The PWR_ANLG1 reg not only contains various power on/off bits, it also
contains 2 bits which select if the LDO generates 1.0, 1.1 or 1.2V. Note
there is a separate on/off bit for the LDO.
rt5651_set_bias_level(BIAS_OFF) used to unconditionally clear the entire
register, when jack-detection support was introduced a special case for
jack-detect was added which hard-codes a register value to keep the LDO
voltage at 1.2 volt.
This commit removes the jack-detect special case, instead simply always
leaving the LDO voltage control bits as is on BIAS_OFF.
Tested-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt5651_set_bias_level() function was turning everything off at
SND_SOC_BIAS_STANDBY, rather then at SND_SOC_BIAS_OFF, requiring the bias-
level to be raised to SND_SOC_BIAS_PREPARE before turning anything on.
This is not how the bias-levels are supposed to work, this commit fixes
this by turning everything off at the SND_SOC_BIAS_OFF level and enabling
the pwr-bits needed for minimum functionality at SND_SOC_BIAS_STANDBY.
This fixes the minimum set of pwr-bits not getting enabled when
force-enabling some dapm-supplies (e.g. for jack type detection),
which raises the bias-level to standby.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
is_sys_clk_from_pll() is used as a snd_soc_dapm_route.connected callback,
checking RT5651_GBL_CLK to determine if the sys-clk is PLL1 and thus the
PWR_PLL bit in reg PWR_ANLG2 must be set.
RT5651_GBL_CLK is changed by rt5651_set_dai_sysclk(), which gets called by
the pre_pmu / post_pmd functions of the "Platform Clock" dapm-supply.
This creates an ordering issue, during a dapm transition first all
connected() callbacks are called to build a list of supplies to enable
and then the complete list is walked to enable the supplies. Since the
connected() check happens before enabling any supplies,
is_sys_clk_from_pll() ends up deciding if the PWR_PLL bit should be set
based on the state the "Platform Clock" supply had *before* the transition.
This sometimes results in PWR_PLL being off, even though *after* the
transition PLL1 is configured as sys-clk.
This commit removes is_sys_clk_from_pll() instead simply setting / clearing
PWR_PLL in rt5651_set_dai_sysclk() based on the selected sys-clk, which
fixes this and as a bonus results in a nice cleanup.
Tested-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Codes is stereo only with playback and capture streams bind
to the same rate.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tell user what are clock rate limits and reindent log messages.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the applying of the differential input and dmic properties to a new
rt5651_apply_properties() helper function. This new function can be called
by platform code which attaches properties after probe() has run to apply
these new properties.
Note this also moves the time when we apply these properties for DT
platforms from i2c-probe to snd-component-probe time, this should not
result in any functional difference.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the standard component set_jack callback instead of defining a codec
private API for this.
Tested-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move 2 functions higher up in rt5651.c, this is a preparation patch
to avoid needing forward declarations when moving over from a codec
private function to the standard snd_soc_component_set_jack().
This commit purely moves these 2 functions up, not a single line is
changed.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move all jack-detect initialization to rt5651_set_jack_detect. The main
reason to do this is so that platform code can setup jack-detect properties
after the device has been probed, which unfortunately is necessary on some
platforms.
This has 2 additional advantages:
1) Grouping all jack-detect init together makes it easier to follow what
is happening and results in a small reduction in the number of loc.
2) Before we would register the irq handler before rt5651->hp_jack was
assigned, leading to a potential NULL deref if the jack_detect work runs
before the machine driver has called set_jack.
Tested-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are no in tree users of platform-data for the rt5651 codec driver,
so lets remove support for it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Normal 512-byte get/set of a TLV isn't supported but we were
registering the normal get/set anyway and relying on omitting
the SNDRV_CTL_ELEM_ACCESS_[READ|WRITE] flags to prevent them
being called.
Trouble is if this gets broken in the core ALSA code - as it has
been since at least 4.14 - the standard get/set can be called
unexpectedly and corrupt memory.
There's no point providing functions that won't be called and
it's a trivial change. The benefit is that if the ALSA core gets
broken again we get a big fat immediate NULL dereference instead
of a memory corruption timebomb.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.
The driver has been written from scratch using Motorola's Android
driver, register dumps from running Android and datasheet for NXP
MC13783UG (which is similar to Motorola CPCAP, but not the same).
The chip provides two audio interfaces, that can be muxed to two
different audio codecs. One provides support for stereo output
(named StDAC or HiFi), while the other only provides mono output
(named Voice). Only the Voice codec provides a Capture interface.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support of the ROHM BD28623MUV
Class D speaker amplifier for Flat-panel TVs.
This IC delivers an output power of 20W + 20W.
Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro@socionext.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clear the buffer data structure on each trigger start such that the
buffer is in a sensible state even if the DSP itself didn't restart.
This is necessary to support voice control streams which can trigger
multiple times without reloading the firmware.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Repair any formatting/style issues that can be fixed without major
code refactoring.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use // for .c and /* for .h files according to:
Documentation/process/license-rules.rst file.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hcp->chmap_info must not be freed up in the hdmi_codec_remove()
function as it leads to kernel crash due ALSA core's
pcm_chmap_ctl_private_free() is trying to free it up again when the card
destroyed via snd_card_free.
Since the driver was converted to devm_snd_soc_register_component() the
whole platform_device.remove can be removed.
Commit cd6111b262 ("ASoC: hdmi-codec: add channel mapping control")
should not have added the kfree(hcp->chmap_info); to the hdmi_codec_remove
function.
[Separate verison of the fix for -next -- broonie]
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hcp->chmap_info must not be freed up in the hdmi_codec_remove()
function as it leads to kernel crash due ALSA core's
pcm_chmap_ctl_private_free() is trying to free it up again when the card
destroyed via snd_card_free.
Commit cd6111b262 ("ASoC: hdmi-codec: add channel mapping control")
should not have added the kfree(hcp->chmap_info); to the hdmi_codec_remove
function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Tested-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The max9759 is a gpio controlled amplifier.
Tested on a Variscite Dart MX6 SoM based custom board.
Signed-off-by: Neil Armstrong <narmstrong@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify SPDX comment style to match the rules. Since the original
code had a GPL-2.0 license and SPDX is compatible with GPL-2.0+,
we also modified the MODULE_LICENSE.
Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On some systems a delay is needed after switching on the clocks, to allow
the output to stabilize and avoid a popping noise at the beginning of
the recording. Add the optional device tree property 'wakeup-delay-ms'
and apply the specified delay after enabling the mic. A blocking delay
can't be applied in dmic_daiops_trigger() since the function is called
in atomic context. Instead use a DAPM event handler to set the enable
GPIO and apply the delay in the handler.
Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ALC5651 does not like multi-write accesses, avoid them. This fixes:
rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121
Errors on resume (and all registers after the registers in the error not
being synced).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Enable 5-band graphic equalizer which allows manipulation of
gain in range -11.75dB to 12dB for 5 specific sound bands:
115Hz, 330Hz, 990Hz, 3000Hz, 9900Hz.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add mux for choosing DAC source: ADC, I2S or DAP. DAP - digital
audio processing block. Moreover create choosing mux for DAP main
source (ADC or I2S) and DAP mixer source (ADC or I2S). Mixer block
allows summing DAP main and mixer sources togheter with defined
volumes.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With P0 R29, Bit2 set the I2S clocks will be running when the DAC/ADC is
powered down, but still the codec need to be powered up by needing at least
one complete DAPM path for the stream.
If the AIF is not needed (analog loopback for example) the I2S clocks will
not run as they are not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Suggested-by: Stefan Müller-Klieser <s.mueller-klieser@phytec.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio interface is not really the DAC input. Use more generic name for
it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
base and sysclk_reg should be unsigned int, as that is what
regmap takes as a register address.
sysclk_mask is used to mask unsigned register values so
should be unsigned.
sysclk_shift is a shift value so is not allowed to be negative.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 8419caa727 ("ASoC: sgtl5000: Do not disable regulators in
SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume
sequence:
Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Stereo
aplay: pcm_write:2051: write error: Input/output error
The problem is caused by the fact that the aforementioned commit
dropped the cache handling, so re-introduce the register map
resync to fix the problem.
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
One of the recently introduced commits hadn't been build tested, fix
that.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the const array norm_addrs on the stack, instead make it
static. Makes the object code smaller by over 230 bytes. Also re-format
array data as the insertion of the static keywork made the first line
overly long.
Before:
text data bss dec hex filename
53780 34752 256 88788 15ad4 linux/sound/soc/codecs/tscs42xx.o
After:
text data bss dec hex filename
53461 34840 256 88557 159ed linux/sound/soc/codecs/tscs42xx.o
(gcc version 7.2.0 x86_64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In usecases like hot plug-unplug DP panel or modeset during a playback,
sometimes we observe no audio after codec resets. During no audio
condition, we have noticed that the power state of the pin or the
connector is D3. Optimizing the way we set the power mitigates the
issue. With this changes the verb is sent to set the power state and
waits until actual state reaches target state. Thus ensuring power
state is set.
Signed-off-by: Abhijeet Kumar <abhijeet.kumar@intel.com>
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.
Cc: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
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>
Current implementation mute codec in global way (DAC block).
That means when user routes sound not from I2S but from
AUX source (LINE_IN) it also will be muted by alsa core.
This should not happen.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the reset state of the codec we do not have complete playback or capture
routes.
The audio playback/capture will not work due to missing clock signals on
the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down.
To make sure that even if all output/input is disconnected the codec is
generating clocks, we need to have valid DAPM route in every case to power
up the must needed parts of the codec.
I have verified that switching DAC (during playback) or ADC (during
capture) will stop the I2S clocks, so the only solution is to connect the
'Off' routes as well to output/input.
The routes will be only added if the codec is clock master. In case the
role changes runtime, the applied routes are removed.
Tested on am43x-epos-evm with aic3111 codec in master mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
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 the following sparse warning:
sound/soc/codecs/tlv320aic3x.c:175:47: warning: Using plain integer as NULL pointer
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>
{ 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 the following sparse warning:
sound/soc/codecs/wm9712.c:229:47: warning: Using plain integer as NULL pointer
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>
snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time"
decision. Then, it collects each component driver's .use_pmdown_time
settings by using AND operation.
But, ak4613 has no reason to ignore/use pmdown time.
Thus, It should has use_pmdown_time=0, it means it doesn't have
decision right of "ignore pmdown time".
Otherwise, ak4642 paired driver's pmdown_time settings doesn't
have meaning. In other words, "ignore pmdown time" is determined
irrespective of the value of paired driver's settings
(= ignore will be 0 by ak4642).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
