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>
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>
