devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
devm_regulator_bulk_get() is device managed and makes error
handling and code cleanup simpler.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some ADSP devices can make use of DVFS to optimise power consumption
depending on the operating frequency of the DSP core. Implement
support for this in the generic ADSP code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a haptics widget for use by the haptics driver and expose the DAPM
context for it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a haptics widget for use by the haptics driver and expose the DAPM
context for it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We cannot include any plat or mach headers for the multiplatform
support.
Fix the issue by defining local mcbsp_omap1().
Signed-off-by: Tony Lindgren <tony@atomide.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Hi Mark,
thanks your insisting on a better description for the patch, I found a
more appropriate solution for the problem:
Compiling the SoC Audio driver for Freescale i.MX as a module
(CONFIG_SND_SOC_IMX_PCM=m) results in a non-functional sound driver
indicated by the error message:
| imx-sgtl5000 sound.1: platform imx-pcm-audio not registered
| imx-sgtl5000 sound.1: snd_soc_register_card failed (-517)
| platform sound.1: Driver imx-sgtl5000 requests probe deferral
instead of the message:
| imx-sgtl5000 sound.1: sgtl5000 <-> 63fcc000.ssi mapping ok
that is to be expected upon loading the snd-soc-imx-pcm.ko module.
The build log reveals, that the file imx-pcm-dma.o (or imx-pcm-fiq.o
depending on the kernel configuration), which should be linked
together with imx-pcm.o into snd-imx-pcm.ko, is not being compiled in
this case.
The make rules for these files shows that the target object imx-pcm.o
is assigned to the variable snd-soc-imx-pcm-y while
imx-pcm-{dma,fiq}.o are added to to
snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) and
snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) which resolve to
snd-soc-imx-pcm-m in this case.
According to Documentation/kbuild/modules.txt:
|When the module is built from multiple sources, an additional line is
|needed listing the files:
|
| <module_name>-y := <src1>.o <src2>.o ...
Thus the type of the config variables CONFIG_SND_SOC_IMX_PCM_DMA and
CONFIG_SND_SOC_IMX_PCM_FIQ should be 'bool' instead of 'tristate' to
resolve to 'y' when selected.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to a broken make rule, sound/soc/fsl/imx-pcm-dma.c or
sound/soc/fsl/imx-pcm-fiq.c (whatever is selected via Kconfig) will
not be compiled into imx-pcm.o when building as module, i.e.:
CONFIG_SND_SOC_IMX_PCM=m
CONFIG_SND_SOC_IMX_PCM_DMA=m
resulting in a non-functional sound driver.
This gives the error messages:
| imx-sgtl5000 sound.1: platform imx-pcm-audio not registered
| imx-sgtl5000 sound.1: snd_soc_register_card failed (-517)
| platform sound.1: Driver imx-sgtl5000 requests probe deferral
when loading the driver instead of what's to be expected:
| imx-sgtl5000 sound.1: sgtl5000 <-> 63fcc000.ssi mapping ok
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The device should not be generating interrupts when it does not have power
so ignore incoming interrupts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't rely on the kcontrol for robustness reasons, the widget mechanism
is what the framework uses.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is better style and facilitates implementation of device tree support
for the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This device doesn't have a pdata definition for legacy boards, and
unless anyone need to control the reset GPIO, it's not worth adding one.
So this feature is only available to DT users for now.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
set the 'onwer' field of the registered snd_soc_card object to prevent
removal of the module when its resources are in use.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
also set MODULE_AUTHOR and MODULE_DESCRIPTION
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes following warning.
sound/soc/codecs/wm8753.c:1594:1-6: WARNING: invalid free of devm_ allocated data
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes following warning.
sound/soc/codecs/wm8510.c:614:1-6: WARNING: invalid free of devm_ allocated data
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM is a pseudo-device. It doesn't have any of it's own registers
or hardware. It rather acts as a layer of abstraction for DMA
transfers. Hence, instead of classifying it as a device in its own
right, we call the initialisation from the MSP driver.
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Ola LILJA2 <ola.o.lilja@stericsson.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The info record at the start of the dsp firmware file has been
expanded to incorporate additional version information. We need
to check the version to make sure we understand the layout of
the information in the record. The srec2image tool is currently
used to create this record during creation of the .dfw file.
Signed-off-by: Scott Ling <sl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When lowering SYSCLK to 50kHz for accessory detection also lower the
AIFnCLK divisor to normalise the clocking configuration within the
device. This will not disrupt audio as we cannot support active audio
with such a low SYSCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Increase timeout to be more reliable and avoid the chance of
missing interrupts during boot.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the max98090 codec prototype driver.
It supports Headphone only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removed struct ak4642_priv which had
meaningless variable.
It is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC has a mixture of message prefixes e.g. "ASoC", "asoc"
or none and message types e.g. pr_debug or dev_dbg.
Make sure all ASoC core messages use the same "ASoC" prefix and
convert any component device specific messages to use dev_dbg
instead of pr_debug.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
With SPDIF passthrough, we are not restricted to just two channels of
audio; we can support however many channels the non-audio stream can
itself support. In any case, kirkwood-dma is not involved in the
format selection. So yet rid of this restriction.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox, and cleaned up by me.
Some platforms provide an external clock which can be used to allow
other sample rates to be selected. Provide support for this.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
The kirkwood DMA hardware for ASoC does not impose any restrictions
on the sample rates available, so it's silly to impose an artificial
set in the DMA code. The restrictions come from the availble clocks
to the I2S module, which are already handled in the I2S part of the
driver.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simplify the cleanup paths in the driver by using the devm_* APIs,
ensuring that all error paths are correctly checked.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't even momentarily set the pause status when starting the channel;
if we do, we should check the busy bit to ensure that we comply with
the spec. In any case, it isn't necessary; we will not active on a
START event so there is no need to pause the DMA.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stress testing the driver with multiple start/stop events causes
kirkwood-dma to report underrun errors (which used to cause the kernel
to lock up solidly). This is because kirkwood-i2s is not respecting
the restrictions imposed on clearing the 'pause' bit. Follow what the
spec says; the busy bit must be read as being clear twice before the
pause bit can be released. This solves the underruns.
However, it has been noticed that the busy bit occasionally does not
clear itself, hence the waiting is bounded to 5ms maximum to avoid a
new reason for the kernel to lockup.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox, which is further attributed to Sebastian Hesselbrath.
Rather than masking the KIRKWOOD_DCO_SPCR_STATUS register contents
against the registers virtual address, let's actually use the bit
definition for the locked status, as required in the documentation.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ignoring the real cause of the interrupt is not a good idea; this
behaviour has been observed to bring Dove platforms to silently
lockup. Instead, on error fall through to the normal interrupt
processing.
This is especially important on Dove platforms as errors are
handled separately, and allows us to clear down the real cause of
the interrupt.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is part of a patch found in Rabeeh Khoury's git tree for the
cubox.
You can not use virt_to_phys() on the address returned from
dma_alloc_coherent(); it may not be part of the kernel direct-mapped
memory. Fix this to use the DMA address instead.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seems WM_ADSP2("DSP1", 0) is added twice to the widgets list, remove
that and in place use ARIZONA_DSP_WIDGETS(DSP1, "DSP1").
We need to make sure that the DSP1 Aux widgets are provided otherwise
we'll see errors such as "Failed to add route DSP1 Aux 1 -> DSP1" etc.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CONFIG_HOTPLUG is going away as an option so __devexit_p is no longer
needed, remove it.
Also fix the indentation for the initialization of the
max98088_i2c_driver struct to make chkpatch happy.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch adds SND_SOC_DAIFMT_INV_xxx support,
and it is possible to independent from platform information pointer.
Old type SH_FSI_xxx_INV is still supported,
but it will be removed soon.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes stream mode format
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes spdif format
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is requesting sh_fsi_platform_info pointer from platform,
and it didn't allowed NULL pointer.
This patch fixes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch tidyup to use fsi pointer for FSIA/B settings
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CONFIG_HOTPLUG is going away as an option so __devinitconst is no
longer needed.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return the value obtained from get_coeff() instead of EINVAL.
Silences a smatch warning.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
convert at91sam9g20ek with wm8731 to device tree support
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove unneeded code with the new method of dai and pcm register
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
change the method for register dai and pcm
- let the atmel-ssc-dai no longer as a standalone platform device
- remap ssc and then register dai directly
- register pcm from dai directly
- modify the code which related with this change
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove remuxing GPIO1. Leave control of this up to the platform device.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize performance by providing a 512fs based CLKIN.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
It seems git has been getting confused by the very similar contexts
for the speaker DAIs and has been applying patches to the wrong places
causing all sorts of confusion. Fix this up by hand.
Reported-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no mixer attached to the ASRC on the wm5110 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5110 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no mixer attached to the ASRC on the wm5102 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5102 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Asynchronous Sample Rate Converters on the wm5102/wm5110 have no
mixer attached to their input, but they do allow the input to be
selected from a number of sources via a multiplexer. Currently the
platform assumes the presence of 4 multiplexers and a mixer for each
block.
This patch adds support multiplexed single input blocks into the Arizona
platform.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case of probe deferral, the allocated GPIO line is not freed, which
prevents it from being claimed and properly asserted in later attempts.
Fix this by using devm_gpio_request().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Michael Hirsch <hirsch@teufel.de>
Cc: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_put_volsw_sx function fails to update second control
if first control is updated by snd_soc_update_bits_locked.
Signed-off-by: Mukund Navada <navada@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Move the firmware load and record parsing functionality out into
a separate function from the boot function.
Signed-off-by: Scott Ling <sl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the return value of cs42l52_set_fmt() when clock inversion is
not allowed and also remove the useless variable ret.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
[We had been assigning to ret but then ignoring the value we assgined
-- broonie]
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Remove the boot_done counter variable and check the wm0010 state
variable instead.
Signed-off-by: Scott Ling <scott.ling@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Playing 24-bit format file leads to channel swap on mx28 and the reason is that
the current driver performs one write/read to/from the SAIF_DATA register to
trigger the transfer.
This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register
and thus is capable of storing the 16-bit left and right channels, but for the
S24_LE case it can only store one channel, so in order to not lose the FIFO sync
an extra read/write is needed.
Reported-by: Dan Winner <DWinner@tc-helicon.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Dan Winner <DWinner@tc-helicon.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we select the WM1250-EV1 as the current software system
configuration demands it.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver didn't care fsi_hw_start/stop() return value,
and it causes WARNING() call if SNDRV_PCM_TRIGGER_START failed.
This patch solved this issue
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In its previous status, the first capture didn't work properly;
nothing was actually recorded from the microphone. This
behaviour was observed using a Visstrim M10 board.
In order to solve this BUG a workaround has been added that,
during the initialization process of the codec, powers on and
off the ADC.
The issue seems related to a HW BUG or some behavior that
is not documented in the datasheet.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Evalation of the WM5102 has identified a number of register values which
should be written after SYSCLK is enabled on revision A in order to
improve performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using fsi_set_master_clk() if it needs system clock.
But this function was called from
fsi_hw_shutdown()/fsi_dai_trigger()/fsi_resume() without a sense of unity.
Because of this, sound playback after suspend failed sometimes.
To keep consistency, fsi_master_clk() was called from
fsi_hw_start/stop() only now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many Arizona class devices contain ADSP2 cores with a standard method for
hooking them into the audio map. Define standard helpers for this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many current Wolfson devices feature DSPs based around an architecture
known as ADSP. Since there is a lot of commonality in the system
integration of these devices a common library will be used to provide
support for them.
This version provides equivalent support for ADSP1 to that currently
included in the WM2200 driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
We should really use "fck" when asking for the functional clock and not
"dmic_fck".
This way we can ensure that multiple dmic modules can exist in the system.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also drop the includes that are no longer needed and just
cause problems for the ARM common zImage.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
[tony@atomide.com: updated to drop unneeded headers]
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use regmap-mmio instead of open-coding caching and register accessors.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use devm_request_and_ioremap for requesting and mapping the IO region. This
makes the code a bit smaller and simpler.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add dB TLV ranges for the various volume controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ARIZONA_MICB1_ENA_SHIFT was used for micbias 2 and 3. This change
correctly uses the ARIZONA_MICBX_ENA_SHIFT for each corresponding DAPM
supply. This should not have caused any problems as the micbias enables
are in the same place in each register.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than always assuming the maximum possible BCLK rate will be
required generate BCLKs for stereo if either one or two channels is
enabled. In order to support this we also need to ensure that only
the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When lowering SYSCLK to 50kHz for accessory detection also lower the
AIFnCLK divisor to normalise the clocking configuration within the
device. This will not disrupt audio as we cannot support active audio
with such a low SYSCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This small reference boards has a Freescale P1022 dual-core PowerPC SOC
and a Wolfson Microelectronics WM8960 codec.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign each dai_link a unique name to avoid this run-time error.
[ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok
[ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97'
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removes the DaVinci private SRAM API and replaces it with
the genalloc API. The SRAM gen_pool is passed in pdata since
DaVinci is in the early stages of DT conversion.
[zonque@gmail.com: stub out gen_pool functions for
!CONFIG_GENERIC_ALLOCATOR]
Signed-off-by: Matt Porter <mporter@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable oldstatus is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix wm2200.c printk format warnings (seen on x86_64):
sound/soc/codecs/wm2200.c:1027:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm2200.c:1139:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int'
sound/soc/codecs/wm2200.c:1181:2: warning: format '%d' expects type 'int', but argument 7 has type 'size_t'
sound/soc/codecs/wm2200.c:1201:5: warning: format '%x' expects type 'unsigned int', but argument 3 has type 'long unsigned int'
sound/soc/codecs/wm2200.c:1264:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm2200.c:1328:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When switching to common clock driver for ux500 this clock needs to
be handled as well. Before this clock was internally managed by the
clock driver itself.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure clocks are being prepared and unprepared as well
as enabled and disabled.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch DAPMises headphone and lineout output enable controls.
Earlier these output enable bits were permanently turned on in probe.
In da9055 codec, right outmixer is directly connected with right HP and
Line out. This resulted in two side effects,
(1) When you only want to use lineout, right HP (and connected charge
pump) also gets enabled
(2) When you only want to use stereo HP, lineout also gets enabled
This patch adds three switches to select which output(s) should be
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <david.chen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.
Some ux500_msp_i2s patches clashed with:
b18e93a493
ASoC: ux500_msp_i2s: better use devm functions and fix error return code
... leaving the driver uncompilable. This patch fixes the
issues encountered.
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When of_parse_phandle() is used to find a device node, its
reference count is incremented by the helper. Once we're
finished with them, it's our responsibly to ensure they
are freed in the correct manor.
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bunch of updates to the regmap range support, the most important ones
being to rename "n_ranges" to "num_ranges" for consistency and to allow
block writes to cross page boundaries, making things more transparent.
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Merge tag 'regmap/range' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap into asoc-wm2200
regmap: Range API changes
A bunch of updates to the regmap range support, the most important ones
being to rename "n_ranges" to "num_ranges" for consistency and to allow
block writes to cross page boundaries, making things more transparent.
As of commit 99c2aa (firmware loader: fix creation failure of fw loader
device) we can have more than one firmware request outstanding at once so
there is no need to daisychain our requests any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This can be used to provide some additional settling time to ensure that
we don't start microphone detection while the microphone pin is connected
to one of the headphone pins.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is more idiomatic and ensures we don't try to do the ASoC card setup
until we've got all the required resources.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the device is now idle_bias_off these never have any useful effect,
the device will be brought to _OFF when idle, and will at best leave it
powered for longer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This commit add a sound codec driver for Silicon Laboratories 476x
series of AM/FM radio chips.
Signed-off-by: Andrey Smirnov <andrey.smirnov@convergeddevices.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all CODEC devices have three audio interfaces and the clock rates
which support these things vary. Support this by using driver data to
supply the clock rates and by only completing the parts of system setup
which are required for the system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sachin Kamat