Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now we have done bitwise NOT against the mask bits for the defines of
WM8900_REG_CLOCKING1_BCLK_MASK,
WM8900_REG_CLOCKING1_OPCLK_MASK and WM8900_LRC_MASK.
But we don't have the bitwise NOT against the mask bits for the defines of
WM8900_REG_CLOCKING2_DAC_CLKDIV,
WM8900_REG_CLOCKING2_ADC_CLKDIV and WM8900_REG_DACCTRL_AIF_LRCLKRATE.
It is error prone to mix the inconsistent meaning for different mask defines.
So lets make the defines for each mask to be corresponding to the bits
defines in datasheet. Don't add extra "bitwise NOT" to the defines.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After checking the datasheet, I think what we want to do here is to
clear the WM8900_REG_CLOCKING2_DAC_CLKDIV/WM8900_REG_CLOCKING2_ADC_CLKDIV/
WM8900_REG_DACCTRL_AIF_LRCLKRATE bits and then OR with div value.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the datasheet:
Format Control (05h)
BITS[3:2]
FMT[1:0] Audio data format selection
00 = right justified mode
01 = left justified mode
10 = I2S mode
11 = DSP mode
BIT[4] LRP Polarity selec for LRCLK/DSP mode select
0 = normal LRCLK poalrity/DSP mode A
1 = inverted LRCLK poarity/DSP mode B
For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003.
For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch add controls for setting cut-off for high pass and voice
filters of ADC and DAC. There are also switches to enable/disable
these filters.
Also removed hard coded, fixed values of these parameters used by
previous version of driver.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for ADC and DAC five band equalizers
available on DA7210 codec.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes the following sparse warnings:
sound/soc/tegra/tegra_das.c:215:8: warning: Using plain integer as NULL pointer
sound/soc/tegra/tegra_das.c:237:8: warning: Using plain integer as NULL pointer
sound/soc/tegra/tegra_pcm.c:370:32: warning: symbol 'tegra_pcm_platform' was not declared. Should it be static?
Signed-off-by: Olof Johansson <olof@lixom.net>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The COEF #0 value represents a sort of device id, so it's supposedly
constant while operation. Better to use the cached value instead of
reading it at each time from the performance POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a static table for detecting the codec renames.
Also clean up the error paths in each patch_*() function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to have as less time between McPDM shutdown,
and power down of the DAC on the twl6040 codec as possible.
Request core to ignore the pmdown_time for the playback
stream.
Backround: with the McPDM protocol we are sendning not only
the pure audio stream, but OMAP McPDM also transmits
additional information (for example offset cancellation).
If McPDM is stopped prior to the DAC this information will
be not sent to the codec, which can result noise rendered
by the twl6040 codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this flag codec drivers can indicate that it is desired
to ignore the pmdown_time for DAPM shutdown sequence when
playback stream is stopped.
The DAPM sequence will be executed without delay in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch takes care of reserved bits of headphone volume
register by using correct volume range.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current code defines AD193X_PLL_INPUT_MASK as (~0x6) which is quite
different from other MASK defines.
To make it consistent with other mask defines, define AD193X_PLL_INPUT_MASK
as 0x6 and change the code accordingly.
I think this change improves the readability.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
This patch also includes a comment fix in wm8990_set_dai_pll(),
if freq_in and freq_out are 0, what we do is to clear WM8990_PLL_ENA bit.
Thus the comment should be "Turn off PLL".
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | (1 << WM8990_AINRMUX_PWR_BIT)))
is false, we should clear WM8990_AINR_ENA bits instead of WM8990_AINL_ENA.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If (fakepower & ((1 << WM8400_INMIXR_PWR) | (1 << WM8400_AINRMUX_PWR)))
is false, we should clear WM8400_AINR_ENA bits instead of WM8400_AINL_ENA.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If (fakepower & ((1 << WM8991_INMIXR_PWR_BIT)|(1 << WM8991_AINRMUX_PWR_BIT))))
is false, we should clear WM8991_AINR_ENA bits instead of WM8991_AINL_ENA.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec->hw_read is broken now, let's covert to snd_soc_read.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec->hw_read is broken now, let's covert to snd_soc_read.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec->hw_read is broken now, let's covert to snd_soc_read.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the event handler is only used by the Earphone Driver, it is better
to rename it from twl6040_power_mode_event to twl6040_ep_drv_event.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is better to switch HS Power Mode (if it was in low power mode) before
we enable the Earpiece driver. The switched off EP driver can filter out
noise coming from the Low Power to High Performance transition on the
HSL DAC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no limitation dictated by outputs or inputs regarding to the
selected PLL (LP/HP).
Remove the checks for this, and allow all path with any PLL configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Capture is supported in all PLL configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_DM365_EXTERNAL_CODEC does not exist, so it's a useless default.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
ak4535_reg should be 8bit, but cache table is defined as 16bit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
ak4642 register was 8bit, but cache table was defined as 16bit.
ak4642 doesn't work correctry without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched.
codec->hw_read is broken now.
Here we use below trick to avoid writing to reserved registers while resume.
Write the register default value to cache for reserved registers,
so the write to the these registers are suppressed by the cache
restore code when it skips writes of default registers.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The audio_feature_info[] array should contain all entries for UAC2_FU_*,
but currently a few last entries are missing. Even though, the driver
tries to probe these entries in parse_audio_feature_unit() and may
access the range over the array. This patch fixes the bug by limiting
the loop size properly using ARRAY_SIZE() instead of a hard-coded
magic number.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds partial support for the Maschine controller by Native Instruments.
Supported now are the 1x1 MIDI interface and the 41 buttons, 11 endless
rotary encoders, and 16 pressure-sensitive drum pads. Still to work on are the
dimmable LEDs and the two monochrome screens.
Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a case where a newly-registered input device could be opened before
a necessary variable in the device structure was set. When code tried to use
the variable in the URB reply callback, it would cause an Oops.
This fix sets the aforementioned variable before calling input_register_device.
Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This typo caused headphone pins not to be initialized correctly.
BugLink: https://bugs.launchpad.net/bugs/871582
Reported-by: Effenberg
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The purpose of this patch is to remove a section of "bad" code that
assigns the last DAC to ports E or F in order to support notebooks
with docking in earlier days, around ALSA 1.0.19 - 21. This is not
necessary now and actually breaks some configurations that use these
ports as other devices. This have been tested on several different
configurations to make sure that it is working for different combinations.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In saarb_pm860x_init() and evb3_pm860x_init(), we call
pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn
calls pm860x_set_bits().
Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X.
This patch fixes below build error if CONFIG_MFD_88PM860X is not configured.
LD .tmp_vmlinux1
sound/built-in.o: In function `pm860x_write_reg_cache':
last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write'
sound/built-in.o: In function `pm860x_set_bias_level':
last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits'
last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write'
last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write'
sound/built-in.o: In function `pm860x_read_reg_cache':
last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read'
sound/built-in.o: In function `pm860x_probe':
last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read'
sound/built-in.o: In function `pm860x_codec_handler':
last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read'
last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read'
sound/built-in.o: In function `pm860x_mic_jack_detect':
last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits'
sound/built-in.o: In function `pm860x_hs_jack_detect':
last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits'
last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits'
last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits'
make: *** [.tmp_vmlinux1] Error 1
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patchs adds support for following,
(1) DAI 20 and 32 bit word sizes
(2) DAI left and right justified formats
(3) DAI slave mode
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Both Headset DAC need to be turned on/off at the same time before
any of the output drivers are enabled (HS Left/Right, Earpiece).
Move the HS DAC enable code to sequenced DAPM_SUPPLY, and attach
it to the DACs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
twl6040 have two vibra output drivers.
They can be operated with audio stream coming through
the PDM interface (fifth channel).
The vibra outputs can be controlled via the input/FF
driver as well.
Selection between the two mode is implemented within
the codec driver, the input/FF driver can only operate if
the routing is set to "Input FF".
Changing from "Input FF" to "Audio PDM" mode is protected
as well: The switchin can only be done, if there is no
running effect from the input/FF.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params.
Thus no need to check wm8994-aif3 in wm8994_hw_params.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is not required after multi-component patch.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM5100 includes an advanced, low power, accessory detect subsystem
capable of detecting both accessory presence and button presses while
the device is in an ultra low power mode. Implement initial support for
this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is currently named "TVL" instead of "TLV".
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Cc: Javier Martin <javier.martin@vista-silicon.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8983 can be reset by performing a write of any value to
the software reset register.
To avoid writing to the software reset register while resume,
we should write the same value in wm8983_reg_defs to software
reset register in wm8983_probe().
The write to the reset register is suppressed by the cache
restore code when it skips writes of default registers.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes via the snd_soc_card struct
to core. With this change we do not need the init function
since the remaining snd_soc_dapm_enable_pin calls are
not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes and static controls via the
snd_soc_card struct to core. In this way the machine driver
does not need to handle the DAPM widgets/routes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes via the snd_soc_card struct
to core. With this change we do not need the init function
since the remaining snd_soc_dapm_enable_pin calls are
not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Widgets are connected by default.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Widgets are connected by default.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to call soc_dapm_sync at init time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Anuj Aggarwal <anuj.aggarwal@ti.com>
Cc: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Gražvydas Ignotas <notasas@gmail.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The rpcm_file parameter is never used in current ALSA code, so remove
it to make it cleaner.
Signed-off-by: Feng Tang <feng.tang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Michael Opdenacker <michael.opdenacker@linaro.org>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We really should be doing this in the core, not in a driver...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
The ASoC code always uses -1 as the error code due to reporting errors in
band with the value. Ensure we don't confuse anything by making sure we
don't pass actual error codes back into the rest of the code on read.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The number of connected input and output endpoints for a given widgets
can't change during a DAPM run so there is no need to redo the recursion
through branches of the tree we've already visited. Doing this on one of
my test systems gives an improvement of:
Power Path Neighbour
Before: 63 607 731
After: 63 141 181
which scales up well as more widgets are involved in paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handling of user control elements was implemented for all types except
ENUMERATED. This type will be needed for the device-specific mixers of
upcoming FireWire drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This ensures none of the rest of the code ever encounters a widget which
does not have a power check function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
OMAP drivers are omitted as I know Peter already has patches for them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we only have one sync during the initial startup of the card by
making snd_soc_dapm_sync() a noop on non-instantiated cards. This avoids
any bounces due to things like jacks reporting their initial state on
partially initialised cards. The callers that don't also get called at
runtime should just be removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* pm-runtime:
PM / Tracing: build rpm-traces.c only if CONFIG_PM_RUNTIME is set
PM / Runtime: Replace dev_dbg() with trace_rpm_*()
PM / Runtime: Introduce trace points for tracing rpm_* functions
PM / Runtime: Don't run callbacks under lock for power.irq_safe set
USB: Add wakeup info to debugging messages
PM / Runtime: pm_runtime_idle() can be called in atomic context
PM / Runtime: Add macro to test for runtime PM events
PM / Runtime: Add might_sleep() to runtime PM functions
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Checking the pdata-flags used 'or', so the check is always true. Use 'and' to
correctly mask the flags.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Cc: Javier Martin <javier.martin@vista-silicon.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We synchronise jack state on startup - when we do that make sure that we
have set up all the DAPM widgets first in case we end up touching any of
the partially set up widgets when syncing the jack pins.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
The snd_soc_*_volsw_2r functionality has been merged to
*volsw callbacks.
Few places still used the get, or put variant of volsw_2r,
replace those with the corresponding *_volsw.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit ef18beded8 introduced a
mechanism to assign the previously used slot for the next reopen of a
PCM stream. But the PCM device number isn't always unique (it may
have multiple substreams), and also the code doesn't check the stream
direction, thus both playback and capture streams share the same
device number.
For avoiding this conflict, make a unique key for each substream and
store/check this value at reopening.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the speaker outputs are more than the headphone outputs, it implies
that the system has surround speakers while the headphones are only for
monitoring the front. In such a case, it's better to put speakers as
the primary outputs so that the driver can build up and keep the
surround setup. Otherwise the system will pick up the headphone as
primary, and offers less channels than the speakers do support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since hda_proc.c is now the only user of snd_print_pcm_rates(), better to
put it back locally to hda_proc.c and revert to the old style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SAD sampling rate information reported in
/proc/asound/cardX/eldX is incorrect due to a mismatch
between HDA and HDMI frequencies. Add new routine to provide
relevant values.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliminate below build warning:
CC sound/soc/codecs/wm9090.o
sound/soc/codecs/wm9090.c: In function 'wm9090_probe':
sound/soc/codecs/wm9090.c:550: warning: unused variable 'wm9090'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions are removed in commit f0fba2ad
"ASoC: multi-component - ASoC Multi-Component Support".
Let's remove the leftover function declaration in header file.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Let the user know, that the callback has been called with unexpected
register parameter.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core now have one callback function, which can handle
single, and double register mixer controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the put_volsw/put_volsw_2r in one function.
To avoid build breakage in twl6040 keep the
snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the get_volsw/get_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the info_volsw/info_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoid using the mc->rreg to identify the 2r type of gain control.
Introduce a variable to track this.
This change is needed to avoid breakage with the upcoming volsw volsw_2r
merger.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The custom get_volsw does not need to call any core get_volsw calls,
since we are returning the shadow values for the gains.
Return -EINVAL in the unlikely event, if the function has been called
for unhandled control. This way we can remove one check in the code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the macros for controls require custom get/put function.
This is to make sure that the soc_mixer_control is used
consistently among the drivers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Arun KS <arunks@mistralsolutions.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we run out of DACs when trying to assign a DAC to a secondary
headphone, prefer the DAC of the first headphone to the primary
(usually line out) DAC.
BugLink: http://bugs.launchpad.net/bugs/845275
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't really care how many widgets a supply is supplying, we just care
if the number is non-zero. This didn't actually produce any improvement
in the test cases I've been using but seems obviously sensible enough that
I'm pushing it out anyway.
We could do a similar thing for other widgets but this may be unhelpful
for further refactorings Liam was working on aiming to allow us to
identify connected audio paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The whole point of supply widgets is that they aren't inputs to their
sinks so a state change in a supply should never affect the state of the
widget being supplied and we don't need to mark them as dirty.
Power Path Neighbour
Before: 69 727 905
After: 63 607 731
This is particularly useful where supplies affect large portions of the
chip (eg, a bandgap supplying the analogue sections).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some widgets will get power_check() run on them more than once during a
DAPM run, most commonly due to supply widgets checking to see if their
consumers are powered up. It's wasteful to do this so cache the result
of power_check() during a run. For one system I tested this on I got an
improvement of:
Power Path Neighbour
Before: 106 970 1186
After: 69 727 905
from this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help diagnose why we're checking widgets by providing some logging when
we first dirty them. This should possibly be a trace point if it's useful
but can be absurdly verbose if enabled, we can always change it later if
desired.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://github.com/tiwai/sound:
ASoC: omap_mcpdm_remove cannot be __devexit
ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC
ASoC: use a valid device for dev_err() in Zylonite
If two widgets are not currently connected then there is no need to
propagate a power state change between them as we mark the affected
widgets when we change a connection. Similarly if a neighbour widget is
already in the state being set for the current widget then there is no
need to recheck.
On one system I tested this gave:
Power Path Neighbour
Before: 114 1066 1327
After: 106 970 1186
which is an improvement, although relatively small.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to reduce the number of DAPM power checks we run keep a list of
widgets which have been changed since the last DAPM run and iterate over
that rather than the full widget list. Whenever we change the power state
for a widget we add all the source and sink widgets it has to the dirty
list, ensuring that all widgets in the path are checked.
This covers more widgets than we need to as some of the neighbour widgets
won't be connected but it's simpler as a first step. On one system I tried
this gave:
Power Path Neighbour
Before: 207 1939 2461
After: 114 1066 1327
which seems useful.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're not actually doing any dynamic power management based on connection
and output drivers (which are pretty much the same thing) are marked as
unconditionally connected already.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We've got the same code in two different places, let's have it in a single
place instead.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Future patches will try to reduce the number of widgets we check on each
DAPM run but we're still going to need to look and see if the devices is
on at all so we can manage the overall device bias. Move these checks out
into the main dapm_power_widgets() function so we don't have to think about
them for now.
Once we're doing more incremental updates it'll probably be worth using
refcounts for each bias level to avoid having to do the sweep over all
widgets but that's not going to be where the big performance wins are.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Split the decision about what the new power should be out from the
implementation of that decision.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return -EINVAL in the unlikely event, if the function has been called
for unhandled control. This way we can remove one check in the code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not required after commit 8d50e447
"ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs"
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Writing to WM8971_RESET resets all registers to the default state.
Thus we should avoid writing to WM8971_RESET on resume.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For wm8994-aif2, the rate_reg should be WM8994_AIF2_RATE.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
snd_soc_update_bits() will only write new register value
if the old value is different from the new value.
In additional, snd_soc_update_bits() returns 0 for no change.
No need to read WM8995_CLOCKING_1 register before calling snd_soc_update_bits().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_update_bits() will only write new register value
if the old value is different from the new value.
In additional, snd_soc_update_bits() returns 0 for no change.
No need to read WM8994_CLOCKING_1 register before calling snd_soc_update_bits().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After omap_request_dma the BLOCK_IRQ is enabled as default
configuration for the channel.
If we are requested for no period wakeup, we need to disable
the BLOCK_IRQ in order to not receive any interrupts.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the biquad channel names to a separate array and iterate over it in
max98095_get_bq_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.
Also added a BUILD_BUG_ON to ensure that the bq_mode_name and controls
arrays are the same size.
Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the EQ channel names to a separate array and iterate over it in
max98088_get_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.
Also added a BUILD_BUG_ON to ensure that the eq_mode_name and controls
arrays are the same size.
Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM1811 is mostly register compatible with the WM8994 and WM8958,
providing a high performance audio hub CODEC in a small form factor
suitable for ultra compact system designs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ever since it was written the Samsung DMA driver has had a TODO in the
hw_free() function wondering if we need to flush the DMA buffers. Up until
now the answer has been no but with the recent improvements Boojin has
done to the DMA infrastructure for the Samsung port the answer has changed
to yes for at least S3C6410 systems.
If we don't then when we next prepare() the channel the API will get
confused trying to run callbacks on the transfers hanging around from the
previous time the stream was open and oops.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Boojin Kim <boojin.kim@samsung.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
This is mostly a static checker fix more than anything else. We're
copying from a 64 char buffer into a 44 char buffer.
The 64 character buffer is str[] in snd_mixer_oss_build_test_all().
The call tree is:
snd_mixer_oss_build_test_all()
-> snd_mixer_oss_build_test()
-> snd_mixer_oss_build_test().
We never actually do fill str[] buffer all the way to 64 characters.
The longest string is:
sprintf(str, "%s Playback Switch", ptr->name);
ptr->name is a 32 character buffer so 32 plus 16 characters for
" Playback Switch" still puts us over the 44 limit from "id.name".
Most likely ptr->name never gets filled to the limit, but we can't
really change the size of that buffer so lets just use strlcpy() here
and be safe.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a lock inversion between fwspk->mutex and pcm->open_mutex
reported by lockdep when fwspk_hw_free is called.
Fixed by copying the fix from the same former issue in the isight
sound driver (commit f3f7c1837f
"ALSA: isight: fix locking").
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAI init function may want to do something that needs the widgets to
be instantiated.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Include <linux/module.h> to fix below build error:
CC sound/soc/samsung/s3c-i2s-v2.o
sound/soc/samsung/s3c-i2s-v2.c:573: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:573: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:573: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:638: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:638: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:638: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:677: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:677: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:677: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c: In function 's3c_i2sv2_register_dai':
sound/soc/samsung/s3c-i2s-v2.c:736: warning: initialization discards qualifiers from pointer target type
sound/soc/samsung/s3c-i2s-v2.c: At top level:
sound/soc/samsung/s3c-i2s-v2.c:754: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:754: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:754: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:756: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/s3c-i2s-v2.c:756: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:756: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE'
sound/soc/samsung/s3c-i2s-v2.c:756: warning: function declaration isn't a prototype
make[3]: *** [sound/soc/samsung/s3c-i2s-v2.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Sigmatel/IDT parser should have the same naming convention
for input jacks as the other codecs have.
BugLink: http://bugs.launchpad.net/bugs/859704
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Any driver that selects SND_SOC_WM8994 should also make sure that
MFD_WM8994 is set, since the codec relies on the mfd code:
sound/built-in.o: In function `wm8994_read':
last.c:(.text+0x20160): undefined reference to `wm8994_reg_read'
sound/built-in.o: In function `wm8994_write':
last.c:(.text+0x20e68): undefined reference to `wm8994_reg_write'
This solves the problem by selecting the MFD driver directly
and adding extra 'depends on' statements to make sure that we
respect the dependencies of that driver.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Applications may want to read ELD information to
understand what codecs are supported on the HDMI
receiver and handle the a-v delay for better lip-sync.
ELD information is exposed in a device-specific
IFACE_PCM kcontrol. Tested both with amixer and
PulseAudio; with a corresponding patch passthrough modes
are enabled automagically.
ELD control size is set to zero in case of errors or
wrong configurations. No notifications are implemented
for now, it is expected that jack detection is used to
reconfigure the audio outputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After checking the code and datasheet, I think what we want in the second
snd_soc_update_bits call is to update WM8741_DACRMSB_ATTENUATION register
instead of WM8741_DACRLSB_ATTENUATION.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have __exit annotation for txx9aclc_generic_remove(),
thus add __devexit_p to wrap it.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mika Westerberg <mika.westerberg@iki.fi>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is exported via resume callback of struct dev_pm_ops rather than referenced
directly and so should be staticised.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This assignment is done by the snd_soc_register_codec so there is no need
to redo it in probe function of a codec driver.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This removes a few unnecessary type casts and avoids
sparse warnings.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a missing dependency that is required for random configurations.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
omap_mcpdm_remove is used from asoc_mcpdm_probe, which is an
initcall, and must not be discarded when HOTPLUG is disabled.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the confi
options."
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Cc: Jaswinder Singh <jassi.brar@samsung.com>
Cc: Ben Dooks <ben@simtec.co.uk>
Cc: Seungwhan Youn <sw.youn@samsung.com>
Cc: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
We have __devexit annotation for kirkwood_i2s_dev_remove(), thus add __devexit_p
at necessary place.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
We have __devexit annotation for wm8782_remove(), thus add __devexit_p at
necessary place.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to use two hw_params callbacks in sdp3430 and zoom2 as
thet are now identical. Use instead the same snd_soc_ops structure and
hw_params callback for both DAI links.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Before commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the
dai_link") expectation for omap-mcbsp was that snd_soc_dai_set_fmt is to be
called first in machine hw_params callback before other CPU DAI functions.
Thus it was enough that only omap_mcbsp_dai_set_dai_fmt cleared the
mcbsp->regs structure. [Note that this was pure convention, it's always
been OK to set things on init -- broonie]
Now this doesn't hold anymore since machine drivers can set the DAI format
only once on init time and thus mcbsp->regs may get out of sync when other
CPU DAI functions are modifying them dynamically with different values
between the calls. Therefore clear the accessed mcbsp->regs bits and
bitfields in other functions too.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current code set update bits for WM8753_LDAC and WM8753_RDAC twice,
but missed setting update bits for WM8753_LADC and WM8753_RADC.
I think it is a copy-paste bug in commit 776065
"ASoC: codecs: wm8753: Fix register cache incoherency".
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
A recent conversion has introduced references to &pdev->dev, which does
not actually exist in all the contexts it's used in.
Replace this with card->dev where necessary, in order to let
the driver build again.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Save model information in driver_data so we can simplify the implementation.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of checking, if the work is pending, it is safer to cancel
the pending work, or wait till the scheduled work finishes.
This way we can avoid modifying the variables used by the work
function.
Since we know that no work is pending, we can remove the two additional
checks in POST_PMU, and PRE_PMD for non pending works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the driver handled by out_drv_event have it's power
bit shifted by 3.
Remove the case for shift 3, and also add comment for the cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Headset gain range is 0 - 0xf (4 bit resolution)
The Handsfree gain range is 0 - 0x1d (5 bit resolution,
0x1e, and 0x1f values are invalid)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is a bit overkill to have three (3) separate
workqueue for a single driver.
We can manage things with one workqueue nicely.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No reason to use static variable for channel_index.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The name contains invalid valid character (/), which
causes problems when trying to create the debugfs
directory structure:
ASoC: Failed to create Aux/FM Stereo In debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Vaibhav Bedia <vaibhav.bedia@ti.com>
Cc: Sekhar Nori <nsekhar@ti.com>
Cc: Kevin Hilman <khilman@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit a810364a04
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put parentheses around macro argument uses. This avoids pitfalls
for the programmer, where the argument expansion does not give the
expected result, for example:
SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1);
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound_timer.info.name is a 32 character buffer. This function only
has one caller (in sound/oss/ad1848.c) and it passes as 128 character
buffer as "name". I don't know if this is a problem in real life,
and I doubt we're going to add more OSS drivers so it's unlikely to
become an issue. But we may as well take care of it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.
Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently codec->control_data is not initialized before calling
process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9093 is an enhanced version of the WM9093. Add the device ID to
the driver, further patches will add support for the additional features
in the WM9093.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are numerous broken references to Documentation files (in other
Documentation files, in comments, etc.). These broken references are
caused by typo's in the references, and by renames or removals of the
Documentation files. Some broken references are simply odd.
Fix these broken references, sometimes by dropping the irrelevant text
they were part of.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.
This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the initial bias level to standby during CODEC probe instead of leaving the
CODEC powered off.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Both *socdev and *codec of struct mfld_mc_private are not being used
in this driver, remove it.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the the internal oscillator is powered down when entering BIAS_OFF
state, but not re-enabled when going back to BIAS_STANDBY. As a result the
CODEC will stop working after suspend if the internal oscillator is used to
generate the sysclock signal. This patch fixes it by clearing the appropriate
bit in the power down register when the CODEC is re-enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers. Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.
The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.
Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Handsfree gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain
is -26dB.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Headset gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x0 raw at
one end of the range, and not in the middle.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The delayed_work named 'delayed_work' is for the headset detection,
so move it to the twl6040_jack_data struct.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The delayed works for the output can be moved within the
twl6040_output struct (from the twl6040_data) to be better
organized.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can manage with one set of get, and put function for the gain
controls we need to handle with custom code due to the shadowing
of the register.
For both get, and put function we can call decide based on the
mc->rreg value, if we need to call the volsw, or the vlosw_2r
variant (in 2r case rreg is not 0).
Handling of the shadow values are the same for both type of
controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This event handler is used with the OUT_DRV widgets.
The name pga_event was misleading.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the values from twl6040 codec (HSOTRIM L/R) we can configure
the McPDM offset cancellation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The offset cancellation values can be different from board to board, even
on the same HW platform.
Provide a way for the machine drivers to configure the McPDM offset
cancellation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin