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 MAINTAINERS entry for the ADI sound CODEC drivers currently only lists the
ADI devices-drivers-devel mailing-list. Add myself as additional contact, since
I'm the person at ADI who is currently doing most of the work on these drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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>
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>
The VIBSELL/R bit in the VIBCTLL/R register tells the source of the data,
which is going to be used to drive the attached motor(s).
Do not allow effect execution if any of the channels are set to receive
audio data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the client only interested, if any of the vibra channels enabled, or
if any of the channels are set to receive audio data via PDM.
This function targets mainly the vibra driver, so it can check if it is
allowed to execute effects ot not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <samuel.ortiz@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin