The recently added PM prepare and complete callbacks don't have the
sanity check whether the card instance has been properly initialized,
which may potentially lead to Oops.
This patch adds the azx_is_pm_ready() call in each place
appropriately like other PM callbacks.
Fixes: f5dac54d9d ("ALSA: hda: Separate runtime and system suspend")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210329113059.25035-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card power state change via snd_power_change_state() at the system
suspend/resume seems dropped mistakenly during the PM code rewrite.
The card power state doesn't play much role nowadays but it's still
referred in a few places such as the HDMI codec driver.
This patch restores them, but in a more appropriate place now in the
prepare and complete callbacks.
Fixes: f5dac54d9d ("ALSA: hda: Separate runtime and system suspend")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210329113059.25035-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Logitech ConferenceCam Connect is a compound USB device with UVC and
UAC. Not 100% reproducible but sometimes it keeps responding STALL to
every control transfer once it receives get_freq request.
This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419
Signed-off-by: Ikjoon Jang <ikjn@chromium.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found the alc_update_headset_mode() is not called on some machines
when unplugging the headset, as a result, the mode of the
ALC_HEADSET_MODE_UNPLUGGED can't be set, then the current_headset_type
is not cleared, if users plug a differnt type of headset next time,
the determine_headset_type() will not be called and the audio jack is
set to the headset type of previous time.
On the Dell machines which connect the dmic to the PCH, if we open
the gnome-sound-setting and unplug the headset, this issue will
happen. Those machines disable the auto-mute by ucm and has no
internal mic in the input source, so the update_headset_mode() will
not be called by cap_sync_hook or automute_hook when unplugging, and
because the gnome-sound-setting is opened, the codec will not enter
the runtime_suspend state, so the update_headset_mode() will not be
called by alc_resume when unplugging. In this case the
hp_automute_hook is called when unplugging, so add
update_headset_mode() calling to this function.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210320091542.6748-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found a recording issue on a Dell AIO, users plug a headset-mic and
select headset-mic from UI, but can't record any sound from
headset-mic. The root cause is the determine_headset_type() returns a
wrong type, e.g. users plug a ctia type headset, but that function
returns omtp type.
On this machine, the internal mic is not connected to the codec, the
"Input Source" is headset mic by default. And when users plug a
headset, the determine_headset_type() will be called immediately, the
codec on this AIO is alc274, the delay time for this codec in the
determine_headset_type() is only 80ms, the delay is too short to
correctly determine the headset type, the fail rate is nearly 99% when
users plug the headset with the normal speed.
Other codecs set several hundred ms delay time, so here I change the
delay time to 850ms for alc2x4 series, after this change, the fail
rate is zero unless users plug the headset slowly on purpose.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210320091542.6748-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.
Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c79 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a lot of mostly platform specific fixes here, the only one which
is generic is a fix for regressions on devices with more complex
clocking support with simple-card. There's also a few new device IDs
and platform quirks.
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Merge tag 'asoc-fix-v5.12-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.12
Quite a lot of mostly platform specific fixes here, the only one which
is generic is a fix for regressions on devices with more complex
clocking support with simple-card. There's also a few new device IDs
and platform quirks.
With commit 1e30f642cf ("ASoC: simple-card-utils: Fix device module clock")
simple-card-utils can control MCLK clock for rate updates or enable/disable.
But this is breaking some platforms where it is expected that codec drivers
would actually handle the MCLK clock. One such example is following platform.
- "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"
In above case codec, wm8904, is using internal PLL and configures sysclk
based on fixed MCLK input. In such cases it is expected that, required PLL
output or sysclk, is just passed via set_sysclk() callback and card driver
need not actually update MCLK rate. Instead, codec can take ownership of
this clock and do the necessary configuration.
So the original commit is reverted and codec driver for rt5659 is updated
to fix my board which has this codec.
Sameer Pujar (2):
ASoC: simple-card-utils: Do not handle device clock
ASoC: rt5659: Update MCLK rate in set_sysclk()
sound/soc/codecs/rt5659.c | 5 +++++
sound/soc/generic/simple-card-utils.c | 13 +++++++------
2 files changed, 12 insertions(+), 6 deletions(-)
--
2.7.4
The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316094236.89028-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We do some IO operations in the snd_soc_component_set_jack callback
function and snd_soc_component_set_jack() will be called when soc
component is removed. However, we should not access SoundWire registers
when the bus is suspended.
So set regcache_cache_only(regmap, true) to avoid accessing in the
soc component removal process.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Simple-card/audio-graph-card drivers do not handle MCLK clock when it
is specified in the codec device node. The expectation here is that,
the codec should actually own up the MCLK clock and do necessary setup
in the driver.
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 1e30f642cf ("ASoC: simple-card-utils: Fix device
module clock"). The original patch ended up breaking following platform,
which depends on set_sysclk() to configure internal PLL on wm8904 codec
and expects simple-card-utils to not update the MCLK rate.
- "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"
It would be best if codec takes care of setting MCLK clock via DAI
set_sysclk() callback.
Reported-by: Michael Walle <michael@walle.cc>
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Fixes: 1e30f642cf ("ASoC: simple-card-utils: Fix device module clock")
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP EliteBook 840 G8 Notebook PC is using ALC236 codec which is
using 0x02 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316074626.79895-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP EliteBook 840 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316065452.75659-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When node is removed from IEEE 1394 bus, any transaction fails to the node.
In the case, ALSA dice driver doesn't stop isochronous contexts even if
they are running. As a result, null pointer dereference occurs in callback
from the running context.
This commit fixes the bug to release isochronous contexts always.
Cc: <stable@vger.kernel.org> # v5.4 or later
Fixes: e9f21129b8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210312093407.23437-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we found the micmute led init state is not correct after
freshly installing the ubuntu linux on a Lenovo AIO machine. The
internal mic is not muted, but the micmute led is on and led mode is
'follow mute'. If we mute internal mic, the led is keeping on, then
unmute the internal mic, the led is off. And from then on, the
micmute led will work correctly.
So the micmute led init state is not correct. The led is controlled
by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the
patch_realtek, the gpio data is set to 0x4 initially and the led is
on with this data. In the hda_generic, the led_value is set to
0 initially, suppose users set the 'capture switch' to on from
user space and the micmute led should change to be off with this
operation, but the check "if (val == spec->micmute_led.led_value)" in
the call_micmute_led_update() will skip the led setting.
To guarantee the led state will be set by the 1st time of changing
"Capture Switch", set -1 to the init led_value.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210312041408.3776-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The max boundary check while parsing dai ids makes
sound card registration fail after common up dai ids.
Fixes: cd3484f7f1 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY")
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When initialize cadence qspi controller, it is need to set cqspi
to the driver_data field of struct device, because it will be
used in function cqspi_remove/suspend/resume(). Otherwise, there
will be a crash trace as below when invoking these finctions.
Fixes: 31fb632b5d ("spi: Move cadence-quadspi driver to drivers/spi/")
Cc: stable@vger.kernel.org
Signed-off-by: Meng Li <Meng.Li@windriver.com>
Link: https://lore.kernel.org/r/20210311091220.3615-1-Meng.Li@windriver.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During testing John Stultz and Amit reported few array our bound issues
after enabling bound sanitizer
This patch series attempts to fix those!
changes since v1:
- make sure the wcd is not de-referenced without intialization
Srinivas Kandagatla (3):
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: codecs: wcd934x: add a sanity check in set channel map
sound/soc/codecs/wcd934x.c | 6 ++++++
sound/soc/qcom/sdm845.c | 6 +++---
2 files changed, 9 insertions(+), 3 deletions(-)
--
2.21.0
The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered
down, we need to check ADSPCS_CPA, the Current Power Active bit.
Fixes: 747503b181 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations")
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
set channel map can be passed with a channel maps, however if
the number of channels that are passed are more than the actual
supported channels then we would be accessing array out of bounds.
So add a sanity check to validate these numbers!
Fixes: a61f3b4f47 ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
WCD934x has only 13 RX SLIM ports however we are setting it as 16
in set_channel_map, this will lead to array out of bounds error!
Orignally caught by enabling USBAN array out of bounds check:
Fixes: 5caf64c633 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.
Fix this by properly using AFE_PORT_MAX macro.
Fixes: 1b93a88431 ("ASoC: qcom: sdm845: handle soundwire stream")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
Here is a patch series for reporting to user space jack and button events and
add the support for Capture. With some cleanups and fixes along the way.
Regards,
Lucas Tanure
Lucas Tanure (12):
ASoC: cs42l42: Fix Bitclock polarity inversion
ASoC: cs42l42: Fix channel width support
ASoC: cs42l42: Fix mixer volume control
ASoC: cs42l42: Don't enable/disable regulator at Bias Level
ASoC: cs42l42: Always wait at least 3ms after reset
ASoC: cs42l42: Remove power if the driver is being removed
ASoC: cs42l42: Disable regulators if probe fails
ASoC: cs42l42: Provide finer control on playback path
ASoC: cs42l42: Set clock source for both ways of stream
ASoC: cs42l42: Add Capture Support
ASoC: cs42l42: Report jack and button detection
ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called
Richard Fitzgerald (3):
ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT
ASoC: cs42l42: Only start PLL if it is needed
ASoC: cs42l42: Wait for PLL to lock before switching to it
sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++----------------
sound/soc/codecs/cs42l42.h | 41 +++-
2 files changed, 282 insertions(+), 196 deletions(-)
--
2.30.1
In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans
up a stray header and some Kconfig entries for the codec that
were missed in the process.
Fixes: 61fbeb5dcb (ASoC: remove sirf prima/atlas drivers)
Signed-off-by: Peter Robinson <pbrobinson@gmail.com>
Cc: Arnd Bergmann <arnd@arndb.de>
Cc: Mark Brown <broonie@kernel.org>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these
controls are incorrectly toggling the first bit of the register, which
is part of the FS_RATE field.
Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX"
control, which is to use SND_SOC_NOPM as the register and use an enum in
the shift field instead.
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
An interface can have multiple decimators enabled, so loop over all active
decimators. Otherwise only one channel will be unmuted, and other channels
will be zero. This fixes recording from dual DMIC as a single two channel
stream.
Also remove the now unused "active_decimator" field.
Fixes: 908e6b1df2 ("ASoC: codecs: lpass-va-macro: Add support to VA Macro")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
This delay is part of the power-up sequence defined in the datasheet.
A runtime_resume is a power-up so must also include the delay.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Many systems do not use ACPI and hence do not provide a DMI table. On
non-ACPI systems a warning, such as the following, is printed on boot.
WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name!
The variable 'dmi_available' is not exported and so currently cannot be
used by kernel modules without adding an accessor. However, it is
possible to use the function is_acpi_device_node() to determine if the
sound card is an ACPI device and hence indicate if we expect a DMI table
to be present. Therefore, call is_acpi_device_node() to see if we are
using ACPI and only parse the DMI table if we are booting with ACPI.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We only unregister the platform device during the .remove operation,
but if the probe fails we will never reach this sequence.
Suggested-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Fixes: dd96daca6c ("ASoC: SOF: Intel: Add APL/CNL HW DSP support")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
Here is a series of rt5640/rt5651 volume-control fixes which I wrote
while working on a bytcr-rt5640 UCM profile patch-series adding
hardware-volume control to devices using this UCM profile.
The UCM series will also work on older kernels, but it works best on
kernels with this series applied, giving e.g. finer grained volume
control and support for hardware muting the outputs.
Regards,
Hans
Hans de Goede (5):
ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control
ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback
Volume'
ASoC: Intel: bytcr_rt5640: Add used AIF to the components string
sound/soc/codecs/rt5640.c | 106 +++++++++++++++++++++++---
sound/soc/codecs/rt5640.h | 4 +
sound/soc/codecs/rt5651.c | 4 +-
sound/soc/intel/boards/bytcr_rt5640.c | 11 ++-
4 files changed, 111 insertions(+), 14 deletions(-)
--
2.30.1
Most steps in this table are steps of 3dB (300 centi-dB), so we can
simplify the table.
This not only reduces the amount of space it takes inside the kernel,
this also makes alsa-lib's mixer code actually accept the table, where
as before this change alsa-lib saw the "ADC PGA Gain" control as a
control without a dB scale.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has
the following bit field definitions:
| BITS | FIELD | RW | RESET | DEFINITION |
| 15 | RSVD | RO | 0x0 | Reserved |
| 14 | RSVD | RW | 0x1 | Reserved |
| 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode |
| 11:10 | RSVD | RO | 0x0 | Reserved |
| 9:8 | LBI_RESP | RW | 0x1 | Integrator Response |
| 7:6 | RSVD | RO | 0x0 | Reserved |
| 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode |
| 4:1 | RSVD | RO | 0x0 | Reserved |
| 0 | EN | RW | 0x0 | Enable/Disable AVC |
The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED. It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0. While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.
The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'. Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.
Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.
[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf
Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev
Signed-off-by: Mark Brown <broonie@kernel.org>
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b5 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b5 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the
byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX
models since these use almost the same settings.
While doing this I accidentally also copied and kept the non-standard
OVCD_TH_1500UA setting used on those models. This too low threshold is
causing headsets to often be seen as headphones (without a headset-mic)
and when correctly identified it is causing ghost play/pause
button-presses to get detected.
Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA
setting, fixing these problems.
Fixes: fbdae7d6d0 ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
While working on adding hardware-volume control support to the UCM
profile for the rt5672 and on adding LED trigger support to the
rt5670 codec driver. I hit / noticed a couple of issues this series
fixes these issues.
Regards,
Hans
Hans de Goede (4):
ASoC: rt5670: Remove 'OUT Channel Switch' control
ASoC: rt5670: Remove 'HP Playback Switch' control
ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer
settings
ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control
sound/soc/codecs/rt5670.c | 110 +++++++++++++++++++++++++++++++++-----
sound/soc/codecs/rt5670.h | 9 ++--
2 files changed, 101 insertions(+), 18 deletions(-)
--
2.30.1
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 9208847774 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 08660086ef ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For reliable output-mute LED control we need a "DAC1 Playback Switch"
control. The "DAC Playback volume" control is the only control in the
path from the DAC1 data input to the speaker output, so the UCM profile
for the speaker output will have its PlaybackMixerElem set to "DAC1".
But userspace (pulseaudio) will set the "DAC1 Playback Volume" control to
its softest setting (which is not fully muted) while still showing the
speaker as being enabled at a low volume in the UI.
If we were to set the SNDRV_CTL_ELEM_ACCESS_SPK_LED on the "DAC1 Playback
Volume" control, this would mean then what pressing KEY_VOLUMEDOWN the
speaker-mute LED (embedded in the volume-mute toggle key) would light
while the UI is still showing the speaker as being enabled at a low
volume, meaning that the UI and the LED are out of sync.
Only after an _extra_ KEY_VOLUMEDOWN press would the UI show the
speaker as being muted.
The path from DAC1 data input to the speaker output does have
a digital mixer with DAC1's data as one of its inputs direclty after
the "DAC1 Playback Volume" control.
This commit adds an emulated "DAC1 Playback Switch" control by:
1. Declaring the enable flag for that mixers DAC1 input as well as the
"DAC1 Playback Switch" control both as SND_SOC_NOPM controls.
2. Storing the settings of both controls as driver-private data
3. Only clearing the mute flag for the DAC1 input of that mixer if the
stored values indicate both controls are enabled.
This is a preparation patch for adding "audio-mute" LED trigger support.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_DAPM_MIXER declaration for "Sto1 ADC MIXL" and "Sto1 ADC MIXR"
was using the mute bits from the RT5670_STO1_ADC_DIG_VOL control as mixer
master mute bits.
But these bits are already exposed to userspace as controls as part of the
"ADC Capture Volume" / "ADC Capture Switch" control pair:
SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL,
RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL,
RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
127, 0, adc_vol_tlv),
Both the fact that the mute bits belong to the same reg as the vol-ctrl
and the "Digital Mixer Path" diagram in the datasheet clearly shows that
these mute bits are not part of the mixer and having 2 separate controls
poking at the same bits is a bad idea.
Remove the master-mute bits settings from the "Sto1 ADC MIXL" and
"Sto1 ADC MIXR" DAPM widget declarations, avoiding these bits getting
poked from 2 different places.
This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there already
set the "ADC Capture Switch" as needed.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5670_L_MUTE_SFT and RT5670_R_MUTE_SFT bits (bits 15 and 7) of the
RT5670_HP_VOL register are set / unset by the headphones deplop code
run by rt5670_hp_event() on SND_SOC_DAPM_POST_PMU / SND_SOC_DAPM_PRE_PMD.
So we should not also export a control to userspace which toggles these
same bits.
This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there do not
use the "HP Playback Switch" control.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
