The pointer to of_phandle_args passed to snd_soc_get_dai_name() and
of_xlate_dai_name() implementations is not modified. Since it is being
used only to translate passed OF node to a DAI name, it should not be
modified, so mark it as const for correctness and safer code.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20210221153024.453583-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
sound/soc/codecs/lpass-wsa-macro.c:958:6: style: Variable 'ret' is
reassigned a value before the old one has been
used. [redundantAssignment]
ret = wsa_macro_set_prim_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
sound/soc/codecs/lpass-wsa-macro.c:946:6: note: ret is assigned
ret = wsa_macro_set_mix_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
sound/soc/codecs/lpass-wsa-macro.c:958:6: note: ret is overwritten
ret = wsa_macro_set_prim_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
set_mix_interpolator_rate can return -EINVAL, add a test and bail on error.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210218222916.89809-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warnings:
sound/soc/qcom/qdsp6/q6afe.c:848:25: note: Assignment 'p=NULL',
assigned value is 0
struct q6afe_port *p = NULL;
^
sound/soc/qcom/qdsp6/q6afe.c:854:7: note: Null pointer dereference
if (p->token == token) {
^
sound/soc/qcom/qdsp6/q6afe.c:939:8: style: Redundant initialization
for 'wait'. The initialized value is overwritten before it is
read. [redundantInitialization]
wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:933:26: note: wait is initialized
wait_queue_head_t *wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:939:8: note: wait is overwritten
wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:1191:10: style: Redundant initialization
for 'port_id'. The initialized value is overwritten before it is
read. [redundantInitialization]
port_id = port->id;
^
sound/soc/qcom/qdsp6/q6afe.c:1186:14: note: port_id is initialized
int port_id = port->id;
^
sound/soc/qcom/qdsp6/q6afe.c:1191:10: note: port_id is overwritten
port_id = port->id;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210218222916.89809-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
sound/soc/qcom/lpass-hdmi.c:189:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/qcom/lpass-hdmi.c:186:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/qcom/lpass-hdmi.c:189:9: note: Returning identical
expression 'ret'
return ret;
^
sound/soc/qcom/lpass-hdmi.c:206:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/qcom/lpass-hdmi.c:203:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/qcom/lpass-hdmi.c:206:9: note: Returning identical
expression 'ret'
return ret;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210218222916.89809-2-pierre-louis.bossart@linux.intel.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
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
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>
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>
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
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>
The "OUT Channel Switch" control is a left over from code copied from
thr rt5640 codec driver.
With the rt5640 codec driver the output volume controls have 2 pairs of
mute bits:
bit 7, 15: Mute Control for Spk/Headphone/Line Output Port
bit 6, 14: Mute Control for Spk/Headphone/Line Volume Channel
Bits 7 and 15 are normal mute bits on the rt5670/5672 which are
controlled by 2 dapm widgets:
SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0,
&lout_l_enable_control),
SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0,
&lout_r_enable_control),
But on the 5670/5672 bit 6 is always reserved, where as bit 14 is
"LOUT Differential Mode" on the 5670 and also reserved on the 5672.
So the "OUT Channel Switch" control which is controlling bits 6+14
of the "LINE Output Control" register is bogus -> remove it.
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 "OUT Channel Switch" control.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt()
function rewrites the number of slots previously set by the
fsl_ssi_set_dai_tdm_slot() function to 2 by default.
To fix this, let's use the saved slot count value or, if TDM
is not used and the slot count is not set, proceed as before.
Fixes: 4f14f5c11d ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
