To avoid the actual PLL settings to differ from the state expected by
the clock driver, the codec should only be fully reset before the clocks
are registered. But we also need to ensure that the software reset
happens at all before clock registration, as not all boards have a reset
GPIO.
Move the software reset from aic32x4_component_probe() to
aic32x4_probe() and reorder the reset and registration sequence:
1. Reset via GPIO (if available)
2. Reset via software
3. Register component
4. Register clocks
Note that aic32x4_component_probe() is only called after aic32x4_probe()
has finished, so the reset in aic32x4_component_probe() was happening too
late.
Signed-off-by: Matthias Schiffer <matthias.schiffer@ew.tq-group.com>
Link: https://lore.kernel.org/r/20200902133043.19504-2-matthias.schiffer@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
GPIO_ACTIVE_x flags are not correct in the context of interrupt flags.
These are simple defines so they could be used in DTS but they will not
have the same meaning:
1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE
2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING
Correct the interrupt flags, assuming the author of the code wanted some
logical behavior behind the name "ACTIVE_xxx", this is:
ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200908145954.4629-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the BCLK inversion for DSP modes
This is how it is defined by ASoC:
* BCLK:
* - "normal" polarity means signal is available at rising edge of BCLK
* - "inverted" polarity means signal is available at falling edge of BCLK
The adcx140 defines the BCLK edge based on coding type.
The PCM (DSP_A/B) should drive on rising and sample on falling edge, so
from ASoC pov, it is IB_NF. But from the codec pov if it is configured in
DSP mode, then the BCLK should not be inverted, defaults to the coding
standard.
For i2s, it is NB_NF from ASoC pov (drive on falling, sample on rising).
>From the codec's pov BCLK should not invert either, as this is the default
for the coding.
So, inversion must take the format into account:
IB_NF + DSP_A/B == the codec bclk inversion should be disabled
NB_NF + DSP_A/B == the codec bclk inversion should be enabled
NB_NF + I2S == the codec bclk inversion should be disabled
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200915190606.1744-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It was observed that if the device was active and register writes were
performed there were some unwanted behaviors particularly when writing
the word length and some filter options. So when writing to the device
the device should be placed in sleep mode and then exit sleep mode once
the register update is complete.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200915190606.1744-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to its datasheet, the digital gain should be -100 dB when
CHx_DVOL is 1 and 27 dB when CHx_DVOL is 255. But with the current
dig_vol_tlv, "Digital CHx Out Volume" shows 27.5 dB if CHx_DVOL is 255
and -95.5 dB if CHx_DVOL is 1. This commit fixes this bug.
Fixes: 689c7655b5 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Camel Guo <camelg@axis.com>
Link: https://lore.kernel.org/r/20200908090417.16695-1-camel.guo@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_SND_CTL_VALIDATION is set, accesses to extended bytes
control generate spurious error messages when the size exceeds 512
bytes, such as
[ 11.224223] sof_sdw sof_sdw: control 2:0:0:EQIIR5.0 eqiir_coef_5:0:
invalid count 1024
In addition the error check returns -EINVAL which has the nasty side
effect of preventing applications accessing controls from working,
e.g.
root@plb:~# alsamixer
cannot load mixer controls: Invalid argument
It's agreed that the control interface has been abused since 2014, but
forcing a check should not prevent existing solutions from working.
This patch skips the checks conditionally if CONFIG_SND_CTL_VALIDATION
is set and the byte array provided by topology is > 512. This
preserves the checks for all other cases.
Fixes: 1a3232d2f6 ('ASoC: topology: Add support for TLV bytes controls')
BugLink: https://github.com/thesofproject/linux/issues/2430
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200917103912.2565907-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The series reuses mt8183-da7219-max98357.c for supporting machine
driver with rt1015p speaker amplifier.
The 1st patch adds document for the new proposed compatible string.
The 2nd patch changes the machine driver to support "RT1015P" codec.
Tzung-Bi Shih (2):
ASoC: dt-bindings: mt8183-da7219: add compatible string for using
rt1015p
ASoC: mediatek: mt8183-da7219: support machine driver with rt1015p
.../bindings/sound/mt8183-da7219-max98357.txt | 1 +
sound/soc/mediatek/Kconfig | 1 +
.../mediatek/mt8183/mt8183-da7219-max98357.c | 40 +++++++++++++++++++
3 files changed, 42 insertions(+)
--
2.28.0.526.ge36021eeef-goog
Hi,
Changes since v1:
- Suffix the 2359296000 constant with 'u' to silence C90 warning
When j7200 SOM is connected to the CPB, the audio setup is a bit different:
Only 48KHz family have clock path, 44.1KHz is not supported.
Update the binding documentation and add support for the j7200 version of CPB
to the driver.
Regards,
Peter
---
Peter Ujfalusi (2):
ASoC: dt-bindings: ti,j721e-cpb-audio: Document support for j7200-cpb
ASoC: ti: j721e-evm: Add support for j7200-cpb audio
.../bindings/sound/ti,j721e-cpb-audio.yaml | 92 ++++++++++++++-----
sound/soc/ti/j721e-evm.c | 11 +++
2 files changed, 81 insertions(+), 22 deletions(-)
--
Peter
Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki
Enable support of pm_runtime on STM32 SAI driver to allow
SAI power state monitoring.
pm_runtime_put_autosuspend() is called from ASoC framework
on pcm device close.
The pmdown_time delay is available in runtime context, and may be set
in SAI driver to take into account shutdown delay on playback.
However, this shutdown delay is already handled in the DAPMs
of the audio codec linked to SAI CPU DAI.
So, the choice is made, not to support this delay on CPU DAI side.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200911081507.7276-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS IP on SoCs like SM8250 has Digital Codec part integrated into it.
This ports are exposed in Q6DSP as Codec ports. This patchset adds
support to those q6afe ports along with q6routing and q6afe-dai.
This patchset has been tested along with other patches on
Qualcomm Robotics RB5 Platform with Soundwire and WSA8815 Codec.
Thanks,
srini
Srinivas Kandagatla (8):
ASoC: q6dsp: q6afe: add support to Codec DMA ports
ASoC: q6dsp: q6routing: add support to Codec DMA ports
ASoC: q6dsp: q6afe: prepare afe_apr_send_pkt to take response opcode
ASoC: q6dsp: q6afe: add global q6afe waitqueue
ASoC: q6dsp: q6afe: add lpass hw voting support
ASoC: q6dsp: q6afe: update q6afe_set_param to support global clocks
ASoC: q6dsp: q6afe: add codec lpass clocks
ASoC: q6dsp: q6afe-dai: add support to Codec DMA ports
include/dt-bindings/sound/qcom,q6afe.h | 22 ++
sound/soc/qcom/qdsp6/q6afe-dai.c | 229 ++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.c | 308 +++++++++++++++++++++++--
sound/soc/qcom/qdsp6/q6afe.h | 33 ++-
sound/soc/qcom/qdsp6/q6routing.c | 121 +++++++++-
5 files changed, 689 insertions(+), 24 deletions(-)
--
2.21.0
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In ASoC TXx9 ACLC driver, a tasklet
is still used for offloading the hardware reset function. It can be
achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in TXx9 ACLC driver with a
simple work. The conversion is fairly straightforward.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200903104749.21435-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In ASoC SH SIU driver, a tasklet is
still used for offloading the hardware reset function. It can be
achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in SH SIU driver with a simple
work. The conversion is fairly straightforward.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200903104749.21435-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In ASoC FSL ESAI CPU DAI driver, a
tasklet is still used for offloading the hardware reset function.
It can be achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in fsl esai driver with a simple
work. The conversion is fairly straightforward.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200903104749.21435-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
In some cases we need to probe additional audio components that do
not appear as part of the DAI links specified in the device tree.
Examples for this are auxiliary devices such as analog amplifiers
or codecs.
The ASoC core provides a way to probe these components by adding
them to snd_soc_card->aux_dev.
This patch set allows specifying them in the device tree through
a new "aux-devs" property.
v1: https://lore.kernel.org/linux-arm-msm/20200819091533.2334-1-stephan@gerhold.net/
Changes in v2:
- Fix value type in device tree bindings:
aux-devs should be array of phandles without any arguments, so change
<phandles with arguments> -> <array of phandles>
Stephan Gerhold (2):
ASoC: dt-bindings: qcom: Document "aux-devs" property
ASoC: qcom: common: Parse auxiliary devices from device tree
.../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 7 +++++++
Documentation/devicetree/bindings/sound/qcom,apq8096.txt | 8 ++++++++
Documentation/devicetree/bindings/sound/qcom,sdm845.txt | 8 ++++++++
sound/soc/qcom/common.c | 4 ++++
4 files changed, 27 insertions(+)
--
2.28.0
In some cases we need to probe additional audio components that do
not appear as part of the DAI links specified in the device tree.
Examples for this are auxiliary devices such as analog amplifiers
or codecs.
The ASoC core provides a way to probe these components by adding
them to snd_soc_card->aux_dev. We can use the snd_soc_of_parse_aux_devs()
function to parse them from the device tree.
As an example for this, some MSM8916 smartphones have an analog
speaker amplifier connected to the HPHR output. With the new property
this can be modelled as follows:
speaker-amp: audio-amplifier {
compatible = "simple-audio-amplifier";
enable-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>;
sound-name-prefix = "Speaker Amp";
};
&sound {
aux-devs = <&speaker_amp>;
audio-routing = "Speaker Amp IN", "HPHR";
};
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200826095141.94017-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
In some cases we need to probe additional audio components that do
not appear as part of the DAI links specified in the device tree.
Examples for this are auxiliary devices such as analog amplifiers
or codecs.
To make them work they need to be added as part of "aux-devs"
and connected to some other audio component using the audio routes
configurable using "(qcom,)audio-routing".
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200826095141.94017-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
