The variables are unsigned so the test `>= 0' is always true,
the `< 0' test always fails. In these cases the other part of
the test catches wrapped values.
In dac_audio_write() there does not occur a test for wrapped
values, but the test appears redundant.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.
If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.
Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.
Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.
Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.
Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The power for the amplifier should be handled internally
by the tpa6130a2 driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix for typo in commit 8d50e447d1
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around. Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/asoc:
ASoC: remove unused #include <linux/version.h>
ASoC: S3C lrsync function made to work with IRQs disabled.
ASoC: Fix display of stream name in DAPM debugfs
ASoC: Clean up error handling in MPC5200 DMA setup
* topic/asoc: (226 commits)
ASoC: au1x: PSC-AC97 bugfixes
ASoC: Fix WM835x Out4 capture enumeration
ASoC: Remove unuused hw_read_t
ASoC: fix pxa2xx-ac97.c breakage
ASoC: Fully specify DC servo bits to update in wm_hubs
ASoC: Debugged improper setting of PLL fields in WM8580 driver
ASoC: new board driver to connect bfin-5xx with ad1836 codec
ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
ASoC: davinci: i2c device creation moved into board files
ASoC: Don't reconfigure WM8350 FLL if not needed
ASoC: Fix s3c-i2s-v2 build
ASoC: Make platform data optional for TLV320AIC3x
ASoC: Add S3C24xx dependencies for Simtec machines
ASoC: SDP3430: Fix TWL GPIO6 pin mux request
ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
OMAP: McBSP: Use textual values in DMA operating mode sysfs files
ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
ASoC: Select core DMA when building for S3C64xx
...
The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
Receiver-Amplifier and Headphone-Amplifier.
The datasheet for the ak4671 can find at the following url:
http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Chaithrika U S <chaithrika@ti.com>
Free socdev if snd_soc_init_card() fails.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an
integrated headphone driver. The WM8711/L is designed specifically for
portable MP3 audio and speech players. The WM8711/L is also ideal for
MD, CD machines and DAT players.
Signed-off-by: Mike Arthur <Mike.Arthur@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 analogue control is shared with other devices in the same
product line. Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.
Also allow 0 TDM slots to be configure (for use when disabling TDM).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).
In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.
Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.
Tested on my Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.
This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.
That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.
Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.
Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html
Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. fix "line over 80 characters" checkpatch warnings
2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead
3. fix typos
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().
If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.
This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.
The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.
Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.
Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.
[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrects a bug with ADC Inversion Switch in wm8974 codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.
The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.
The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf
[Slight edit to sort the ALL_CODECS entries -- broonie.]
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the flexibility of the WM9081 FLL this should never happen
in a real system.
Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to use the best value we picked, not the last value we
looked at.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without MODULE_LICENCE("GPL"), when built as a module it will fail
to load because it uses other GPL symbols from kernel.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the last in-kernel direct usage of driver_data, replace it with
the proper dev_get/set_drvdata() calls.
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
While the hardware is capable of some limited asynmmetric modes the
driver does not currently support those modes so tell applications
that only symmetric rates are available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note the slightly tricky cache usage in the volume update function due
to the requirement for a separate write for the VU bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TLV320AIC3x driver is currently the only user of the CODEC hw_read
operation and is jumping through some hoops in order to do so. In order
to support future refactoring to make the hw_read operation more usable
unwrap the usage in this driver to avoid its use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Nothing uses it and the existing hw_read operation needs to be
refectored so it's easier to remove it rather than work with it.
Support can be re-added if the code requires volatile registers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register cache space was not being allocated for the final register,
causing bugs when it was used. Allocate space for it.
Also ensure that the final register is displayed in sysfs.
[Commit message rewritten to document actual issue. -- broonie]
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add "set_tristate" callbacks for HiFi and Voice DAIs.
Machine drivers can enable and disable tristate for each
DAI with "snd_soc_dai_set_tristate" function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The wrong register cache variable was being used to provide the size for
the memcpy(), resulting in a copy of only a void * of data.
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
According to TRM, an external FET controlled by a 1.8V output signal
can be used to reduce the pop-noise heard when the audio amplifier is
switched on. It is suggested that GPIO6 of TWL4030 be used, but any
other gpio can be used instead. This is indicated in machine driver
with the following twl4030_setup_data members:
-hs_extmute. Set to 1 if board has support for EXTMUTE.
-set_hs_extmute. Set to a callback funcion to control an external gpio
line. Set to NULL if MUTE[GPIO6] pin is used.
Codec driver takes care of enabling and disabling this output during
the headset pop attenuation sequence.
Also add a delay to let VMID settle in ramp up sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8523 is a high performance stereo DAC with integral charge
pump providing 2Vrms line driver outputs using a single 3.3V power
supply rail.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The device is a mono device but it can read two channel data and
many I2S controllers only understand 2 channels.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls,
removing word "Switch" from HandsfreeL/HandsfreeR widget name
for avoiding to duplicate it.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Event for voice sidetone was being interpreted as an
analog HiFi bypass event because VSTPGA register offset
is less than ARXR2_APGA_CTL offset. Reordering the
register checks allows to handle voice digital bypass
event properly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AVDAC clk priority allows to determine the path ADC must
be connected when the codec is in option2 and both HiFi
and Voice paths are enabled.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to define second copies of mode and format, they're declared
with exactly the same type at the head of the function and there's no
conflict in their use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add basic support for integration with the regulator API to WM8580.
Since the core cannot yet disable biases when the CODEC is idle we
simply request and enable the regulators for the entire time the
driver is active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.
At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be enabled
below 24kHz and not enabled above that so remove the user visible
control for this and do it autonomously in the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
