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>
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>
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 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
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>
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 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>
Jerome Brunet