The ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC
and output through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
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 UX533FD with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
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 known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack
sensing and enable use of the internal microphone on this laptop
X542UN. However, it's ALC294 so create a new fixup named
ALC294_FIXUP_ASUS_MIC to avoid confusion.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make unified suspend / resume helpers and call them from both the
runtime- and the system-PM callbacks for simplifying code.
There are slight changes of call orders, but there shouldn't be any
functional difference after refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In an initial commit, 'SYNC_STATUS' register is referred to get
clock configuration, however this is wrong, according to my local
note at hand for reverse-engineering about packet dump. It should
be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config()
is correctly programmed.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e1 ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Users reported a mute LED regression on Lenovo X1 Carbon, the root
cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE
to this machine, then the machine can't apply the fixup of
ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup
together.
Fixes: c4cfcf6f42 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Driver rewritten, assign copyright notice and change module author
as original one remains silent and I want to be notified about bugs.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sound capture and line bypass currently do not work as well as
some mixer controls. Fix that by building proper audio paths and
adjusting volume controls to match datasheet.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop "Common NI Values Table" and calculate LRCLK divider, then
add allowed rate constraints based on master clock frequency.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement set_bias_level to drive shutdown bit, so device is
put to sleep when unused.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will enable headset button for new Chrome platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend some structs to add the support for jack button changes.
Now snd_hda_jack_add_kctl() receives two more arguments: the jack type
and the jack keymaps. Both are optional, and when zero are passed,
the function behaves just like before.
For reporting button state changes, you'd need to update
jack->button_state bits accordingly, typically in the jack callback.
Then the value OR'ed with button_state and the jack plug state is
passed to snd_jack_report().
Note that currently the code assumes only the one-shot button events,
i.e. it tries to send the button release soon after sending the button
event. If a driver really supports the button release handling by
itself, we may need to introduce some flag to control this behavior in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing the callee to evaluate the associated jack information
and the unsolicited event data, add the new fields to
hda_jack_callback. They can be used, for example, to retrieve the
headset button state in the callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If it plugged headphone or headset into the jack, then
do the reboot, it will have a chance to cause headphone no sound.
It just need to run the headphone mode procedure after boot time.
The issue will be fixed.
It also suitable for ALC234 ALC274 and ALC294.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek codec ALC3277 is 100% compatible with the codec RT5660
in I2S mode. And on the Dell IoT platform, the codec is ALC3277,
and the HID of the codec in the BIOS is 10EC3277, so adding this
ID to the ACPI match table.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
For the FSL ASoC card, the full node names appear to be "ssi", "esai",
and "sai", so there's not any reason to use strstr and of_node_name_eq
can be used instead.
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.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
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD platform device acp_audio_dma can only be created by parent PCI
device driver (drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c). Pass struct
device of the parent to snd_pcm_lib_preallocate_pages() so
dma_alloc_coherent() can use correct dma_ops. Otherwise, it will
use default dma_ops which is nommu_dma_ops on x86_64 even when
IOMMU is enabled and set to non passthrough mode.
Though platform device inherits some dma related fields during its
creation in mfd_add_device(), we can't simply pass its struct device
to snd_pcm_lib_preallocate_pages() because dma_ops is not among the
inherited fields. Even it were, drivers/iommu/amd_iommu.c would
ignore it because get_device_id() doesn't handle platform device.
This change shouldn't give us any trouble even struct device of the
parent becomes null or represents some non PCI device in the future,
because get_dma_ops() correctly handles null struct device or uses
the default dma_ops if struct device doesn't have it set.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Gnawty model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3
for the mclk, just like the Clapper and Swanky models.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case of
the Gnawty that was breaking audio support since we were not using the
right clock in the cht_bsw_max98090_ti machine driver.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201787
Cc: stable@vger.kernel.org
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-and-tested-by: Jaime Pérez <19.jaime.91@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>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
A couple of open coded iterating thru the child node names are converted
to use for_each_child_of_node() instead.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert soundbus uevent and sysfs OF node name and device type usage to
use printf specifier and helper functions instead of directly accessing
the name and type pointers. This will allow the eventual removal of the
pointers.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
Cc: "David S. Miller" <davem@davemloft.net>
Cc: sparclinux@vger.kernel.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues
with the input from external microphone. The issue can be fixed by
the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer AIO Veriton Z4660G with ALC286 codec has issue with the input
from external microphones connecting via 'Front Mic' jack. The fixup
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of
the headset and fix the audio input issue of external microphone.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer AIO Aspire C24-860 with ALC286 can't detect the headset
microphone. Just like another Acer AIO U27-880, it needs a different
pin value for 0x18 and the headset fixup to make headset mic work.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic
and internal mic not working either. It needs the similar quirk like
Sony laptops to fix headphone jack sensing and enables use of the
internal microphone.
Unfortunately jack sensing for the headset mic is still not working.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch added max98373_reset function to avoid amp software reset failure and code duplication.
Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed.
Chip revision ID is available when the amp is in the idle state which means software reset is completed well.
Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge audio-graph-card and audio-graph-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on audio-graph-card.
It is same logic with audio-graph-scu-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-scu-card didn't care about codec_conf
for multi DPCM case. This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge audio-graph-card and audio-graph-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on audio-graph-scu-card.
It is same logic with audio-graph-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links.
If sound card is caring only DPCM, link count = dai count,
but, if non DPCM case, link count != dai count.
Now, we want to merge audio-graph-card and audio-graph-scu-card,
then, we need to care both link / dai count more carefly
This patch cares it, and prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_get_dai_id() returns DAI ID, but it is based on
DT node's "endpoint" position.
Almost all cases 1 port has 1 endpoint, thus, it was no problem.
But in reality, port : endpoint = 1 : N, thus, counting endpoint
is BUG, it should based on "port" ID.
This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit c16015f36c ("ASoC: rsnd: add .get_id/.get_id_sub")
add new .get_id/.get_id_sub to indicate module ID/subID.
It is used for SSIU and CTU. In SSIU case, subID indicates BUSIF,
but register settings is based on SSIU ID.
OTOH, in CTU case, subID indicates CTU channel, and register settings
is based on it. This means regmap read/write function needs to care it.
This patch fixup this issue. It can't play MIXed sound without this
patch.
Fixes: c16015f36c ("ASoC: rsnd: add .get_id/.get_id_sub")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For TDM debug purpose, indicating Channel and Mode is very
useful. This patch indicate it if it has #define DEBUG
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tegra186 and Tegra194 contain the same codecs as earlier chips and can
be supported using the same patch function.
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recent devices support more than the 4 codecs that the AZX core will
probe by default. Probe up to 8 codecs to make sure all of them are
enumerated.
Suggested-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Clapper model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3
for the mclk, just like the Swanky model.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case of
the Clapper that was breaking audio support since we were not using the
right clock in the cht_bsw_max98090_ti machine driver.
Cc: stable@vger.kernel.org
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
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>
These boards are now fully ported to devicetree and make use of the
simple-card driver, so the platform specific machine driver can be
removed.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49ab ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Entry needed for ICL RVP w/ RT274
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We've got a regression report for some Thinkpad models (at least
T570s) which shows the too low speaker output volume. The bisection
leaded to the commit 61fcf8ece9 ("ALSA: hda/realtek - Enable Thinkpad
Dock device for ALC298 platform"), and it's basically adding the two
pin configurations for the dock, and looks harmless.
The real culprit seems, though, that the DAC assignment for the
speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14
to be coupled with DAC NID 0x03. When more pins are configured by the
commit above, the auto-parser changes the DAC assignment, and this
resulted in the regression.
As a workaround, just provide the fixed pin / DAC mapping table for
this Thinkpad fixup function. It's no generic solution, but the
problem itself is pretty much device-specific, so must be good
enough.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304
Fixes: 61fcf8ece9 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a series of patches for conversion to LEDs audio-mute
trigger. It's based on 4.20-rc3 to be an immutable branch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Signed-off-by: Young_X <YangX92@hotmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's similar to other AMD audio devices, it also supports D3, which can
save some power drain.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 67ec1072b0 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM
stream") fixes deadlock for non-atomic PCM stream. But, This patch
causes antother stuck.
If writer is RT thread and reader is a normal thread, the reader
thread will be difficult to get scheduled. It may not give chance to
release readlocks and writer gets stuck for a long time if they are
pinned to single cpu.
The deadlock described in the previous commit is because the linux
rwsem queues like a FIFO. So, we might need non-FIFO writelock, not
non-block one.
My suggestion is that the writer gives reader a chance to be scheduled
by using the minimum msleep() instaed of spinning without blocking by
writer. Also, The *_nonblock may be changed to *_nonfifo appropriately
to this concept.
In terms of performance, when trylock is failed, this minimum periodic
msleep will have the same performance as the tick-based
schedule()/wake_up_q().
[ Although this has a fairly high performance penalty, the relevant
code path became already rare due to the previous commit ("ALSA:
pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now
this unconditional msleep appears only when using linked streams,
and this must be a rare case. So we accept this as a quick
workaround until finding a more suitable one -- tiwai ]
Fixes: 67ec1072b0 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream")
Suggested-by: Wonmin Jung <wonmin.jung@lge.com>
Signed-off-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the PCM core calls snd_pcm_unlink() always unconditionally
at closing a stream. However, since snd_pcm_unlink() invokes the
global rwsem down, the lock can be easily contended. More badly, when
a thread runs in a high priority RT-FIFO, it may stall at spinning.
Basically the call of snd_pcm_unlink() is required only for the linked
streams that are already rare occasion. For normal use cases, this
code path is fairly superfluous.
As an optimization (and also as a workaround for the RT problem
above in normal situations without linked streams), this patch adds a
check before calling snd_pcm_unlink() and calls it only when needed.
Reported-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default HDA sound card is registered with shortname "tegra-hda".
Same driver is used across tegra platforms and it is necessary to
distinguish between platforms to use platform specific settings from
userspace. One such example is, hdmi port on different platforms use
different alsa pcm device ID. For hdmi playback to work it should
open correct pcm device depending on the platform.
This patch applies shortname from first compatible string provided
in root node of device tree. Userspace then can use this card name
to apply specific settings.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge v4.20-rc4 into drm-next
Requested by Boris Brezillon for some vc4 fixes that are needed for future vc4 work.
Signed-off-by: Dave Airlie <airlied@redhat.com>
Now all relevant platform drivers are providing the LED audio trigger,
we can switch the mute LED control with the LED trigger, finally.
For the mic-mute LED trigger, a common fixup function,
snd_hda_gen_fixup_micmute_led(), is provided to be called for the
corresponding quirk entries. This sets up the capture sync hook with
ledtrig_audio_set() call appropriately.
For the mute LED trigger, which is done currently only for
thinkpad_acpi, the call is replaced with ledtrig_audio_set() as well.
Overall, the beauty of the new implementation is that the whole ugly
bindings with request_symbol() are dropped, and also that it provides
more flexibility to users.
One potential behavior change by this patch is that the mute LED enum
may be created on machines that actually have no LED device. In the
former code, we did test-call and abort binding if the test failed.
But with the LED-trigger binding, this test isn't possible, and the
actual check is done in the LED class device side. So it's the
downside of simpleness.
Also, note that the HD-audio codec driver doesn't select CONFIG_LEDS
and co by itself. It's supposed to be selected by the platform
drivers instead.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introducing a module param for wakeup_delay in order to
align with modeswitch_delay parameter. With this change, both
wakeup_delay and modeswitch_delay parameters can be passed
as module parameters.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On startup, applications such as PulseAudio or CRAS enable playback or
capture on all PCM devices to verify that configurations are correct,
and close them immediately. For DMICs, this can result in the clock
being turned off very quickly, which may not compatible with internal
state machine transition requirements.
This patch add a mode-switch delay which will prevent the clock from
being turned off without complying with manufacturer timing
specifications. While the DMIC clock may be controlled at a lower level,
be it with hardware or firmware, applying the delay during the
STOP_TRIGGER phase ensures that there is no race condition, e.g. with
the hardware/firmware turning off the clock earlier
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a callback for init ops on dai_link to create and setup jack.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add board specific dapm widgets so these widgets can be used
in the route.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the dismod is specified in the DT node, use the specified custom value
to configure the drive on state of the inactive TX slots.
If the dismod is not present or booted in legacy mode, the dismod is set
to low as it was the original behavior.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When McASP is master and the PDIR for the clock pins are configured as
outputs before the clocking is configured it will output whatever clock
is generated at the moment internally.
The clock will switch to the correct rate only when the we start the clock
generators.
To avoid this we must only set the pin as output after the clock is
configured and enabled.
AXR pins configured as outputs behaves somehow interesting as well:
when McASP is not enabled and the pin is selected as output it will not
honor the DISMOD settings for the inactive state, but will pull the pin
down.
Add a new bitfield and mark the pins there which needs to be output and
set the pins only at the time when they will behave correctly.
On stream stop configure the pins back to input which makes them to obey
the global pin configuration regarding to pull up/down.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow the guideline from the TRM:
Before starting, clear the respective transmitter and receiver status
registers
To avoid stale state stored in the status registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If mclk/sclk is already running, FW responds with IPC reply MCLK/SCLK
already running. Add these to the IPC reply lookup table.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add more meaning to the IPC replies for easy debugging. Replace the switch
case with a lookup table to lookup for the IPC replies and print in human
readable form.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide
useful string for the vendor and product names too. In order to share
the UCM with WD15, here we keep the profile_name same as the WD15.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current rsnd dvc.c is using flags to avoid duplicating register for
MIXer case. OTOH, commit e894efef9a ("ASoC: core: add support to card
rebind") allows to rebind sound card without rebinding all drivers.
Because of above patch and dvc.c flags, it can't re-register kctrl if
only sound card was rebinded, because dvc is keeping old flags.
(Of course it will be no problem if rsnd driver also be rebinded,
but it is not purpose of above patch).
This patch checks current card registered kctrl when registering.
In MIXer case, it can avoid duplicate register if card already has same
kctrl. In rebind case, it can re-register kctrl because card registered
kctl had been removed when unbinding.
This patch is updated version of commit b918f1bc7f ("ASoC: rsnd: DVC
kctrl sets once")
Reported-by: Nguyen Viet Dung <dung.nguyen.aj@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Nguyen Viet Dung <dung.nguyen.aj@renesas.com>
Cc: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lots of fixes here, the majority of which are driver specific but
there's a couple of core things and one notable driver specific one:
- A core fix for a DAPM regression introduced during the component
refactoring, we'd lost the code that forced a reevaluation of the
DAPM graph after probe (which we suppress during init to save lots
of recalcuation) and have now restored it.
- A core fix for error handling using the newly added
for_each_rtd_codec_dai_rollback() macro.
- A fix for the names of widgets in the newly introduced pcm3060
driver, merged as a fix so we don't have a release with legacy names.
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Merge tag 'asoc-v4.20-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.20
Lots of fixes here, the majority of which are driver specific but
there's a couple of core things and one notable driver specific one:
- A core fix for a DAPM regression introduced during the component
refactoring, we'd lost the code that forced a reevaluation of the
DAPM graph after probe (which we suppress during init to save lots
of recalcuation) and have now restored it.
- A core fix for error handling using the newly added
for_each_rtd_codec_dai_rollback() macro.
- A fix for the names of widgets in the newly introduced pcm3060
driver, merged as a fix so we don't have a release with legacy names.
This patch will enable ALC300.
[ It's almost equivalent with other ALC269-compatible ones, and
apparently has no loopback mixer -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device makes a loud buzzing sound when a headphone is inserted while
playing audio at full volume through the speaker.
Fixes: bbf8ff6b1d ("ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers")
Signed-off-by: Girija Kumar Kasinadhuni <gkumar@neverware.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The platform_device_register_full() function doesn't return NULL, it
returns error pointers.
Fixes: 7894a7e7ea ("ASoC: amd: create ACP3x PCM platform device")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have several Lenovo laptops with the codec alc285, when playing
sound via headphone, we can hear click/pop noise in the headphone,
if we let the headphone share the DAC of NID 0x2 with the speaker,
the noise disappears.
The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon.
I have tried to set preferred_dacs and override_conn, but neither of
them worked. Thanks for Kailang, he told me to invalidate the NID 0x3
through override_wcaps.
BugLink: https://bugs.launchpad.net/bugs/1805079
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both snd_ctl_add() and snd_ctl_replace() process the things in a
fairly similar way, and indeed the most of the codes can be unified.
This patch is a refactoring to consolidate the both functions to call
a single helper with an extra "mode" argument. There should be no
functional difference, except for one additional sanity check applied
now to snd_ctl_replace() (which was rather overlooking, IMO), too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for adding a user control element has some window opened
for race against the concurrent removal of a user element. This was
caught by syzkaller, hitting a KASAN use-after-free error.
This patch addresses the bug by wrapping the whole procedure to add a
user control element with the card->controls_rwsem, instead of only
around the increment of card->user_ctl_count.
This required a slight code refactoring, too. The function
snd_ctl_add() is split to two parts: a core function to add the
control element and a part calling it. The former is called from the
function for adding a user control element inside the controls_rwsem.
One change to be noted is that snd_ctl_notify() for adding a control
element gets called inside the controls_rwsem as well while it was
called outside the rwsem. But this should be OK, as snd_ctl_notify()
takes another (finer) rwlock instead of rwsem, and the call of
snd_ctl_notify() inside rwsem is already done in another code path.
Reported-by: syzbot+dc09047bce3820621ba2@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of sparc cs4231 driver code. Since
runtime->dma_area is managed by the PCM core helper, we shouldn't
release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of wss driver code. Since runtime->dma_area
is managed by the PCM core helper, we shouldn't release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the
MS-B120, in order for the headset mic to be properly detected.
They both use a single 3-way jack for both mic and headset with an
ALC283 codec, with the same pins used.
Cc: stable@vger.kernel.org
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on the built-in audio
of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to
the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Cc: <stable@vger.kernel.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_ac97_put_spsa() gets the bit shift value from the
associated private_value, but it extracts too much; the current code
extracts 8 bit values in bits 8-15, but this is a combination of two
nibbles (bits 8-11 and bits 12-15) for left and right shifts.
Due to the incorrect bits extraction, the actual shift may go beyond
the 32bit value, as spotted recently by UBSAN check:
UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7
shift exponent 68 is too large for 32-bit type 'int'
This patch fixes the shift value extraction by masking the properly
with 0x0f instead of 0xff.
Reported-and-tested-by: Meelis Roos <mroos@linux.ee>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, ALSA firewire-tascam driver queues events to notify
change of state of control surface to userspace via ALSA hwdep
interface.
This commit implements actual notification of the events. The events are
not governed by real time, thus no need to care underrun.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commits, ALSA firewire-tascam driver will allow userspace
applications to receive notifications about changes of device state,
transferred in tx isochronous packet. At present, all of drivers in ALSA
firewire stack have mechanism to notify change of status of packet
streaming, thus it needs to distinguish these two types of notification.
This commit is a preparation for the above.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Units of TASCAM FireWire series transfer image of states of the unit in
tx isochronous packets. Demultiplexing of the states from the packets
is done in software interrupt context regardless of any process context.
In a view of userspace applications, it needs to have notification
mechanism to catch change of the states.
This commit implements a queue to store events for the notification. The
image of states includes fluctuating data such as level of gain/volume
for physical input/output and position of knobs. Therefore the events
are queued corresponding to some control features only.
Furthermore, the queued events are planned to be consumed by userspace
applications via ALSA hwdep interface. This commit suppresses event
queueing when no applications open the hwdep interface.
However, the queue is maintained in an optimistic scenario, thus without
any care against overrrun. This is reasonable because target events are
useless just to handle PCM frames. It starts queueing when an usespace
application opens hwdep interface, thus it's expected to read the queued
events steadily.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a previous commit, ALSA firewire-tascam driver stores state image
from tx isochronous packets. This image includes states of knob, fader,
button of control surface, level of gain/volume of each physical
inputs/outputs, and so on. It's useful for userspace applications to
read whole of the image.
This commit adds a unique ioctl command for ALSA hwdep interface for the
purpose. For actual meaning of each bits in this image, please refer to
discussion in alsa-devel[1].
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140785.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Units of TASCAM FireWire series multiplex PCM frames and state of
control surface into the same tx isochronous packets. One isochronous
packet includes a part of the state in a quadlet data. An image of the
state consists of 64 quadlet data.
This commit demultiplexes the state from tx isochronous packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without under of overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The latency number is in usec for the pm_qos. Correct the calculation to
give us the time in usec
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit f986907c92 ("ASoC: audio-graph-card: add widgets and routing for
external amplifier support") added new function
asoc_graph_card_outdrv_event(), but the inserted position breaks
define area. This patch tidyup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
1 "simple" is enough on Kconfig help
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-scu-card driver is parsing codec position for DPCM
and consider DAI format. But, current operation is doing totally pointless,
because 1) asoc_simple_card_parse_daifmt() will be called not only for 1st
codec on current implementation, and it will be used as fixed format
2) it should be called for each CPU/Codec pair.
Let's tidyup asoc_simple_card_parse_daifmt() timing.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-scu-card.c is supporting "convert-rate/channels" which is
used for DPCM.
But, sound card might have multi codecs, and each codec might need
each convert-rate/channels.
This patch supports each codec's convert-rate/channles support.
top node convert-rate/channels will overwrite settings if exist.
It can't support each codec's convert-rate/channels if sound card had
multi codecs without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-scu-card.c is supporting "prefix" which is used to avoid
DAI naming conflict when CPU/Codec matching.
But, sound card might have multi sub-devices, and each codec might need
each prefix.
Now, ASoC is supporting snd_soc_of_parse_node_prefix(), let's support
it on audio-graph-scu-card, too. It is keeping existing DT style.
It can't support each codec's prefix if sound card had multi sub-devices
without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-scu-card.c is supporting "prefix" which is used to avoid
DAI naming conflict when CPU/Codec matching.
But, sound card might have multi sub-devices, and each codec might need
each prefix.
Now, ASoC is supporting snd_soc_of_parse_node_prefix(), let's support
it on audio-graph-scu-card, too. It is keeping existing DT style.
It can't support each codec's prefix if sound card had multi sub-devices
without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has snd_soc_of_parse_audio_prefix() to get codec_conf
settings from DT which is used to avoid DAI naming conflict when
CPU/Codec matching.
Currently, it is parsing from "top node",
but, we want to parse from "each sub node" if sound card had multi
cpus/codecs.
This patch adds new snd_soc_of_parse_node_prefix() to allow parsing
settings from selected node.
It is keeping existing snd_soc_of_parse_audio_prefix() by using macro.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amplifier may have assosicated regulator, so add a widget for it
and appropriate route.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the Allwinner A64 SoCs, the audio codec has a built-in headphone
amplifier. This amplifier has a power supply separate from the rest of
the analog audio circuitry, labeled cpvdd.
This patch adds a DAPM widget for this supply, and ties it to the
headphone amp widget.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_of_parse_routing() had "option" parameter
to consider error handling, but it is very pointless parameter.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils has asoc_simple_card_parse_convert() to parse
convert channel/rate for be_hw_params_fixup.
But, it is parsing from top of node.
If sound card had multi subnode, we need to parse it from each sub node.
This patch tidyup asoc_simple_card_parse_convert() to allow parsing
settings from each node.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If simple-card-utils accept NULL pointer on asoc_simple_card_xxx(),
each driver code will be more simple.
Let's accept NULL pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_clk_register() is used but only 1 user,
and very pointless code. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ssi.c only is using rsnd_ssi_is_dma_mode().
Let's move it as static function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_parse_connect_ssiu_compatible() is doing
- using rsnd_ssiu_id(), but we use it via rsnd_mod_id()
- we can break loop if rsnd_dai_connect() was called
This patch fixup these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to configure bit clock for secondary MI2S
TX interface.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change slot_width for quaternary TDM port to 16 and
update bclk rate for TDM and MI2S interfaces
accordingly.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason we have different mechanisms for passing data to
machine drivers. Use the solution used by Atom/SST and SOF instead of
using drv_data as done by Skylake.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The changes for HDaudio overlooked the fact that the machine drivers
used for Chromebooks rely on the dmic number information passed as
pdata.
Add dmic_num field to standard interface and use standard interface
instead of SKL-specific one.
Also clean-up pdata definition to remove fields that are no longer
used.
Fixes: 842bb5135f ('ASoC: Intel: use standard interface for Hdaudio machine driver')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The machine_quirk may return NULL which means the acpi entries should be
skipped and search for next matched entry is needed, here add return
check here and continue for NULL case.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Skylake driver currently has a set of problems supporting
load/unload modules. We need to make the HDaudio codec support
optional to help narrow down the issues.
Support for HDaudio codecs also leads to a Kconfig issue. We want the
hdac_hda codec to be compilable independently of Skylake (e.g. with
ALL_CODECS) but when Skylake is selected as built-in the hdac_hda
codec needs to use the same option due a a code dependency
Solve both problems by adding a user-selectable boolean Kconfig,
select HDAC_HDA as needed and make the HDaudio codec support in the
Skylake driver optional. Tests on a Chell Chromebook device without
HDaudio show no regression for speaker and HDMI playback.
This is submitted as an RFC to allow for comments and more validation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec can support any variation of bclk/fs master/slave configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove directly accessing device_node.type pointer and use the accessors
instead. This will eventually allow removing the type pointer.
Replace the open coded iterating over child nodes with
for_each_child_of_node() while we're here.
Cc: Johannes Berg <johannes@sipsolutions.net>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: linuxppc-dev@lists.ozlabs.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the pincfg assignment for the AE-5, which was
previously using the Recon3D pincfg's by mistake.
Fixes: d06feaf02f ("ALSA: hda/ca0132 - Add pincfg for AE-5")
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new PCI subsys ID for the ZxR, as found and tested by
other users. Without a way to know if any Z's use it as well, it keeps
the quirk of QUIRK_SBZ and goes through the HDA subsys test function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct declaration is not indented correctly. Fix this by replacing
spaces with a tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default implementation of regulator_set_load returns
REGULATOR_MODE_NORMAL, which is positive. [This was a bug which is
being fixed but the change is valid anyway -- bronie]
rt5663_i2c_probe should only do error handling when return value of
regulator_set_load is negative.
In this case, rt5663_i2c_probe should return error.
Also, consolidate err_irq into err_enable.
Fix the missing goto for temporary regmap and rt5663->regmap.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
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>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented too much by one level, fix this by
removing the extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented incorrectly. Fix this by adding in
the missing tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The goto statement is indented too much by one level, fix this by
removing the extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented too much by one level, fix this by
removing an extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to remove the node name pointer from struct device_node,
convert printf users to use the %pOFn format specifier.
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: Olivier Moysan <olivier.moysan@st.com>
Cc: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Cc: Maxime Coquelin <mcoquelin.stm32@gmail.com>
Cc: Alexandre Torgue <alexandre.torgue@st.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-stm32@st-md-mailman.stormreply.com
Cc: linux-arm-kernel@lists.infradead.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add regulator support to turn on cpvdd and avdd in probe.
If a regulator is not given from device tree, a dummy regulator will be
used.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AK4118A is a digital audio transceiver supporting 8 input channels
at 192kHz and with 24bits resolution.
It converts the S/PDIF signal to I2S format and is configurable over I2C.
This driver introduce a minimal support of the AK4118, like selecting the
input channel, reading input frequency and detecting some errors.
Datasheet is available here:
https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf
Signed-off-by: Adrien Charruel <adrien.charruel@devialet.com>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the current device datasheet (TI Lit # SLAS831D, revised
March 2018) the value written to the device's PAGE register to trigger
a complete register reset should be 0xfe, not 0xff. So go ahead and
update to the correct value.
Reported-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Tested-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/amd/raven/acp3x-pcm-dma.c: In function 'acp3x_dma_hw_params':
sound/soc/amd/raven/acp3x-pcm-dma.c:333:25: warning:
variable 'dma_buffer' set but not used [-Wunused-but-set-variable]
It never used since introduction in commit
8de1b5ed03 ("ASoC: amd: add acp3x system resume pm op")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license
compliance management.
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixed build errors.
- Implicit declaration of pci_enable_msi() & pci_disable_msi()
api's for openrisc architecture.
- type defaults to 'int' in declaration of 'module_pci_driver'
Enabled build for x86 architecture.
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch updates license to SPDX-License-Identifier
instead of verbose license text.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio map are possible in wrong state before card->instantiated has
been set to true. Imaging the following examples:
time 1: at the beginning
in:-1 in:-1 in:-1 in:-1
out:-1 out:-1 out:-1 out:-1
SIGGEN A B Spk
time 2: after someone called snd_soc_dapm_new_widgets()
(e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c)
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN A B Spk
time 3: routes added
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN -----> A -----> B ---> Spk
In the end, the path should be powered on but it did not. At time 3,
"in" of SIGGEN and "out" of Spk did not propagate to their neighbors
because snd_soc_dapm_add_path() will not invalidate the paths if
the card has not instantiated (i.e. card->instantiated is false).
To correct the state of audio map, recalculate the whole map forcely.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The machine driver fails to probe in next-20181113 with:
[ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered
[ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517
...
[ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040
[ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040
[ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22
[ 3.715148] omap-abe-twl6040: probe of sound failed with error -22
Bisect pointed to a merge commit:
first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master'
and a diff between a working kernel does not reveal anything which would
explain the change in behavior.
Further investigation showed that on the second try of loading fails
because the dai_link->platform is no longer NULL and it might be pointing
to uninitialized memory.
The fix is to move the snd_soc_dai_link and snd_soc_card inside of the
abe_twl6040 struct, which is dynamically allocated every time the driver
probes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when snd_pcm_hw_constraint_integer fails there is
a memory leak of i2s_data on the error return path. Fix this by
kfree'ing i2s_data before returning.
Detected by CoverityScan, CID#1475479 ("Resource leak")
Fixes: 0b87d6bcd6 ("ASoC: amd: add acp3x pcm driver dma ops")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAC output may be differential (default) or single-ended.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the initial commit [1], I added differential output of the codec as
separate `+` and `-` widgets:
OUTL+
OUTR+
OUTL-
OUTR-
Later, in the commit [2], I added a device tree property to configure the
output as single-ended or differential. Having this property, the `+` and
`-` separation in widgets seems for me confusing. There are no functional
benefits in such separation, so I find reasonable to get rid of it:
OUTL
OUTR
The new naming is more friendly for sound cards, and is better aligned with
other codec drivers in kernel.
Renaming the output widgets now should not be a problem from the backwards-
compatibility perspective, as the driver for PCM3060 is added into the
mainline very recently, and did not yet appear in any releases.
[1] commit 6ee47d4a8d ("ASoC: pcm3060: Add codec driver")
[2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended
output")
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
sizeof() when applied to a pointer typed expression gives the
size of the pointer, not that of the pointed data.
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Frontend dai_link id is used for closing ADM sessions.
During concurrent usecase when one session is closed,
it closes other ADM session associated with other usecase
too. Dai_link->id should always point to Frontend dai id.
Set cpu_dai id as dai_link id to fix the issue.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it
to the driver to avoid adding it to every dts.
Fixes: eda85d1fee ("ASoC: sun8i-codec: Add ADC support for a33")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x drivers can be built by selecting necessary kernel config option.
The patch enables build support of the same.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When system wide suspend happens, ACP will be powered off.
When system resumes, all the runtime configuration data for
ACP needs to be programmed again.
Added 'resume'pm call back to ACP pm ops.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x I2S (CPU DAI) can act in normal I2S and TDM modes.
Added support for TDM mode.
Desired mode can be selected from ASoC machine driver.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x has a i2s controller block for playback and capture.
This patch adds ACP3x i2s DAI operations.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x has a DMA controller to access system memory.
This controller transfers data from/to system memory
to/from the ACP internal FIFO.
The patch adds PCM driver DMA operations.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Whenever audio data equal to the I2S FIFO watermark level are
produced/consumed, interrupt is generated.
Acknowledge the interrupt.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM platform driver binds to the platform device created by ACP3x PCI
device. PCM driver registers ALSA DMA and CPU DAI components with ASoC
framework.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
