| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
Based on 1 normalized pattern(s):
licensed under the gpl 2 or later
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 82 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Reviewed-by: Richard Fontana <rfontana@redhat.com>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190524100845.150836982@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so
making them static instead of exported.
Signed-off-by: Robert Rosengren <robertr@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Safeload support has been implemented which is used
when updating for instance filter parameters using
alsa controls. Without safeload support audio can
become distorted during update.
Signed-off-by: Danny Smith <dannys@axis.com>
Signed-off-by: Robert Rosengren <robertr@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The playback DAI is connected to the DSP and the DSP might be sourcing
signals from the playback stream. Add a DAPM route between the two to make
sure that the playback DAI is powered up, when the DSP is active.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Reloading fw causes an audiable popping sound, we can avoid this
by not reloading if the samplerate is the same as before.
Signed-off-by: Danny Smith <dannys@axis.com>
Signed-off-by: Robert Rosengren <robert.rosengren@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
DSP_RUN needs to be disabled during firmware write otherwise
we can end up with undefined behavior if writing to a dsp which
is already running firmware.
Signed-off-by: Danny Smith <dannys@axis.com>
Signed-off-by: Robert Rosengren <robert.rosengren@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
{ 0 } only clears the first member of the structure.
The first member of the snd_soc_dapm_update struct is a pointer,
and writing 0 to a pointer results in a sparse warning.
Use the empty struct initializer that clears all the struct members
and fixes the sparse warning.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Now we can replace Codec to Component. Let's do it.
Because there are many drivers which are using adau17x1,
we need to update these all related drivers in same time.
Otherwise compile error/warning happen
adau1761:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
adau1781:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The ADC in the ADAU1361 (and possibly other Analog Devices codecs)
exhibits a cyclic variation in the noise floor (in our test setup between
-87 and -93 dB), a new value being attained within this range whenever a
new capture stream is started. The cycle repeats after about 10 or 11
restarts.
The workaround recommended by the manufacturer is to toggle the ADOSR bit
in the Converter Control 0 register each time a new capture stream is
started.
I have verified that the patch fixes this problem on the ADAU1361, and
according to the manufacturer toggling the bit in question in this manner
will at least have no detrimental effect on other chips served by this
driver.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
|
|
In adau17x1_pll_event(), 'ret' is initialized as return value of
regmap_raw_write() but never checked, so remove this and assignement.
sound/soc/codecs/adau17x1.c: In function ‘adau17x1_pll_event’:
sound/soc/codecs/adau17x1.c:68:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret;
Cc: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
To support double channel shared controls split across 2 registers, one
for each channel, we must be able to update both registers together.
Add a second set of register fields to struct snd_soc_dapm_update, and
update the DAPM control writeback (put) callbacks to support this.
For codecs that use custom events which call into DAPM to do updates,
also clear struct snd_soc_dapm_update before using it, so the second
set of fields remains clean.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The devices from the ADAU17X1 family all have a MCLK clock input which
supplies the master clock for the device. The master clock is used as the
input clock for the PLL. Currently the MCLK rate as well as the desired PLL
output frequency need to be supplied by calling snd_soc_dai_set_pll() form
a machine driver.
Add support for specifying the MCLK using the common clock framework. In
addition to that also automatically configure the PLL to a suitable rate
if the master clock was provided using the CCW. This allows to use the
CODEC driver without any special configuration requirements from the
machine driver.
While the PLL output frequency can be configured over a (more or less)
continuous range the narrowness of the range and the other constraints of
the clocking tree usually only result in two output frequencies that will
actually be chosen. One for 44.1kHz based rates and one for 48kHz based
rates, these are the rates that the automatic PLL configuration will use.
For the rare case where a non-standard setup is required a machine driver
can disable the auto-configuration and configure a custom frequency using
the existing mechanisms.
If the common clock framework is not enabled clk_get() will return NULL and
the driver will function as before and the clock rate needs to be
configured manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Multiple devices from the ADAU family share the same PLL structure and
configuration register layout. Introduce a new helper module that can be
used to calculated the PLL configuration registers based on a specified
input frequency and the desired output frequency of the PLL.
The ADAU1761/ADAU1781 and ADAU1373 drivers are updated to make use of this
new helper module. But future drivers for additional devices from the ADAU
family are also expected to make use of it.
In anticipation of sharing more infrastructure code between different
devices from the ADAU family the new module is called adau-utils.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
'asoc/topic/simple', 'asoc/topic/sirf' and 'asoc/topic/sn95031' into asoc-next
|
|
To be able to read back data from the DSP parameter memory the register
range needs to be marked as readable. At the same time we do not want them
to e.g. appear in debugfs output so mark them as precious as well.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The v2 file format of the SigmaDSP takes a more declarative style compared
to the imperative style of the v1 format. In addition some features that are
supported with v2 require the driver to keep state around for the firmware.
This requires a bit of restructuring of both the firmware loader itself and
the drivers making use of the firmware loader.
Instead of loading and executing the firmware in place when the DSP is
configured the firmware is now loaded at driver probe time. This is required
since the new firmware format will in addition to the firmware data itself
contain meta information describing the firmware and its requirements and
capabilities. Those will for example be used to restrict the supported
samplerates advertised by the driver to userspace to the list of samplerates
supported for the firmware.
This only does the restructuring required by the v2 format, but does not
yet add support for the new format itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The codec field of the snd_soc_widget struct is eventually going to be
removed, use snd_soc_dapm_to_codec(w->dapm) instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The CODEC doesn't care how data is laid out in memory.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
|
|
The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite
different in the analog domain. This patch adds support for the common parts of
the ADAU1X61 and ADAU1X81 CODECs.
The patch also restores some of the alphabetical order in the Makfile and
Kconfig.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
