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, compressed buffers can only be specified in the XM memory
region. There is no reason to have such a restriction with the newer
meta-data based way of specifying the buffers, so remove it.
Signed-off-by: Andrew Ford <aford@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a compressed stream is restarted after getting an error, the cached
error value will still be used on the next pointer request, preventing
the stream from starting. Resolve this by ensuring the error status is
updated on trigger start.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Also contains the prep work in the component helpers plus adjustements
for the snd-hda/i915 component interface.
Plus one small static inline in the drm_hdcp.h header that both i915
and mei_hdcp will need.
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Merge tag 'topic/mei-hdcp-2019-02-19' of git://anongit.freedesktop.org/drm/drm-intel into drm-intel-next-queued
Prep patches + headers for the mei-hdcp/i915 component interfaces
Also contains the prep work in the component helpers plus adjustements
for the snd-hda/i915 component interface.
Plus one small static inline in the drm_hdcp.h header that both i915
and mei_hdcp will need.
Signed-off-by: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
From: Daniel Vetter <daniel.vetter@ffwll.ch>
Link: https://patchwork.freedesktop.org/patch/msgid/20190219071619.GA11016@phenom.ffwll.local
We forgot to unreference the node when aborting from the loop of
for_each_child_of_node() in snd_pmac_tumbler_init(). This leads to
unbalanced node refcount. Fix it by adding the missing of_node_put()
call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to unreference a node obtained via of_find_node_by_name()
after its usage.
Reviewed-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ac97_of_get_child_device() take the refcount of the node explicitly
via of_node_get(), but this leads to an unbalance. The
for_each_child_of_node() loop itself takes the refcount for each
iteration node, hence you don't need to take the extra refcount
again.
Fixes: 2225a3e6af ("ALSA: ac97: add codecs devicetree binding")
Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ADCs are sleeping when the SLEEP bit is set and running when it's
cleared, so the bit should be inverted.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio
interfaces" and the driver will try to register there multiple times
same platform device for the secondary FIFO, which of course fails
miserably. To fix this we derive the secondary platform device name
from the primary device name. The secondary device name will now
be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec".
The fixed platform_device_id table entry is removed as the secondary
device name is now dynamic and device/driver matching is done through
driver_override.
Reported-by: Marek Szyprowski <m.szyprowski@samsung.com>
Suggested-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This fixes unregistration of the secondary platform device so all
resources are properly released. Additionally the removal sequence
is corrected so it is in reverse order comparing to probe sequence.
The test against NULL priv->pdev_sec is removed as it is not necessary.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add FE DAI link to support parallel playback on 2 ports
simultaneously.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch enables support for GeminiLake with the DA7219 codec and
MAX98357A amplifier. To avoid duplicating code, the existing machine
driver for ApolloLake is reused with only changes in hardware
connectivity (SSP2 for DA7219 and SSP1 for MAX98357A).
The dailinks are directly modified in this patch. Using a helper would
be nicer, but it'll be done in a follow-up step with validation done
across multiple machine drivers.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A64 datasheet lists the supply rail for the headphone amp's charge
pump as "CPVDD". cpvdd-supply is the name of the property for this power
rail specified in the device tree bindings. "HPVCC" was the name used in
the A33 datasheet for the same function.
Rename the supply so it matches the datasheet, bindings, and the subject
from the original commit.
Fixes: ca0412a057 ("ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If platform_data is NULL add reading of optional adi,micbias
property from DT. If adi,micbias is not set keep the default
value for micbias.
Signed-off-by: Bogdan Togorean <bogdan.togorean@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should free "w" on the error path.
Fixes: 199ed3e81c ("ASoC: dapm: fix use-after-free issue with dailink sname")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With old DTS there will be missing DAPM routes linking BE with CODECs.
Add those routes in the card driver so sound works properly on Odroid
XU3/4 also without DTS updates enabling the secondary PCM.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warning:
sound/soc/codecs/wm8741.c:371:5: warning:
symbol 'wm8741_mute' was not declared. Should it be static?
Fixes: 36b1599340 ("ASoC: wm8741: Add digital mute callback")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to
wrong register, correct it.
Fixes 43d24e76b6 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Ackedy-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
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Merge tag 'asoc-fix-v5.0-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
Signed-off-by: Wen Yang <yellowriver2010@hotmail.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Shawn Guo <shawnguo@kernel.org>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Pengutronix Kernel Team <kernel@pengutronix.de>
Cc: NXP Linux Team <linux-imx@nxp.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-arm-kernel@lists.infradead.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
commit da215354eb ("ASoC: simple-card: merge simple-scu-card")
merged simple-card and simple-scu-card. Then it had refcount
underflow bug. This patch fixup it.
We will get below error without this patch.
OF: ERROR: Bad of_node_put() on /sound
CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514
Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT)
Workqueue: events deferred_probe_work_func
Call trace:
dump_backtrace+0x0/0x150
show_stack+0x24/0x30
dump_stack+0xb0/0xec
of_node_release+0xd0/0xd8
kobject_put+0x74/0xe8
of_node_put+0x24/0x30
__of_get_next_child+0x50/0x70
of_get_next_child+0x40/0x68
asoc_simple_card_probe+0x604/0x730
platform_drv_probe+0x58/0xa8
...
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_get_child_by_name() takes a reference we'll need to drop
later so when we substitute in top we need to take a reference
as well as just assigning.
Without this patch we hit the following error:
[ 1.246852] OF: ERROR: Bad of_node_put() on /sound-wm8524
[ 1.262261] Hardware name: NXP i.MX8MQ EVK (DT)
[ 1.266807] Workqueue: events deferred_probe_work_func
[ 1.271950] Call trace:
[ 1.274406] dump_backtrace+0x0/0x158
[ 1.278074] show_stack+0x14/0x20
[ 1.281396] dump_stack+0xa8/0xcc
[ 1.284717] of_node_release+0xb0/0xc8
[ 1.288474] kobject_put+0x74/0xf0
[ 1.291879] of_node_put+0x14/0x28
[ 1.295286] __of_get_next_child+0x44/0x70
[ 1.299387] of_get_next_child+0x3c/0x60
[ 1.303315] simple_for_each_link+0x1dc/0x230
[ 1.307676] simple_probe+0x80/0x540
[ 1.311256] platform_drv_probe+0x50/0xa0
This patch is based on an earlier version posted by Kuninori Morimoto
and commit message includes explanations from Mark Brown.
https://patchwork.kernel.org/patch/10814255/
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when playing sound with different sample rates actual
sample rate will be determined by audio stream which starts first
on either primary or secondary PCM. The audio root clock will be
configured appropriately only for the first stream. As the hardware
is limited to same sample rate on both interfaces we need to disallow
streams with different sample rates. It is done by this patch by
returning error in FE hw_params if there is already active stream
running with different sample rate.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_configure_clock':
sound/soc/stm/stm32_sai_sub.c:902:11: warning:
variable 'mask' set but not used [-Wunused-but-set-variable]
sound/soc/stm/stm32_sai_sub.c:902:6: warning:
variable 'cr1' set but not used [-Wunused-but-set-variable]
It's not used any more after 8307b2afd3 ("ASoC: stm32: sai: set sai as
mclk clock provider")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology resources are no longer needed if any element failed to load.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warnings:
sound/soc/codecs/cs35l36.c:135:20: warning:
symbol 'cs35l36_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:248:6: warning:
symbol 'cs35l36_readable_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:398:6: warning:
symbol 'cs35l36_precious_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:410:6: warning:
symbol 'cs35l36_volatile_reg' was not declared. Should it be static?
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sdm845 uses dmic on EC so it should select CROS_EC_CODEC.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to RM SPDIF STC SYSCLK_DF field is 9-bit wide, values
being in 0..511 range. Use a proper type to handle sysclk_df.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The on-chip PLL can be disabled if on the MCLKI pin we have an external
clock at 512 x fs. This clock can be used as direct internal clock for
ADCs or DACs.
To support this, we add an extra clock id that can be configured
using the set_sysclk() callback.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver only supports DPS_A for DAC, which is configured at probe.
This patch adds support for DSP_A and I2S modes by using the set_fmt()
callback.
A trivial break is also removed from a case's default branch.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By default, the codec starts to interpret the left (first) channel on
the falling edge (low polarity) of LRCLK. However, for DSP_A, the left
channel needs to start on the rising edge of LRCLK. This patch fixes
this channel swap by toggling the bit which selects the LRCLK polarity.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DACs and ADCs on ad193x codecs require a 32 bit slot size. We should
assure that no other size is used.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some ad193x codecs don't have ADCs, so they have no capture capabilities.
This way, we can use this driver in multicodec cards.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
At least some USB devices use (MSB-aligned) audio format larger
than the actual resolution of the device. In order to expose the
actual device resolution (bBitResolution), add extra field to the
procfs stream info interface.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
drm/i915 is tracking all wakeref owners with a cookie in order to
identify leaks. To that end, each rpm acquisition ops->get_power is
assigned a cookie which should be passed to ops->put_power to signify
its release (and removal from the list of wakeref owners). As snd/hda is
already using a bool to track current status of display_power extending
that to an unsigned long to hold the boolean cookie is a trivial
extension, and will quell all doubt that snd/hda is the cause of the
device runtime pm leaks.
v2: Keep using the power abstraction for local wakeref tracking.
v3: BUILD_BUG_ON impedance mismatch
Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jani Nikula <jani.nikula@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Mika Kuoppala <mika.kuoppala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20190213152109.16997-1-chris@chris-wilson.co.uk
When np is NULL i2s_pdata could also be NULL but i2s_pdata is now being
dereferenced without proper check. Fix this and shorten the error message
so we don't exceed 80 characters limit.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is now no users of this flag so remove it together with
related code. The chan_name field of snd_dmaengine_dai_dma_data
data structure is not removed as it is still in use by the PXA
platform.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous implementatation was restrictive with regards to
BCLK rates for slave mode where the driver would not allow rates
the codec couldn't provide itself as clock master. The codec
is able to automatically determine and handle whatever rate is
provided so this restriction isn't necessary for slave mode. The
code was also flawed with regards to setting of the frame offset
as using rx_mask to explicitly set the offset has the knock on
effect of impacting the min and max channels for the codec, in
soc_pcm_hw_params() through the call to
soc_pcm_codec_params_fixup().
With this update, the driver now only limits frame size if codec
is clock master, and dynamically determines the BCLK offset
relating to WCLK using the tx_mask for slot offset along with the
slot width provided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously the driver would default the BCLK periods per WCLK to
64, to cover all possible non-TDM scenarios when the codec was
DAI clock master. However some devices require a lower BCLK rate
to operate correctly so with this in mind, this commit updates
the code to be more dynamic, with BCLK rate now based on SR and
word length provided to hw_params().
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Paul Cercueil
Andra Danciu
Arnaud Pouliquen
Ranjani Sridharan
Zubin Mithra
Olivier Moysan
S.j. Wang
Pierre-Louis Bossart
Jerome Brunet
Charles Keepax
Annaliese McDermond
Oleksandr Andrushchenko
Sugar Zhang
Colin Ian King
Mark Brown
Kuninori Morimoto
Baolin Wang
YueHaibing
Richard Sailer
Kailang Yang
Hans de Goede
Anders Roxell
kbuild test robot
Jian-Hong Pan
Wen Yang
Liam Girdwood
Randy Dunlap
Andrew F. Davis
Andy Shevchenko
Shunli Wang
Oder Chiou
Linus Torvalds
Takashi Iwai
Viorel Suman
Bernhard Rosenkraenzer
Jenny TC
Dave Airlie
Pankaj Bharadiya
Timo Wischer
Aditya Pakki
Andi Kleen
John Hsu
Guenter Roeck
Chris Chiu
Piotr Stankiewicz
Codrin Ciubotariu
Adam Thomson
Daniel Mack
Gustavo A. R. Silva
Sylwester Nawrocki
Stuart Henderson
Richard Fitzgerald
Wen Shi
Pi-Hsun Shih
Bard liao
Hui Wang
Shuming Fan
Tzung-Bi Shih
Takashi Sakamoto
Jaroslav Kysela
KaiChieh Chuang
Mylène Josserand
Kangjie Lu
Arnd Bergmann
Mariusz Ceier
Luca Ceresoli
Jiada Wang
Rander Wang
Joonas Lahtinen
Jonathan Hunter
Russell King
Manuel Reinhardt
Philipp Puschmann
Cheng-Yi Chiang
Dan Carpenter
Sameer Pujar
Daniel Vetter
Andrew Ford
Yong Zhi
Chen-Yu Tsai
Bogdan Togorean
Wei Yongjun
Wen Yang
Daniel Baluta
Jussi Laako
Chris Wilson