Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec.
Jack detect code from Kevin Strasser <kevin.strasser@intel.com>, GPIO
resolving from Mika Westerberg <mika.westerberg@linux.intel.com> and fixes
and cleanups from Liam Girdwood <liam.r.girdwood@linux.intel.com>.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some unknown reason the parameters for snd_soc_test_bits() were in wrong
order:
It was:
snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */
while it should be:
snd_soc_test_bits(codec, reg, mask, val);
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
WM8804 can run with PLL frequencies of 256xfs and 128xfs for
most sample rates. At 192kHz only 128xfs is supported. The
existing driver selects 128xfs automatically for some lower
samples rates. By using an additional mclk_div divider, it
is now possible to control the behaviour. This allows using
256xfs PLL frequency on all sample rates up to 96kHz. It
should allow lower jitter and better signal quality. The
behavior has to be controlled by the sound card driver,
because some sample frequency share the same setting. e.g.
192kHz and 96kHz use 24.576MHz master clock. The only
difference is the MCLK divider.
Signed-off-by: Daniel Matuschek <daniel@matuschek.net>
Tested-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X81 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X61 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1381 and ADAU1781 audio
CODECs. The device is a low-power, 24-bit stereo audio CODEC with multiple
analog inputs and outputs, two digital microphone inputs and an I2S interface.
The device can be controlled either using I2C or SPI. The main difference
between the two variants is that the ADAU1781 has a freely programmable SigmaDSP
processor, while the ADAU1381 has a fixed function wind noise reduction filter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1361 and ADAU1761 CODECs.
The device is a a low-power, 24-bit stereo audio CODEC with multiple analog
input and outputs, one digital microphone input and an I2S interface. The device
can be controlled either via I2C or SPI. The main difference between the two
variants is that the ADAU1761 has a built-in SigmaDSP, while the ADAU1361 has
not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
We have defines for adsp messages best to consistently use them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit e5d80e82e3 (ASoC: sgtl5000: Convert to use regmap directly) a
kernel oops is observed after a suspend/resume sequence.
The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no
longer a valid pointer.
Add the remaining register entries into sgtl5000_reg_defaults[] and remove
sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and
make the code simpler.
Tested on a im53-qsb board.
Reported-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
commit 8c5178fca4 ("ALSA: Add params_width() helpers") introduces
a helper to get the sample width. Updating Samsung related sound
drivers to use this helper.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Having the binary ones complement operator in the new bitmak value makes the
code hard to read.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add "depends on I2C" to shut up the build errors from randconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make including the omap-pcm.h outside sound/soc/omap more convenient.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow jack GPIO pins be defined also using GPIO descriptor-based interface
in addition to legacy GPIO numbers. This is done by adding two new fields to
struct snd_soc_jack_gpio: idx and gpiod_dev.
Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is
NULL and otherwise idx is the descriptor index within the GPIO consumer
device.
New function snd_soc_jack_add_gpiods() is added for typical cases where all
GPIO descriptor jack pins belong to same GPIO consumer device. For other
cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before
calling snd_soc_jack_add_gpios().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs
are still passed and requested using legacy GPIO numbers the driver
internals are converted to use GPIO descriptor API.
Motivation for this is to prepare soc-jack so that it will allow registering
jack GPIO pins using both GPIO descriptors and legacy GPIO numbers.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Failure to terminate this match table can lead to boot failures
depending on where the compiler places the match table.
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DMAC src/dst addr needs to be set from driver when DT case.
(It was set from SoC/DMAEngine code when non-DT case)
This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting to use DMAEngine.
But, DMA slave channel name "tx", "rx" is not enough
in DT case.
Becuase, it has many ports and path combination.
This patch adds rsnd_dma_of_name() to find
DMA channel name, for example
memory to SSI0 is "mem_ssi0",
SSI0 to memory is "ssi0_mem",
SSI0 to SRC0 is "ssi0_src0",
SRC0 to SSI0 is "src0_ssi0",
SRC0 to DVC0 is "src0_dvc0"...
Renesas sound want to use PIO transfer mode for some reasons.
It will be PIO tranfer mode if device node doesn't have
DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver uses many modules (= SSI/SRC/DVC),
and each module had own name.
But, each module name can be used as several purpose,
like clock name, DMA name etc...
This patch uses common name for each module.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting Gen1/Gen2.
SRC probe can return error if it was unknown
generation.
Now, rsnd_src_non_ops is not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DT DMA support needs struct platform_device pointer,
and it can get struct device pointer from platform_device.
Save platform_device instead of device.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Driver needs to call of_node_put() after of_get_chile_by_name()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, gGuard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
This change fixes all Tegra machine drivers. By code inspection, I
believe some non-Tegra machine drivers have the same issue. I'll send a
patch for that separately, once this is reviewed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes sure a format string can never get processed into the worker
thread name from the device name.
Signed-off-by: Kees Cook <keescook@chromium.org>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some platforms require that the codecs mclk is a fixed multiplication
factor of the audio stream rate. Add a optional property to the
binding to hold this factor and implement a hw_params() function to
make use of it.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the clock divisor and multiplier NI, MI values for audio
sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful
for the Odroid X2/U2 boards when the codec works in master mode and its
MCLK clock is fed from the I2S CDCLK output.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
[s.nawrocki@samsung.com: edited the commit description]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' into asoc-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
clk_get in probe function can be safely replaced with devm_clk_get.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAPM routes for this board are complete, hence we can let the core take care
of disconnecting non-connected pins rather than doing it manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for setting source clock of the "Core CLK"
of the IIS Multi Audio Interface.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we are moving the mmp platform towards multiplatform support,
we have to stop including platform header files.
This changes the pxa-ssp sound driver file to no longer depend
on mach/hardware.h and mach/dma.h. The code using the definitions
from those headers is actually gone already, the only thing
that was still being used was the pxa_dma_desc typedef, which
we can easily work around by using the normal 'struct pxa_dma_desc'
name.
The pxa2xx-dma driver still uses this header, so we include it
explicitly there, which is ok because that is only used on pxa,
not on mmp.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.
Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsl_ssi_set_dai_sysclk will be called from fsl_ssi_hw_params in the
next patch. Move up to avoid forward declaration and to keep the next patch
more readable. No functional change.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the fsl-ssi unit is used in i2s slave mode, it is possible that the
SSI unit starts transmitting data on the wrong channel. This happens
because the SSI does not synchronize with the left-right-clock by
default.
This patch enables transmit enable synchronization.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are some variables defined in struct fsl_ssi_private that describe
states that are also described by other variables.
This patch adds some helper functions that return exactly the same
information based on available variables. This helps to clean up struct
fsl_ssi_private and remove them from the probe function.
It also removes some not really used variables (new_binding, name).
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder the probe function to be able to move the second imx-specific
block to the seperate imx probe function. The patch also removes some
comments/variables/code that are not used anymore or could be simply
replaced by other variables.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify dma DT property handling. fsl,ssi-dma-events is not used
anymore. It passes invalid data to imx_pcm_dma_params_init_data() which
copies some data into an imx dma struct. This struct is never used in
imx-dma or imx-sdma because of generic OF DMA handling. The
"fsl,ssi-dma-events" is not used anywhere in dts files.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move imx specific probe code to a seperate function. It reduces the
size of the probe() function and makes the code and error handling
easier to understand.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of creating a name using string manipulation functions, we can
simply use the device name for the DAI driver struct.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move all code that is only used for debugging to a seperate file. This
makes it easier to see what functions are used for debugging only.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The bits we have to clear when disabling are different when the other
stream is still active.
This patch fixes the calculation of new register values after disabling
one stream. It also adds a more detailed description of the new register
value calculation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.
Also fixes the following warnings from smatch:
sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.
While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In some cases the pcm stream is closed while context has been
scheduled to be restored, causing a null pointer deref panic.
Cancel work to ensure stream does not get freed while work is
still active/pending.
Also, restoring the pcm context can be safely skipped after the
stream has been stopped. Check if pcm stream is still running
before restoring stream context to help pending work finish
more quickly in stream close path.
Signed-off-by: Kevin Strasser <kevin.strasser@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This looks like a copy-paste bug, fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Microphone bias level configuration register can configure voltage between
2.2 V and 2.8 V but doesn't manage is voltage on or off. Microphone bias
on/off state is controlled by "MICBIAS" DAPM widget.
Therefore there is no need to update bias voltage conditionally depending on
jack state each time when codec goes to SND_SOC_BIAS_ON state and setting
can be moved to max98090_probe() as driver currently doesn't support other
levels than 2.8 V.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
BCL is being configured in two places producing a warning message.
Make sure we only configure BCLK once and when we are master.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add ACPI ID for MAX98090 and ACPI 5 I2C device probing support.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure the cache is fully flushed at resume time.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we already have a memeber struct snd_sst_params.codec to fill this.
so removing duplicate instance
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the support to implement drain_notify in Intels mfld driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 0cac6fc3eb.
This comiit was dropped from rev2 and would not be required as it renames the
platform ops as well which is not required.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I managed to drop a change to stream ID setting from commit 49fee17816
("ASoC: Intel: Only export one Baytrail DAI") leading to non-working
simultaneous capture-playback since after one DAI conversion
rtd->cpu_dai->id + 1 will be the same for both playback and capture.
Use substream->stream + 1 like it was in original Liam's patch.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the %pap printk specifier to print resource_size_t variables. This
fixes warnings on platforms where resource_size_t has a different size
than int.
Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas@ideasonboard.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We don't need more than one DAI for Baytrail SST. Usage becomes also more
straightforward by grouping playback and capture streams under the same PCM
device.
[Jarkko: I made Liam's sst-baytrail-pcm.c change a few lines smaller and
squashed together with my byt-rt5640.c change]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Prepare for single Baytrail DAI playback/capture link by accessing PCM data
using stream ID instead of rtd->dev. Now rtd->dev is unique for playback
and capture since they are exported as separate DAIs but not once converted
to single DAI.
[Jarkko: Separated from another commit with updated commit log]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The return statement was not indented correctly. I lined up the
condition a bit as well.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
Adding missing pm ops so that audio playback works across
suspend and resume cycle.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
