Detailed SoC changes v1.0.23 v1.0.24

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Detailed SoC changelog between 1.0.23 and 1.0.24 releases

alsa-driver

SoC PXA2xx Core

- arm/pxa2xx: reorgazine SSP and SPI header files
The PXA-SPI driver relies on some files / defines which are arm specific
and are within the ARM tree. The CE4100 SoC which is x86 has also the
SPI core.
This patch moves the ssp and spi files from arm/mach-pxa and plat-pxa to
include/linux where the CE4100 can access them.
This move got verified by building the following defconfigs:
cm_x2xx_defconfig corgi_defconfig em_x270_defconfig ezx_defconfig
imote2_defconfig pxa3xx_defconfig spitz_defconfig zeus_defconfig
raumfeld_defconfig magician_defconfig
- ASoC: fix SND_PXA2XX_LIB Kconfig warning
Fix following warning observed when SND_PXA2XX_SOC is set and SND_ARM isn't:
warning: (SND_PXA2XX_AC97 && SOUND && !M68K && SND && SND_ARM && ARCH_PXA ||
SND_PXA2XX_SOC && SOUND && !M68K && SND && SND_SOC && ARCH_PXA) selects
SND_PXA2XX_LIB which has unmet direct dependencies (SOUND && !M68K && SND &&
SND_ARM)
- ASoC: pxa-ssp: fix a memory leak in pxa_ssp_remove()
The "priv" allocated in pxa_ssp_probe() should be kfreed in pxa_ssp_remove().
- ASoC: Enable autoloading of pxa2xx CPU I2S driver with module alias
- ASoC: remove include of pxa2xx-pcm.h in pxa2xx-ac97.c
Fix reference to moved header file, which was unused anyway.
This change fixes below build error:
CC sound/soc/pxa/pxa2xx-ac97.o
sound/soc/pxa/pxa2xx-ac97.c:27:24: error: pxa2xx-pcm.h: No such file or directory
make[3]: *** [sound/soc/pxa/pxa2xx-ac97.o] Error 1
make[2]: *** [sound/soc/pxa] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: add saarb machine driver for 88pm860x
88PM860x codec is used in Marvell saarb development board. 88PM860x codec
is used as master mode for SSP communication. Only I2S format is supported.
- ASoC: add tavorevb3 machine driver for 88pm860x
- ASoC: update setting for pxa ssp slave mode
SCFR bit is required to be always set if pxa ssp is in slave mode. This bit
indicates clock input to SSPSCLK is only active during data transfers.
- ASoC: fix pxa2xx-pcm.h path
Since pxa2xx-pcm.h is removed from sound/soc/pxa, we need to update the
path in related files.
Tested-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
- ASoC: multi-component: Fix reference to moved header file, which was unused anyway.
Removed #include of pxa2xx-pcm.h
- [ARM] pxa: add namespace on ssp
In order to prevent code ambiguous, add namespace on functions in ssp driver.
- [ARM] pxa: remove incorrect select PXA_SSP in Kconfig
PXA_SSP is actually used by drivers like drivers/spi/pxa2xx_spi.c and
sound/soc/pxa/pxa-ssp.c, not by boards. Remove those incorrect 'select'
from Kconfig and make SOC_PXA_SSP to select.
- [ARM] pxa: move ssp into common plat-pxa
- [ARM] pxa: merge regs-ssp.h into ssp.h
No need to separate them as they should be together from the begining.

SoC Audio for Freecale i.MX1x i.MX2x CPUs

- ASoC: phycore-ac97: fix resource leak
Fix imx_phycore_init() error path and imx_phycore_exit() to properly free
allocated resources.
- ASoC: imx-ssi: fix resource leak
Fix imx_ssi_probe() error path and imx_ssi_remove() to properly free
allocated resources.
- ASoC: i.MX: we can do mono
Whether we can do mono or not depends on the codec. No need
to limit this in the ssi driver.
- phycore-ac97: add ac97 to cardname
We have different codecs on the pcm038 (ac97 wm9712 and mc13783).
To make alsactl restore work correctly these should have different
names.
- ASoC i.MX: switch to new DMA api
- ASoC i.MX: register dma audio device
We have two different transfer methods on i.MX: FIQ and DMA. Since
the merge of the ASoC multicomponent support the DMA device is lost.
Add it again. Also, imx_ssi_dai_probe has to be called for !AC97
aswell.
- ASoC i.MX phycore ac97: remove unnecessary includes
- ASoC i.MX eukrea tlv320: Fix for multicomponent
- ASoC: Remove -dai suffix from i.MX SSI driver
While it is a generic serial port in practice the i.MX SSI is only supported
in Linux as an audio port (the i.MX has dedicated SPI controllers and so on).
This means we don't need to disambiguate against other uses of the hardware
and so can drop the -dai suffix from the driver name which fixes merge
issues with the i.MX tree in -next.
- ASoC: i.MX ssi: use SSI_STCCR in synchronous mode
In synchronous mode the SSI_SRCCR values are ignored. Instead
SSI_STCCR must be used for both receiving and transmitting.
- ASoC: Fix for changed Eureka Kconfig symbol names
- fix comment typos concerning "challenge"
- ASoC: eukrea-tlv320: add support for cpuimx35sd
- ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
- ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
- ASoC: clean i.MX Kconfig
- ASoC: eukrea-tlv320: add support for our i.MX25 board
* tdm slot has to be configured to get sound working on i.MX25
- ASoC: imx: add eukrea-tlv320
Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.
- ASoC: imx-ssi.c: add new choices to platform configuration
* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
IMX_SSI_NET : enable Network Mode
IMX_SSI_SYN : enable Synchronous Mode
IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.
- ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1
- ASoC: Add missing Kconfig entry for Phytec boards
- ASoC: Update Freescale i.MX SSI driver DMA parameter handling
This updates the i.MX SSI driver to make it compatible with the ASoC tree
following the move of DMA parameters from the DAI to the audio substream
object.
- ASoC: imx-ssi: do not call hrtimer_disable in trigger function
Doing so causes a deadlock, so just signal the timer to stop
using an atomic variable.
- ASoC: imx-ssi: increase minimum periods to 4
Currently the notification of elapsed periods is not very exact.
Increase minimum periods to 4 as suggested by Liam Girdwood.
- ASoC: imx-ssi: Use a hrtimer in FIQ mode
Using a regular timer results in poll times < 1 jiffie with small
buffers, so we loaded the timer with the actual jiffie value. We can
be more accurate using a hrtimer. Also, we have to call
snd_pcm_period_elapsed after playing period_bytes and not
runtime->period_size (which is in samples and not in bytes).
- ASoC: imx-pcm-dma-mx2: restart DMA after an error
- ASoC: imx-ssi: honor IMX_SSI_DMA flag
When checking if we are DMA capable we have to check for the
IMX_SSI_DMA flag which is already set from platform_data instead
of setting it again when we want to do DMA.

SoC Audio for TXx9

- DMAENGINE: extend the control command to include an arg
This adds an argument to the DMAengine control function, so that
we can later provide control commands that need some external data
passed in through an argument akin to the ioctl() operation
prototype.
[dan.j.williams@intel.com: fix up some missed conversions]
- DMAENGINE: generic slave control v2
Convert the device_terminate_all() operation on the
DMA engine to a generic device_control() operation
which can now optionally support also pausing and
resuming DMA on a certain channel. Implemented for the
COH 901 318 DMAC as an example.
[dan.j.williams@intel.com: update for timberdale]
- MIPS: TXx9: Add missing MODULE_ALIAS definitions for TXx9 platform devices
This enables autoloading of the TXx9 sound driver on RBTX4927.
To: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Patchwork: http://patchwork.linux-mips.org/patch/1101/

SoC Audio for the Atmel AT32/AT91 System-on-Chip

- ASoC: Fix codec device id format used by some dai_links
The id part of an I2C device name is created with the "%d-%04x" format string.
So for example for an I2C device which is connected to the adapter with the id 0
and has its address set to 0x1a the id part of the devices name would be
"0-001a".
Currently some sound board drivers have the id part the codec_name field of
their dai_link structures set as if it had been created by a "%d-0x%x" format
string. For example "0-0x1a" instead of "0-001a".
As a result there is no match between the codec device and the dai_link and no
sound card is instantiated.
This patch fixes it.
- ASoC: Do not include soc-dapm.h
There is no need to include soc-dapm.h since soc.h includes it.
- ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path
Fix the error path to properly free allocated resources.
- ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path
- ASoC: atmel: test wrong variable
After clk_get() mclk is checked second time instead of pllb check.
In patch v1 Jarkko Nikula noticed that PTR_ERR() is also has wrong argument.
- Fix Atmel soc audio boards Kconfig dependency
Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC
audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing
clk_set_parent and clk_set_rate when building without
AT91_PROGRAMMABLE_CLOCKS.
- ASoC: Add simplfied device registration for Atmel SSC devices
Since the SSC is already being registered as a device under arch and
the DMA and SSC hardware are pretty much the same provide a simplified
device registration function for the Atmel SSC which will add the
ASoC-specific devices within the ASoC code, parenting the SSC device
off the actual SSC device. Also use it in the sam9g20-ek driver.
- ASoC: Fix device name for AT91SAM9G20-EK devices
A couple of typos in the multi-component conversion.
- ASoC: Support non-crystal master clocks for WM8731
Instead of unconditionally enabling the crystal oscillator on the WM8731
only enable it when explicitly selected via set_sysclk(), allowing machine
drivers to specify that they drive a clock into MCLK alone. This avoids
any conflicts between the oscillator and the external MCLK source and saves
power for systems which do not need the oscillator.
This should also deliver a small power saving on systems using the crystal
since the oscillator will only be enabled when the ADC or DAC is active.
- ASoC: atmel: trivial code cleanup
Remove break after return, it is not needed.
- ASoC: atmel: patch for the unnecessary variable removal
The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.

SoC Audio for the Samsung S3C24XX chips

- Move to build stub soc/samsung
- ASoC: Samsung: Rename from s3c24xx to samsung
Finally, move the 's3c24xx' directory to 'samsung'
- ASoC: Fix compile breakage in jz4740.c and smartq_wm8987.c
Commit ce6120c require that soc-dapm.h cannot be included before soc.h but
these two drivers were not checked. Fix them by including only soc.h as it
includes soc-dapm.h.
- ASoC: RX1950: Fix hw_params function
Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis
when MPLLin source for master clock is selected, prescaler has
no effect. Remove dividor calculation for 44100 rate; remove 88200
rate at all, rx1950 can't do it.
- ASoC: s3c24xx: Fix compilation problem for mini2440
When make mini2440_defconfig compilation end with undefined
references to DMA functions. There was missing selection
for S3C2410_DMA when compile ASoC audio for S3C24xx CPU.
Tested on mini2440 board.
- ASoC: SAMSUNG: Add Machine driver for S/PDIF PCM audio
This patch add S/PDIF machine driver to support S/PDIF PCM audio
on SMDKC100, SMDKC110 and SMDKV210 boards.
- ASoC: SAMSUNG: Add S/PDIF CPU driver
This patch adds S/PDIF CPU driver for various Samsung SoCs.
- Fix typo configue => configure in comments
- ASoC: rx1950: Fix clkdiv for 16khz and 48khz
Usage of 256 as clkdiv gives better rounding error (<1%)
for 16khz and 48khz
- ASoC: rx1950: remove unnecessary headers
- ASoC: rx1950: check that machine is rx1950 in glue driver
- ASoC: Add debug logging for s3c-ac97 resets
Helps tracing errors further up the stack.
- ASoC: Convert s3c-ac97 to pr_() macros
Could use dev_() but we'd have to remember the struct device somewhere
and it wouldn't make the logging clearer.
- ASoC: Clean up the CODEC device as well as the board for SMDK WM9713
Otherwise we try to re-register the CODEC device if the module is reloaded
and sysfs becomes miserable.
- ASoC: Clarify naming for SMDK WM9713 driver
At least some of the systems using this device have multiple audio
subsystems so provide some guidance to userspace about which one this
is.
- ASoC: Remove unneeded WM9713 header include from SMDK WM9713 driver
- ASoC: Prototype s3c64xx_i2s_get_clock()
So machine drivers can see the declaration.
- ASoC: Samsung: Debug PCM snd_soc_dai_driver registration
Each of the two PCM controllers need to be registered during probe
with appropriate 'name' of the dai driver.
- ASoC: Set more meaningful name for SMDK64xx WM8580 audio
Since the SMDK64xx boards have two audio subsystems using the board
name as the card name by itself isn't so user friendly as it might
be.
- ASoC: S3C: Fix PCM TXFIFO_DIPSTICK value
This patch modify FIFO_DIPSTICK value of PCM TX FIFO to be a optimal one.
Privious value (0x20) did not support 'Almost_full' of PCM FIFO for the DMA
request.
- ASoC: S3C: Fix PCM RX FIFO settings
When PCM capture, sound recorded abnormally because of RX FIFO
threshold settings are missing. So, This patch modify PCM RX FIFO
setting codes same as TX.
And for DMA, if PCM RXFIFO_DIPSTICK is not '0', it doesn't effect
to DMA request, because DMA refer RX_FIFO_EMPTY flag as the DMA
request.
- ASoC: Samsung: Debug PCM platform device name
The PCM controller platform devices are registered by the
name 'samsung-pcm', so use the same in the CPU driver.
- ASoC: S3C: AC97: Remove the -dai suffix
Drop the invalid -dai suffix appended to the Samsung AC97 CPU DAI.
- ASoC: Add HP iPAQ RX1950 support
- ASoC: Automatically calculate clock ratio for WM8580
Implement set_sysclk() and then rather than assuming 256fs use the
supplied value to calculate and configure the clock ratio for the
currently used sample rate. As a side effect we also end up
implementing clock selection for the ADC path.
In order to avoid confusion remove the existing set_clkdiv() based
configuration of the clock source for the DAC and update the SMDK64xx
driver (which is the only in-tree user of the CODEC).
- ASoC: Add a bit of resource unwinding in the S3C IISv4 driver
There's much more needed but this'll get us started.
- ASoC: Add MODULE_ALIAS to Samsung DAI drivers
- ASoC: multi-component - Add Goni sound driver
This patch add sound support for the Goni board based on S5PV210.
The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
 : When TV-OUT cable is inserted on Goni board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
 : When TV-OUT cable is inserted on Goni board,
the TV-OUT cable is connected to television.
- ASoC: multi-component - Add Aquila sound driver
This patch add sound support for the Aquila board based on S5PC110.
The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
 : When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
 : When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable is connected to television.
ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
This patch modify the wrong field name on Aquila board.
- ASoC: Document CFG switch settings for SMDK6410 WM8580 usage
Sadly these aren't soft controllable and can't be read back either :(
- ASoC: Remove /s from widget names on SMDK64xx WM8580
Otherwise debugfs gets upset when we try to create filenames with /
in them.
- sound: Remove pr_<level> uses of KERN_<level>
- ASoC:Support Samsung SoC(S5P) in I2Sv2
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).
- ASoC: Invert speaker enabling behaviour in SmartQ sound driver
The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.
- ASoC: Add SmartQ sound driver
This adds sound support for the SmartQ board.
The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.
- ASoC: s3c: patch for the unnecessary variable 'state' removal
The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.
- ASOC: S5PV210: Enable AC97 support
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.
- ASOC: S5PC100: Enable AC97 support
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.
- ASoC: SAMSUNG: I2S: Add bit definitions
Define more bit definitions in the order of mainline
support for the SoC.
- ASoC: Allow active paths from the GSM modem while the GTA02 is suspended
- ASoC: SMDK64XX: Switch to IISv4 CPU driver
Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.
- ASoC: S3C64XX: IISv4: Add CPU driver
Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.
- ASoC: S3C: I2S: Move set_sysclk to common code
Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.
- ASoC: S3C: I2Sv2: New field for controller feature
In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c
While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.
- ASoC: S3C64XX: I2S: Use s3c2412 defines
Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.
- ASoC: S3C: I2Sv2: Unify i2s_get_clock callback
Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock
- ASoC: S3C: I2Sv2: Discard redundant field iis_clk
No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.
- ASoC: S3C2412: I2S: Return correct source clock
Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.
- ASoC: S3C2412: I2S: Debug IMS field
The IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.
- ASoC: S3C: I2Sv2: Move defines closer to driver
The header for I2Sv2
linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/

SoC Audio for the Samsung chips

- Move to build stub soc/samsung
- ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s
During the multi-component patch the s3c24xx i2s driver was renamed from
"s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not
updated to reflect this change as well.
As a result there is no match between the dai_link and the i2s driver and no
sound card is instantiated.
This patch fixes the problem by updating the sound board drivers to use
"s3c24xx-iis" for the cpu_dai_name.
- ASoC: SAMSUNG: Clean-up header includes
This patch remove including unnecessary/duplicated headers which relative
with Samsung SoCs.
- ASoC: SAMSUNG: Move PCM specific definitions into pcm.c
This patch moves PCM specific definitions, they doesn't need to be
shared, into 'pcm.c' from 'pcm.h'.
- ASoC: SAMSUNG: Clean-up DMA header file
This patch moves DMA specific definitions, they doesn't need to be
shared, into 'dma.c' from 'dma.h'. And remove unnecessery definitions
on 'dma.h'.
- ASoC: SAMSUNG: Remove AC97 header file
This patch moves AC97 specific definitions, they doesn't need to be shared,
into 'ac97.c' from 'ac97.h'. This patch also remove header file, 'ac97.h',
and remove includes on machine drivers.
- ASoC: RX1950: Enable Mic Jack during glue driver init
Enable Mic Jack during glue driver init, otherwise capture will not work.
- ASoC: Change Samsung Kconfig from ASOC_ to SND_SOC_
The rest of ASoC is using SND_SOC_ as the prefix for all the Kconfig
symbols so do so for the new Samsung drivers too, rather than using
ASOC_ as they currently are.
- ASoC: SAMSUNG: Debug wrong parameter
snd_soc_jack_new()'s first parameter was changed from snd_soc_card to
snd_soc_codec after Multi-Component support patches. So, this patch
fixes parameter that we missed.
- ASoC: SMDKV310: Add I2S support
Add ASoC machine driver for SMDKV310/C210 boards that have
a WM8994 attached to I2S-0.
- ASoC: SMDKV310: Enable AC97 device
Enable AC97 audio device on SMDKV310/C210.
- ASoC: SMDKC110: Enable I2S device
- ASoC: SMDKV210: Enable I2S device
- ASoC: SMDK6442: Enable I2S device
- ASoC: SMDK6450: Enable I2S device
- ASoC: SMDK6440: Enable I2S device
- ASoC: SMDK_WM8580: Make I2S0 as default dai
Since most newer SMDKs have I2S0 routed to the WM8580's Primary DAI,
future changes can be minimized if the default CPU DAIs are set to
0, rather than 2.
- ASoC: Samsung: I2S: Flush FIFO after stop
Flush the FIFO while stopping the channel rather than starting.
This saves time during stream start and keeps the FIFOs clean
when the channel is idling.
- ASoC: Samsung: Set default rclk source rate
Since the rclk_srcrate is cleared upon startup, it should be
initialized upon second and later 'open' calls to the device
with same root-clock source. The bug is otherwise visible in
Codec-Slave mode.
- ASoC: Add HP iPAQ H1940 support
Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on
H1940.
- ASoC: SAMSUNG: Remove duplicated snd_card on smdk_spdif
This patch remove duplicated snd_card defination on smdk_spdif.
- ASoC: SAMSUNG: Fix initial return value
This patch fixed intial return value to be a '0' as asuccess on
set_audio_clock_heirachy(). This avoids unintended error on initialize.
- ASoC: Fix resource leak in smdk_spdif.c
Properly free allocated resources in smdk_init() error path.
Add missing platform_device_unregister() in smdk_exit().
- ASoC: Add missing gpio_free(S3C64XX_GPK(12)) in smartq_exit()
- ASoC: Fix resource leak in neo1973_gta02_init() error path
Properly free allocated resources in neo1973_gta02_init() error path.
- ASoC: Fix resource leak in goni_wm8994.c
Properly free allocated resources in goni_init() error path.
Add missing snd_soc_unregister_dai() in goni_exit().
- ASoC: Make SMDK WM8580 driver unloadable
Supply a module exit function so that the driver can be unloaded.
- ASoC: Fix compile error for smartq_wm8987.c
Fix below compile error:
CC sound/soc/samsung/smartq_wm8987.o
sound/soc/samsung/smartq_wm8987.c: In function 'smartq_hifi_hw_params':
sound/soc/samsung/smartq_wm8987.c:42: error: 'struct snd_soc_pcm_runtime' has no member named 'dai'
sound/soc/samsung/smartq_wm8987.c:43: error: 'struct snd_soc_pcm_runtime' has no member named 'dai'
sound/soc/samsung/smartq_wm8987.c: In function 'smartq_wm8987_init':
sound/soc/samsung/smartq_wm8987.c:192: warning: passing argument 1 of 'snd_soc_jack_new' from incompatible pointer type
sound/soc/samsung/smartq_wm8987.c: At top level:
sound/soc/samsung/smartq_wm8987.c:216: warning: initialization from incompatible pointer type
make[3]: *** [sound/soc/samsung/smartq_wm8987.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: neo1973_gta02_wm8753: fix wrong parameter for snd_soc_register_dai and snd_soc_unregister_dai
- ASoC: smdk_wm9713: fix resource leak in smdk_init error path
Fix the error path to properly free allocated resources.

SoC Blackfin

- ASoC: Fix codec device id format used by some dai_links
The id part of an I2C device name is created with the "%d-%04x" format string.
So for example for an I2C device which is connected to the adapter with the id 0
and has its address set to 0x1a the id part of the devices name would be
"0-001a".
Currently some sound board drivers have the id part the codec_name field of
their dai_link structures set as if it had been created by a "%d-0x%x" format
string. For example "0-0x1a" instead of "0-001a".
As a result there is no match between the codec device and the dai_link and no
sound card is instantiated.
This patch fixes it.
- ASoC: Blackfin: fix DAI/SPORT config dependency issues
While I2S/TDM/AC97 DAI is built-in, others are compiled as modules,
SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols.
Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too.
- ASoC: Blackfin TDM: use external frame syncs
We don't want to use internal frame syncs otherwise we sometimes
get out of sync, so don't enable them when setting up the SPORT.
- ASoC: Blackfin AC97: fix build error after multi-component update
We need to tweak how we query the active capture/playback state after
the recent overhauls of common code.
- ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update
One spot was missed in this driver when converting from
snd_soc_dai.private_data to snd_soc_dai_get_drvdata.
- ASoC: add AD1980 obsolete information
This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.
- ASoC: Remove unnecessary casts of private_data

SoC Codec 88PM860x

- ASoC: Yet more x86 tracepoint workarounds
- ASoC: codecs: Remove unused reg_cache fields from device structs
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but there are quite a few drivers left which now have an unused reg_cache field in
their private device struct.
This patch removes these unused fields.
- ASoC: ifdef out trace points from modules for x86
No idea why this works on ARM but not x86.
- ASoC: Add jack IRQ trace to 88pm860x driver
- ASoC: Simplify pm860x_probe error handling
Simplify pm860x_probe error handling and return actual error code we got.
- ASoC: add 88pm860x codec driver
Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it
also supports headset/mic/hook/short detection.

SoC Codec AC97

- ASoC: ac97: add MODULE_ALIAS for the platform driver
So that modprobe can load the driver automatically when the platform device
appears.
- ASoC: ac97: don't call snd_soc_new_ac97_codec at probe
It is not needed since snd_ac97_mixer() will create a new ac97 object for us.
Removing the call also fixes a memory leak since codec->ac97 is set to NULL at
the beginning of snd_ac97_mixer().
- ASoC: Clean up AC'97 glue driver
Remove version number and clean up some indentation.

SoC Codec AD1836

- ASoC: Fix incorrect kfree in ad1836_probe error path
We allocated memory for ad1836 in ad1836_spi_probe,
and will free the memory in either ad1836_spi_probe error path or
ad1836_spi_remove.
Thus we should not call kfree(ad1836) in ad1836_probe, otherwise
we have double free of ad1836.
- ASoC: ad1836: fix a memory leak if another ad1836 is registered
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).
- ASoC: Add indirection for CODEC private data
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

SoC Codec AD193X

- ASoC: codecs: Remove unused reg_cache fields from device structs
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but there are quite a few drivers left which now have an unused reg_cache field in
their private device struct.
This patch removes these unused fields.
- ASoC: Fix incorrect kfree in ad193x_probe error path
We allocated memory for ad193x in ad193x_spi_probe,
and will free the memory in either ad193x_spi_probe error path or
ad193x_spi_remove.
Thus we should not call kfree(ad193x) in ad193x_probe, otherwise
we have double free of ad193x.
- ASoC: ad193x: add set_sysclk entry to support different clock input
- ASoC: ad193x: fix typo, delete redundant space
- ASoC: ad193x: fix wrong register setting in ad193x_set_dai_fmt
- ASoC: Add indirection for CODEC private data
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

SoC Codec AD1980

- ASoC: Staticise AD1980 DAI
It doesn't need to be exported with multi-component.
- ASoC: ad1980 - set reg_cache_default to ad1980_reg
- ASoC: ad1980: remove unneeded function declaration
- ASoC: add AD1980 obsolete information
This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.

SoC Codec AD73311

- ASoC: sound/ad73311: add missing __devexit marker
This fixes the following warning:
sound/soc/codecs/ad73311.c:50:12: warning: 'ad73311_remove' defined but not used

SoC Codec AK4104

- ASoC: Add indirection for CODEC private data
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

SoC Codec AK4535

- ASoC: ak4535: Improve readability for setting mute
The mute/unmute is controled by SMUTE (Soft Mute Control bit):
0: Normal Operation (Default)
1: DAC outputs soft-muted
I think this change improves readability.
- ASoC: Remove unneeded suspend bias managment from CODEC drivers
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>

SoC Codec AK4642

- ASoC: ak4642: make sure name of register/value
This patch replace magic code with defined name,
and remove unnecessary settings which set default value
- ASoC: ak4642: simultaneous playback/recorde support
- ASoC: ak4642: code clean up
- ASoC: ak4642: Revive ak4642_snd_controls
This patch revive ak4642_snd_controls which was removed on
f0fba2ad1b6b53d5360125c41953b7afcd6deff0
- ASoC: ak4642: fix a memory leak if failed to initialise AK4642
ak4642 should be kfreed if ak4642_init() return error.
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- ASoC: ak4642: Add Digital Playback Volume control
- ASoC: header cleanup for ak4642

SoC Codec AK4671

- ASoC: codecs: Remove unused reg_cache fields from device structs
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but there are quite a few drivers left which now have an unused reg_cache field in
their private device struct.
This patch removes these unused fields.
- ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
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.

SoC Codec CQ0093 Voice

- ASoC: Do not include soc-dai.h
There is no need to include soc-dai.h since soc.h includes it. Convert
drivers to include only soc.h.
- ASoC: Remove unneeded suspend bias managment from CODEC drivers
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>

SoC Codec CS4270

- ASoC: Update users of readable_register()/volatile_register()
- ASoC: cs4270: use the built-in register cache support
Update the CS4270 driver to use ASoC's internal codec register cache feature.
This change allows ASoC to perform the low-level I2C operations necessary to
read the register cache. Support is also added for initializing the register
cache with an array of known power-on default values.
The CS4270 driver was handling the register cache itself, but somwhere along
the conversion to multi-compaonent, this feature broke.
- ASoC: codecs: Remove unused reg_cache fields from device structs
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but there are quite a few drivers left which now have an unused reg_cache field in
their private device struct.
This patch removes these unused fields.
- ASoC: Staticise CS4270 DAI
It's not needed with multi-component.
- ASoC: Remove export of CS4270 DAI
Not needed with multi-component.

SoC Codec CS42L51

- ASoC: Change my mail address
Like other coworkers, I'm about leave Mandriva/Edge-It so I'm changing
my mail address to use my personal one.
- ASoC: OpenRD Client : Fix naming breakage due to multicomponent support
multicomponent support added/changed some device name but added some typos,
breaking existing OpenRD Client support.
- ASoC: Add driver for cs42l51
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.

SoC Codec CX20442

- ASoC: Move pop time from DAPM context to sound card
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
- ASoC: Use a more adequate name for the CX20442 codec DAI
In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.
Use a better name, "cx20442-voice", as suggested by Liam Girdwood.

SoC Codec DA7210

- ASoC: da7210: code clean up
- ASoC: da7210: fix a memory leak if failed to initialise da7210 audio codec
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.
- ASoC: da7210: Add HeadPhone Playback Volume control
HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.
- ASoC: header cleanup for da7210
- ASoC: da7210: Fencepost error in reg cache read
An index equal to the array size may not be accessed.

SoC Codec DIT SPDI/F

- ASoC: Fix S/PDIF build
- ASoC: spdif: Add codec driver to use spdif stand-alone
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.

SoC Codec General Digital MICs

- ASoC: DMIC codec: Adding a generic DMIC codec
This codec is to be used by the DMIC driver to
control the DMIC codec. This driver will be used on future
implementations of the DMIC driver to support codec specific
features.
At this time, the codec driver just registers the codec DAI.

SoC Codec MAX98088

- ASoC: Update users of readable_register()/volatile_register()
- ASoC: codecs: max98088: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the max98088 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the max98088 driver to use the
generic register cache in its private functions.
- ASoC: max98088 - fix a memory leak
- ASoC: Add support for MAX98089 CODEC
This patch adds initial support for the MAX98089 CODEC.
- ASoC: sound/max98088: add missing __devexit marker
This fixes the following warning:
sound/soc/codecs/max98088.c:2054:12: warning: 'max98088_i2c_remove' defined but not used
- ASoC: max98088: Staticise m98088_eq_band
This function is not exported and it does not seem to be called from
anywhere else therefore it should be static.
- ASoC: Restore MAX98088 CODEC driver
This reverts commit f6765502f8daae3d237a394889276c8987f3e299 and adds
the missing include file.
- Revert "ASoC: Add max98088 CODEC driver"
This reverts commit e65d255a5a610ce9e185e0df855250706aca7c21 since it
does not build.
- ASoC: Add max98088 CODEC driver
This patch adds the MAX98088 CODEC driver.

SoC Codec Philips UDA134x

- ASoC: uda134x - set reg_cache_default to uda134x_reg
After checking the code in 2.6.36,
I found this is missing during multi-component conversion.
- ASoC: uda134x: correct bias level setup for codecs family
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.
- ASoC: uda134x: add DATA011 register found in codecs family
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.
- ASoC: uda134x: fix bias level setup on initialization
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.
- ASoC: uda134x: replace a macro with a value in platform struct.
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
- ASoC: UDA134X: Add UDA1345 CODEC support
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.

SoC Codec Philips UDA1380

- ASoC: UDA1380: Add delay between power on and reset
- ASoC: uda1380: make driver more powersave-friendly
Disable some codec modules in standby mode, completely disable
codec in off mode to save some power.
Fix suspend/resume: mark mixer regs as dirty on resume to
restore mixer values, otherwise driver produces no sound
(master is muted by default).

SoC Codec SSM2602

- ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
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.
- ASoC: Remove unneeded suspend bias managment from CODEC drivers
The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>

SoC Codec STAC9766

- ASoC: stac9766 - set reg_cache_default to stac9766_reg
Looks like this is missing during multi-component conversion.

SoC Codec TI sn95031

- ASoC: sn95031 fix the code style and format inconsistencies
this patch fixes inconsistencies commented by Mark.
This also fixes few other style things in audio_map & header file
- ASoC: sst v2: Add sn95031 codec driver
This patch adds the sn95031 asoc codec driver. This driver currently
supports only playback. Capture and jack detection to be added later

SoC Codec TLV320AIC23

- sound: codecs/tlv320aic23: fix bias management for suspend/resume
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.
in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF

SoC Codec TLV320AIC26

- ASoC: missing conversions to snd_soc_codec_*_drvdata()
Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.

SoC Codec TLV320AIC3X

- ASoC: Move DAPM paths from DAPM context to snd_soc_card
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
- ASoC: Remove unused aic3x_i2c_init and aic3x_i2c_exit functions
- ASoC: tlv320aic3x - fix variable may be used uninitialized warning
If aic3x_read failed , val is used uninitialized.
Fix it by initializing val to 0.
This patch fixes below compile warning:
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio':
sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected':
sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed':
sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function
- ASoC: Fix incorrect kfree in aic3x_probe error path
We allocated memory for aic3x in aic3x_i2c_probe,
and will free the memory in either aic3x_i2c_probe error path or
aic3x_i2c_remove.
Thus we should not call kfree(aic3x) in aic3x_probe, otherwise
we have double free of aic3x.
- ASoC: tlv320aic3x: Add support to shared common reset line
This is aimed to configurations where multiple aic3x codecs share the same
reset line and are powered from same supply voltages.
Currently aic3x_probe will fail if trying to request already requested
gpio_reset and passing -1 to another aic3x instances cause that those
instances cannot release reset in aic3x_set_power. That is, another
instances can work only if primary aic3x instance is powered and reset is
released.
Solve this by implementing a list of probed instances that is used for
checking if other instance shares the same gpio_reset number. If a shared
reset line exists, then only first instance tries to request and configure
it and the last instance releases it.
Runtime modifications are not needed since aic3x_regulator_event with help
of regulator framework takes already care that reset is pulled down only
when some or all supplies are disabled meaning that all instances using them
are idle.
- ASoC: tlv320aic3x: Use gpio_is_valid in checking for valid gpio_reset
I promised to convert this at some point.
- ASoC: tlv320aic3x: Let the codec hit SND_SOC_BIAS_OFF when idle
Now codec hits the SND_SOC_BIAS_OFF also when it is idle. This is also
the default state after probing and codec is left unconfigured and
unpowered by default. Initialization will happen when the bias state changes
and aic3x_set_power does power-up and cache sync.
- ASoC: tlv320aic3x: Use regulator notifiers for optimizing the cache sync
There is no need to reset the codec and perform cache sync if none of the
supply regulators were not disabled. Patch registers a notifier callback for
each supply and callback then sets a flag to indicate when cache sync is
required.
HW writes are also needless when codec bias is off so cache_only flag is set
independently of actual supply regulators state.
- ASoC: tlv320aic3x: Add runtime regulator control to aic3x_set_bias_level
Now all the regulators are disabled when entering into SND_SOC_BIAS_OFF
and enabled when coming back to SND_SOC_BIAS_STANDBY state. Currently this
runtime control happens only with suspend/resume as this patch does not
change the default idle behavior.
This patch manages all the regulators and reset since it seems that register
sync is needed even if only analog supplies AVDD and DRVDD are disabled.
This was noted when the system was running with idle behavior changed and
IOVDD and DVDD were on.
It is not known are all the registers needed to sync or only some subset of
them. Therefore patch plays safe and does always full shutdown/power-up.
- ASoC: tlv320aic3x: Move regulator management from i2c to soc domain
It will be easier to keep regulator enable/disable calls in sync when dynamic
regulator management is added if regulator management is moved from
aic3x_i2c_probe/_remove to aic3x_probe/_remove.
- ASoC: tlv320aic3x: Add virtual output pin Detection
Purpose of this virtual Detection pin is to keep codec bias on whenever the
GPIO or jack detection features are needed.
Jack detection needs a mic bias so machine drivers can construct a following
route for instance for keeping the path and codec bias on:
"Input Jack" -> "Mic Bias xV" -> "Detection" -> detection block inside codec.
For the GPIO the machine driver can force the pin on with
snd_soc_dapm_force_enable_pin.
- ASoC: tlv320aic3x: Complete the soc-cache conversion
Complete the phasing out of aic3x_read_reg_cache, aic3x_write_reg_cache,
aic3x_read and aic3x_write calls.
This patch uses in aic3x_read the codec->hw_read that points to a function
implemented by soc-cache. Only use for aic3x_read is if wanting to read
volatile bits from those registers that has both read-only and read/write
bits. All other cases should use snd_soc_read.
- ASoC: tlv320aic3x: Switch to soc-cache helpers
Continue phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read
and aic3x_write calls.
This patch takes the soc-cache in use and removes aic3x_read_reg_cache and
aic3x_write.
- ASoC: tlv320aic3x: Use snd_soc_read and snd_soc_write
Start phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read and
aic3x_write calls in order to switch to soc-cache helpers.
This patch replaces aic3x_read_reg_cache and aic3x_write with snd_soc_read
and snd_soc_write. This is basically null-op since .read and .write in
soc_codec_dev_aic3x points to aic3x_read_reg_cache and aic3x_write.
- ASoC: tlv320aic3x: Optimize PLL programming in aic3x_set_bias_level
There is only need to enable/disable once the PLL when the bias is going
between on, prepare, standby and off states.
- ASoC: tlv320aic3x: Fix null pointer dereference when pdata is not set
Null pointer dereference will occur from *setup = pdata->setup if pdata
is not set. Fix this by moving assignments from pdata inside non-null case.
Thanks to Jiri Slaby <jirislaby@gmail.com> for noticing.
- ASoC: tlv320aic3x: Sanitize output controls
Currently output controls are not uniform. Some routes are adjusted by
mono controls that don't match to associated mixer switch, many routes are
not covered at all and stereo controls have following variants:
- L-to-L & R-to-R
- R-to-L & R-to-R
- L-to-L & R-to-L
This patch attempts to fix these issues. First, for the convenience, only
direct L-to-L, R-to-R and [L | R]-to-Mono routes are controlled by the
stereo controls. This logic is also used with the output pin mute controls
so all of them except mono output are controlled by stereo switches.
Then rest of the swapped L-to-R and R-to-L routes are controlled by the
mono controls that map to mixer switches with a same name. Mixers can then
associate these switches and volumes together.
- ASoC: tlv320aic3x: Reimplement output mixers
It turned out that the output mixers and their routes were misdefined: They
are not mixing output pins to internal signals but opposite. This has worked
for direct left-to-left and right-to-right routes since for those there are
complete routes. For swapped left-to-right and right-to-left routes this is
not working since there are no routes defined between them.
Another consequence is that those misdefined mixers are incorrectly routed
to several output pins leading unnecessary pin powerings even if there is no
route active to them.
Fix these by reimplementing the output mixers and routes as they are in
hardware. For completeness add also a few missing links between internal
signals and outputs.
- ASoC: tlv320aic3x: Sort output pin control registers in header file
Each output pin has 7 consecutive control registers in tlv320aic3x register
map. First 6 of them control the signal mixing and one is for output level
and power control.
Sort these registers as they are sorted clearly in hardware, it makes also
definitions more readable and easier to pinpoint missing register
definitions.
- ASoC: tlv320aic3x: Fix remaining output pin switch names
Bit 3 in output pin_CTRL register mutes the whole output pin not just the
route from DAC so remove misleading DAC from control name. Currently only
"Line[L | R] Playback Switch" were correct.
- ASoC: Fix tlv320aic3x GPIO initialization
aic3x_init does a soft reset first and thus TLV320AIC3x GPIO setup must be
done after doing the basic init. Before multi-component the init was done
at i2c probe time and GPIO setup at soc probe time.
- ASoC: Add support for tlv320aic3007 to tlv320aic3x codec.
This patch adds support for the tlv320aic3007 codec to the tlv320aic3x
driver.
The tlv320aic3007 is similar to the aic31, but has an additional class-D
speaker amp. The speaker amp control register overlaps with the mono
output register of other codecs in this family, so we add logic to
identify the actual codec being registered to set things up accordingly.
- ASoC: Configure symmetric rates for tlv320aic3x
The tlv320aic3x codec driver only supports symmetric rates for capture/
playback. Set the flag in the DAI accordingly.
- ASoC: tlv320aic3x: Add platform data and reset gpio handling
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
- ASoC: tlv320aic3x: Add basic regulator support
This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.
- ASoC: tlv320aic3x: Change bias management semantics
Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.
- ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_level
These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.
- ASoC: tlv320aic3x: Remove unused version string

SoC Codec TLV320DAC33

- ASoC: tlv320dac33: Add DAPM selection for LOM invert
The L/R LOM line can be invertined side of the
corresponding DAC, or inverted from the corresponding
LOP.
Add control for user space to select the source of the
LOM inversion.
When only the analog bypass is enabled, and the LOM
is inverted from DAC output, we need to power the
corresponding DAC.
- ASoC: tlv320dac33: Add 32/24 bit audio support
Add support for 24 bit audio (with S32_LE msbits 24).
The reason to limit the msbits to 24, is that the FIFO
can be configured for 16 or 24 bit layout.
It is unknown how the codec would downsample from 32 to
24 bit, if the interface is configured to receive 32
bit data.
- ASoC: tlv320dac33: Some cleanup for 32/24 bit support
Change the structure of FIFO handling in order to
pave the way for adding 32/24 bit audio support.
- ASoC: tlv320dac33: Remove manual FIFO configuration
The manual FIFO configuration was the first version to enable
the use of the FIFO in the codec.
It had served it's purpose as debugging aid, but the automatic
FIFO configuration is much safer to use.
The removal of the manual controls, and configuration makes
it easier to add new features for the codec later, since
the manual mode neded different ways to calculate, and
protect against misconfiguration.
- ASoC: tlv320dac33: Power down digital parts, when not needed
If the following scenario has been followed:
1. Enable analog bypass
amixer sset 'Analog Left Bypass' on
amixer sset 'Analog Right Bypass' on
2. Start playback
aplay -fdat -d3 /dev/zero
After the playback stopped (3 sec), and the soc timeout (5 sec),
the digital parts of the codec will remain powered up.
This means that the DAI clocks are continue to run, the
oscillator remain operational, etc.
Use the SND_SOC_DAPM_POST_PMD widget to get notification
about the stopped stream, and power down the digital
part of the codec.
If the analog bypass is enabled, than the codec will remain in
BIAS_ON level, and things will work correctly.
In case, if the bypass is disabled, than the codec will
fall to BIAS_STANDBY than to BIAS_OFF level, as it used
to.
The digital part of DAC33 is initialized at every stream start
(DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec)
will have working DAI.
When the codec is coming out from BIAS_OFF, the full power-up
sequence followed by the same DAPM_PRE widget event will power up
the digital part.
- ASoC: tlv320dac33: Fix compillation error
Fix the compilation error introduced by patch:
ASoC: tlv320dac33: Avoid multiple soft power up
- ASoC: tlv320dac33: Move DAC LR power on to a supply widget
The power for the DACs need to be enabled, even when only
the analog bypass is in use with the codec, otherwise
the audio is going to be distorted.
Make sure that the DACs are powered all the time, when
there is audio activity.
- ASoC: tlv320dac33: Rename outpup amplifier widget
Use better name for the widget, and remove the 'Power'
from it's name.
- ASoC: tlv320dac33: Avoid multiple soft power up
During playback start the codec has been already powered at
BIAS_ON event time, so there's no need to enable the codec again.
- ASoC: tlv320dac33: Do not enable the codec in init_chip
No need to enable the codec at this time.
The codec will be enabled later by other events
- ASoC: tlv320dac33: Mode1 FIFO auto configuration fix
Do not allow invalid (too big) nSample value, when FIFO Mode1
and automatic fifo configuration has been selected.
- ASoC: tlv320dac33: Limit the US_TO_SAMPLES macro
Limit the time window to maximum 1s in the macro.
The driver deals with much shorter times (<200ms).
This will fix a rare division by zero bug in Mode1.
This could happen, when the work is not executed in
time (within mode1_latency) after the interrupt.
In this case the DAC33 will not receive the needed
nSample command in time, and enters to an unknown
state, and won't recover.
In such event the time window will increase, and
eventually going to be bigger than 1s, resulting
devision by zero.
- ASoC: tlv320dac33: Error handling for broken chip
Correct/Implement handling of broken chip.
Fail the soc_prope if the communication with the chip
fails (can not read chip ID).
- ASoC: tlv320dac33: Use usleep_range for delays
Switch to use the more precise usleep_range instead of
msleep().
Replace the udelay with usleep_range to remove the busy loop
waiting.
- ASoC: tlv320dac33: Control for line output gain
New control to select the line output gain.
This gain control affects the linein-to-lineout and
dac-to-loneout gain differently.
Use enum type to select the desired gain combination.
- ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
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.
- ASoC: tlv320dac33: Add support for automatic FIFO configuration
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
- ASoC: tlv320dac33: Revisit the FIFO Mode1 handling
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
- ASoC: tlv320dac33: Add support for changing upper threshold
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
- ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
- ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
- ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power
Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control
It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.
The codec reset values are considered safe in all environmnts.
- ASoC: tlv320dac33: Support for turning off the codec
Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).
After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).
The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.
There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, but we does not
need to execute the playback related configuration
2. Playback caused BIAS_OFF -> BIAS_ON
We need to power on the codec, and do the chip init, and also we need
to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
We need to execute the playback related configuration. The codec is
still on.
Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.
Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.
- ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure
As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.
The substream must be easily available in other places than
pcm_* callbacks.
Manage a pointer in _startup, and _shutdown for this.
- ASoC: tlv320dac33: Revised module loading, and DAC33 ID read
Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.
- ASoC: tlv320dac33: Optimize power up, and restore
On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.
- ASoC: tlv320dac33 - disable regulators at i2c remove()
- ASoC: tlv320dac33: FIFO caused delay reporting
Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.
Bypass mode: FIFO is bypassed, report 0 as delay
Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.
Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.
Phase1: when we received the alarm threshold, but our workqueue has
not been executed (safeguard phase). Just count the played out
samples since ts1 and subtract it from the alarm threshold
value.
Phase2: During nSample burst (after writing to nSample register), count
the played out samples since ts1, count the samples received
since ts2 (in a burst). Estimate the FIFO depth using these and
alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
samples since ts1. Estimate the FIFO depth using the nSample
configuration and the alarm threshold value.
Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received
Interrupts are coming when DAC33 FIFO depth reaches upper threshold.
Phase1: Draining phase (after the burst), counting the played out
samples since ts1, and subtract it from the upper threshold
value.
Phase2: During burst operation. Using the pre calculated time needed to
play out samples from the buffer during the drain period (from
upper to lower threshold), move the time window to cover the
estimated time from the burst start to the current time.
Calculate the samples played out since lower threshold and also
the samples received during the same time.
- ASoC: tlv320dac33: Calculate the interface speed during bursts
When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.
- ASoC: tlv320dac33: Change magic numbers used in Mode7
Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.
- ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode
There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.
- ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1
Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.

SoC Codec TPA6130A2

- ASoC: tpa6130a2: Fix compiler warning
sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls':
sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm'
Introduced by commit 39646871a47fd8808c08de0ce7d7ca8393af2805 ("ASoC:
tpa6130a2: Replace DAPM code with direct interface").
The DAPM code has been removed from the driver, but the
dapm struct remained.
- ASoC: tpa6130a2: Replace DAPM code with direct interface
The use of DAPM widgets, and extra routing can cause ordering
problems in the system.
Machine drivers should use the exported direct interface with
SND_SOC_DAPM_HP's event callback to manage the state of the
amplifier.
- ASoC: tpa6130a2: Add stereo DAPM path
New DAPM widgets, and paths to enable both channels at the
same time (for stereo output).
With this path the switch time difference can be avoided
between left and right channels.
The original DAPM paths can be still used, if only one of
TPA's output has been connected to a speaker, but for most of
the cases, switching to the stereo path is better.
- ASoC: tpa6130a2: Use one event handler for PGA_E
Reduce the amount of duplicated code by using single
event handler for PGA_E to enable the needed channel.
Use the w->shift to pass the channel information to
the handler function.
- ASoC: tpa6130a2: Defer SW enable from power enable
Do not enable the amplifier right after the power has been
restored to the amplifier.
The DAPM_SUPPLY widget turns on the amp early in the DAPM
power walk, and the unmuting of channel happens quite late.
Keeping the amp in SW reset state ensures better muting.
In this way the pop noise coming from other components (codec)
can be filtered out.
- ASoC: tpa6130a2: Simplify power state management
Use simpler way to avoid setting the same power state
for the amplifier.
Simplifies the check introduced by patch:
ASoC: tpa6130a2: Fix unbalanced regulator disables
- ASoC: Do not update the cache if write to hardware failed
- ASoC: tpa6130a2: Revisit power-up sequence
There are no known problems with current power-up sequence which first sets
the /shutdown pin high and then enables the supply. However, swap the order
so that the device is kept in shutdown/reset mode during the supply voltage
transition since slowly rising voltages can usually cause problems if the
device is not kept in reset.
- ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power
Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a
compiler warning "‘ret’ may be used uninitialized in this function".
Initialize ret to zero to get rid of it and making sure that the function
does not return any random error code when the code is falling through.
- ASoC: tpa6130a2: Fix unbalanced regulator disables
This driver has unbalanced regulator_disable when doing module loading and
unloading. This is because tpa6130a2_probe followed by tpa6130a2_remove
calls twice tpa6130a2_power(0). Fix this by implementing a state checking
in tpa6130a2_power.
- ASoC: tpa6130a2: Error handling for broken chip
Correct/Implement handling of broken chip.
Fail the i2c_prope if the communication with the chip
fails.
- ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.
Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.
Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
- ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.
- ASoC: tpa6130a2: Fix for the custom kcontrol functions
Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.
- sound: Revert "ASoC: tpa6130a2: Support for limiting gain"
This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
- ASoC: tpa6130a2: Support for limiting gain
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
- ASoC: tpa6130a2: TLV mapping for tpa6140a2
Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.
Provide different mixer control for the chips with correct
TLV mapping.
User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140
The way machine drivers are using this amplifier remained
the same.

SoC Codec TWL4030

- ASoC: TWL4030: Fix 24bit support
twl4030 series of codecs supports S32_LE with msbits=24.
Replace the S24_LE with S32_LE format, and add constraint
for 24msbit in case of 32 S32_LE format.
- ASoC: Move kfree(twl4030) to twl4030_soc_remove()
As we allocate memory for twl4030 in twl4030_codec_probe(),
twl4030_codec_remove() is a better place to free the memory.
- ASoC: TWL4030: Use usleep_range when appropriate
Change the busy loop delays with usleep_range or msleep calls.
- ASoC: multi-component: TWL4030: Restore registers on removal
Add back the register restore call, when the codec driver is
removed.
This does not affect normal operation, but it is usefull when
debugging audio through the twl4030 class codecs.
- ASoC: TWL4030: Capture route runtime DAPM ordering fix
Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
- ASoC: TWL4030: Capture route DAPM event fix
There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.
- ASoC: TWL4030: Add configurable delay after digimic enable
When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.
Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.
Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.
Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).
- ASoC: TWL4030: DAC power optimization
Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.
- ASoC: TWL4030: Fix for digital loopback gain range
When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.
- ASoC: TWL4030: Add functionalty to reset the registers
Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.
If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.
If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.
As safety measure do the register reset at remove time also.
- ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use
Restructure the codec power code in order to be able to hit
off when the codec is not in use.
Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.
- ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default
It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.
- ASoC: TWL4030: Helper to check chip default registers
Since the twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.
- ASoC: TWL4030: Optimize the power up sequence
Since the reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.
According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.
Move all chip initialization code under chip_init, and do
it when the codec is initialized.
- ASoC: TWL4030: Make offset cancellation path configurable
Add means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.
Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*
- ASoC: TWL4030: Remove wrapper for power down
There is no need for the power down wrapper.
- ASoC: TWL4030: Revisit codec defaults
Reset most of the codec registers to their chip reset
value.
- ASoC: TWL4030: Add control for digimic Left Right swap
The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
- ASoC: TWL4030: Remove OUTL/R outputs
OUTL/R are leftovers from the original driver, and they
are no longer needed.
- ASoC: TWL4030: AIF/APLL fix in DAPM domain
This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.
We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */
/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */
/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */
The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).
The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.
- ASoC: missing conversions to snd_soc_codec_*_drvdata()
Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.

SoC Codec TWL6040

- ASoC: twl6040: Convert HF and HS drivers to use DAPM OUT_DRV widget
Make the phoenix HS and HF drivers use the new DAPM driver
widget in order to guarantee power ON/OFF order sequence.
- ASoC: twl6040: Add ramp up/down volume for HS and HF
Add ramp functions for the headset and handsfree outputs
in order to reduce the pops during power on/off sequences.
In order to give more control to volume ramp, step size and delay
between steps can be specified.
The patches are based on wm8350 implementation from Liam
Girdwood.
- ASoC: twl6040: Set default gains to minimun value
Updated default values to improve power consumption.
- ASoC: twl6040: Use correct offset for LineInAmp Right
Gain for LineInAmp Right uses LINEGAIN[5:3], which means that
offset for right channel should be 4.
- ASoC: twl6040: Fix TLV dB step values for gains
Some gains were incorrectly configured for dB values.
- ASoC: twl6040: Increase timeout for power up
After coming back from suspend, the timeout waiting for Phoenix
chip to complete its power up sequence is not enough, which leaves
the codec cache value for some registers in an outdated state.
Increase the timeout value to wait for the power up sequence
to correclty complete.
- ASoC: twl6040: Enable plug detection interrupts
Enable plug detection interrupt mask in order to get headset
PLUGINT/UNPLUGINT interrupts.
- ASoC: twl6040: Clear interrupt status at boot time
On Phoenix 1.1, the INTID register default value is an invalid
one, causing the interrupt handler to think the phoenix power on
sequence is ready before it actually finishes.
This causes some i2c errors when trying to configure twl.
- ASoC: twl6040: Enable automatic power for phoenix 1.1
Phoenix 1.1 supports automatic power on sequence, a
verification is added to use it with new revision of
the chip.
- ASoC: twl6040: Fix analog Mic L & R mux controls
The mux control has 4 elements not 3
- ASoC: twl6040: Support other sample rates.
The twl6040 can support more sample rates other than 88.2 and 96k.
- ASoC: twl6040: Fix PCM error handling ops
This patch moves all the PCM error handling for clock config
out of trigger() and startup() and into prepare().
- ASoC: twl6040: Restore bias level at resume
This patch restores the CODEC bias level at resume().
- ASoC: twl6040: Add headset and handset mux controls
This patch adds support for the twl6040 headset and handset
MUX controls.
- ASoC: twl6040: Modify the IRQ handler
Multiples interrupts can be received. The irq handler is modified
to attend all of them.
- ASoC: twl6040: Update twl IO macro
Update the codec to use the new twl core register macros
- ASoC: twl6040: Add jack support for headset and handset
This patch adds support for reporting twl6040 headset and
handset jack events.
The machine driver retrieves and report the status through
twl6040_hs_jack_detect.
A workq is used to debounce of the irq.
- ASoC: TWL6040: Correct widget handling for drivers
In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.
Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.
- ASoC: TWL6040: Fix playback with 19.2 Mhz MCLK
When using MCLK is configured for 19.2 Mhz, clock slicer should be
enabled and HPPLL should be bypassed in clock path.
- ASoC: TWL6040: Enable earphone path in codec
Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.
- ASoC: missing conversions to snd_soc_codec_*_drvdata()
Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.

SoC Codec WL1273

- ASoC: WL1273 FM radio: Fix breakage with MFD API changes
These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.
Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.
- ASoC: Do not include soc-dai.h
There is no need to include soc-dai.h since soc.h includes it. Convert
drivers to include only soc.h.
- ASoC: WL1273 FM Radio: Eliminate unnecessary error return value.
With this change it's not a error to call wl1273_set_audio_route
when the codec is active if the new routing value is the same
as the current active setting.
- ASoC: TI WL1273 FM Radio Codec.
This is an ALSA codec for the Texas Instruments WL1273 FM Radio.

SoC Codec WM2000

- ASoC: Default WM2000 ANC and speaker to enabled
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
- ASoC: wm2000: remove unused #include <linux/version.h>
Remove unused #include <linux/version.h>('s) in
sound/soc/codecs/wm2000.c

SoC Codec WM8350

- ASoC: Use delayed work to debounce WM8350 jack IRQs
This avoids blocking the IRQ thread and allows further bounces to extend
the debounce time.
- ASoC: Yet more x86 tracepoint workarounds
- ASoC: ifdef out trace points from modules for x86
No idea why this works on ARM but not x86.
- ASoC: Trace Wolfson jack detection IRQs
Add jack detection interrupt trace to Wolfson CODEC drivers.
- ASoC: Ensure sane WM835x AIF configuration by default
Ensure that whatever ran before us leaves the WM835x with a sane default
audio interface configuration as we do not override the companding,
loopback or tristate settings and do not reset the chip at startup (as it
is a PMIC).
Reported-by: Keiji Mitsuhisa <Keiji.Mitsuhisa@wolfsonmicro.com>
- ASoC: Remove broken WM8350 direction constants
The WM8350 driver was using some custom constants to interpret the direction
of the MCLK signal which had the opposite values to those used as standard
by the ASoC core, causing confusion in machine drivers such as the 1133-EV1
board.
Reported-by: Tommy Zhu <Tommy.Zhu@wolfsonmicro.com>
- ASoC: Fix dB scales for WM835x
These should be regular rather than linear scales.

SoC Codec WM8400

- ASoC: Fix dB scales for WM8400
These scales should be regular, not linear.

SoC Codec WM8510

- ASoC: Remove unneeded use of spi_bus_type
No need to explicitly set the bus type, spi_register_driver does
that for us.
- ASoC: Remove unneeded control_data management from Wolfson drivers
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
- ASoC: Remove version display from WM8510 driver

SoC Codec WM8523

- ASoC: Avoid direct register cache access when setting defaults
Directly accessing the register cache means that we can't use anything
except a flat register cache so use snd_soc_update_bits().
- ASoC: Update users of readable_register()/volatile_register()
- ASoC: codecs: wm8523: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8523 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8523 driver to use the
generic register cache in its private functions.
- ASoC: Fix multi-component mismerge in WM8523
- ASoC: Remove unneeded control_data management from Wolfson drivers
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
- ASoC: wm8523: fix resource reclaim in wm8523_register error path
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
handling if snd_soc_register_dai failed.

SoC Codec WM8580

- ASoC: WM8580: Debug interface index
We want the index of DAI's driver here.
- ASoC: WM8580: Fix R8 initial value
Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c.
- ASoC: WM8580: Debug BCLK and sample size
In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.
- ASoC: Remove unneeded use of address-of operator
There is no need to use '&' in this case. Either way, if a is an array
of some type, then a == &a == &a[0].
- ASoC: WM8580: Remove useless assignment
The variable is not used anyway.
- ASoC: Fix incorrect register cache size configuration
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.
- ASoC: Fix argument ordering for snd_soc_update_bits() in WM8580
Reported-by: Seungwhan Youn <claude.youn@gmail.com>
- ASoC: Fix WM8580 CLKSEL mask selection
The RX and TX directions were inverted.
Reported-by: Seungwhan Youn <claude.youn@gmail.com>
- ASoC: Automatically manage WM8580 DAC OSR
The DAC OSR should be selected based on the sample clock ratio.
- ASoC: Fix inverted WM8580 capture mute control
- ASoC: Implement BCLK rate selection for WM8580
Drive a minimal supported number of clocks required for the current
bit format in master mode. In slave mode this setting has no effect.
- ASoC: Automatically calculate clock ratio for WM8580
Implement set_sysclk() and then rather than assuming 256fs use the
supplied value to calculate and configure the clock ratio for the
currently used sample rate. As a side effect we also end up
implementing clock selection for the ADC path.
In order to avoid confusion remove the existing set_clkdiv() based
configuration of the clock source for the DAC and update the SMDK64xx
driver (which is the only in-tree user of the CODEC).
- ASoC: Remove unused rate selection bitmasks from WM8580
In the case of the BCLK rate the defines are at best misleading anyway.
- ASoC: Convert WM8580 hw_params to use snd_soc_update_bits()
All the cool kids are using snd_soc_update_bits() these days.
- ASoC: Remove unneeded control_data management from Wolfson drivers
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
- ASoC: Correct WM8580 Capture control names
Should use Capture rather than ADC so the UI tools can identify their
function more readily.
- ASoC: Fix inverted mute controls for WM8580

SoC Codec WM8711

- ASoC: Remove unneeded use of spi_bus_type
No need to explicitly set the bus type, spi_register_driver does
that for us.
- ASoC: Fix incorrect register cache size configuration
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.
- ASoC: Remove unneeded control_data management from Wolfson drivers
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
- ASoC: wm8711: fix a memory leak if another WM8711 is registered
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).

SoC Codec WM8727

- ASoC: Staticise WM8727 codec driver structure
Nothing should be referencing this any more.
- ASoC: wm8727: add a missing return in wm8727_platform_probe
otherwise the error path will always be executed.

SoC Codec WM8728

- ASoC: Remove unneeded use of spi_bus_type
No need to explicitly set the bus type, spi_register_driver does
that for us.
- ASoC: Fix incorrect register cache size configuration
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.

SoC Codec WM8731

- ASoC: Automatically manage WM8731 deemphasis
The deemphasis filter should be selected based on sample rate for
optimal performance.
- ASoC: Split WM8731 enumeration array into individual enums
This is much more maintainable than the array.
- ASoC: Provide WM8731 microphone boost TLV information
- ASoC: WM8731: Fix incorrect mask for bypass path disable
According to the datasheet the bypass path enable/disable is
bit 3 therefore we need 0x8 and not 0x4.
- ASoC: Fix incorrect kfree in wm8731_probe error path
We allocated memory for wm8731 in wm8731_spi_probe / wm8731_i2c_probe,
and will free the memory in either wm8731_spi_probe / wm8731_i2c_probe
error path or wm8731_spi_remove / wm8731_i2c_remove.
Thus we should not call kfree(wm8731) in wm8731_probe, otherwise
we have double free of wm8731.
- ASoC: Remove unneeded use of spi_bus_type
No need to explicitly set the bus type, spi_register_driver does
that for us.
- ASoC: Fix incorrect register cache size configuration
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.
- ASoC: Use more idiomatic driver name for WM8731
Make dev_() prints much prettier.
- ASoC: Refactor WM8731 regulator management into bias management
This allows more flexible integration with subsystem features.

SoC Codec WM8737

- ASoC: Remove redundant hw_write initialisation in WM8737
- ASoC: Add WM8737 ALC support
- ASoC: Add initial WM8737 driver
The WM8737 is a low power, flexible stereo ADC designed for portable
applications. This driver supports most of the functionality of the
WM8737, though some features such as the ALC are not yet implemented.

SoC Codec WM8741

- ASoC: Avoid direct register cache access when setting defaults
Directly accessing the register cache means that we can't use anything
except a flat register cache so use snd_soc_update_bits().
- ASoC: codecs: wm8741: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8741 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8741 driver to use the
generic register cache in its private functions.
- ASoC: Remove unneeded use of address-of operator
There is no need to use '&' in this case. Either way, if a is an array
of some type, then a == &a == &a[0].
- ASoC: Complete supported clock ratios and rate constraints for wm8741
- ASoC: Make codec dai naming for WM8741 consistent
- ASoC: Initial WM8741 CODEC driver
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz. The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.
TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.

SoC Codec WM8750

- ASoC: codec: Add WM8987 device id to WM8750 driver
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.

SoC Codec WM8753

- ASoC: codecs: wm8753: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8753 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Furthermore the generic cache uses zero-based numbering while the wm8753 cache
uses one-based numbering.
Thus we end up with two from each other incoherent caches, which leads to undefined
behaviour and crashes.
This patch fixes the issue by changing the wm8753 driver to use the generic
register cache in its private functions.
- ASoC: wm8753, remove dead code
There is adangling code in wm8753_probe which is never executed.
Remove it.

SoC Codec WM8770

- ASoC: WM8770: Initial driver
The WM8770 is a high performance, multi-channel audio
codec. The WM8770 is ideal for surround sound processing
applications for home hi-fi, automotive and other audio
visual equipment.

SoC Codec WM8776

- ASoC: WM8776: Removed unneeded struct member
The member reg_cache is not used at all and therefore it should be
removed. This member was usually needed for older versions of ASoC
that did not handle caching automatically and had to be done in the
driver itself.
- ASoC: Remove DSP mode support for WM8776
This is not supported by current hardware revisions.
- ASoC: Remove duplicate AUX definition from WM8776

SoC Codec WM8804

- ASoC: checking kzalloc() for IS_ERR() instead of NULL
There is a typo here that got copy and pasted to several probe
functions. kzalloc() returns NULL on allocation failures and not an
ERR_PTR.
- ASoC: WM8804: Retrieve the device revision and print it
Be verbose and print out the device revision.
- ASoC: WM8804: Power down the PLL correctly
The PLL is disabled when the corresponding bit is set not the other
way around. This commit depends on my other commit with Subject
"ASoC: WM8804: Refactor set_pll code to avoid GCC warnings".
- ASoC: WM8804: Refactor set_pll code to avoid GCC warnings
Ensure that no uninitialised variable warnings are generated by
GCC.
- ASoC: WM8804: Use a proper DAI name
Make sure the DAI name does not include a '/' since we might have
per DAI debugfs or sysfs entries in the future.
- ASoC: WM8804: Initial driver
The WM8804 is a high performance consumer mode S/PDIF transceiver with
support for 1 received channel and 1 transmitted channel.

SoC Codec WM8900

- ASoC: Remove volatility from WM8900 POWER1 register
Not all bits can be read back from POWER1 so avoid corruption when using
a read/modify/write cycle by marking it non-volatile - the only thing we
read back from it is the chip revision which has diagnostic value only.
We can re-add later but that's a more invasive change than is suitable
for a bugfix.

SoC Codec WM8903

- ASoC: Fix double comment start
- ASoC: Remove incorrect WM8903 erratum workaround
Due to a typographical error in the erratum workaround it was never
functional so just remove it.
- ASoC: ifdef out trace points from modules for x86
No idea why this works on ARM but not x86.
- ASoC: Trace Wolfson jack detection IRQs
Add jack detection interrupt trace to Wolfson CODEC drivers.
- ASoC: Remove some unused defines from WM8903
These would have been used if we'd done manual clock divider setup,
but we didn't.
- ASoC: Automatically manage WM8903 deemphasis rate
Provide the user with a boolean control then automatically select
the deemphasis filter most closely matching the sample rate.
- ASoC: Remove open coded symmetry implementation from WM8903
We're already flagged as using symmetric rates so we don't need to
have a custom implementation.
- ASoC: Implement WM8903 oversampling rate controls
- ASoC: Implement WM8903 high pass filter support

SoC Codec WM8904

- ASoC: Avoid direct register cache access when setting defaults
Directly accessing the register cache means that we can't use anything
except a flat register cache so use snd_soc_update_bits().
- ASoC: codecs: wm8904: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8904 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8904 driver to use the
generic register cache in its private functions.
- ASoC: fix deemphasis control in wm8904/55/60 codecs
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
- ASoC: wm8904 - fix memory leaks
- ASoC: Remove unnecessary semicolons
- ASoC: wm8904: fix resource reclaim in wm8904_register error path
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
snd_soc_register_dai() fail to properly free resources.

SoC Codec WM8940

- ASoC: codecs: Add missing control_type initialization
Some codec drivers do not initialize the control_type field in their private
device struct, but still use it when calling snd_soc_codec_set_cache_io.
This patch fixes the issue by properly initializing it in the drivers probe
functions.
- ASoC: Remove unnecessary semicolons
- ASoC: wm8940: fix a memory leak if wm8940_register return error
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.

SoC Codec WM8955

- ASoC: Avoid direct register cache access when setting defaults
Directly accessing the register cache means that we can't use anything
except a flat register cache so use snd_soc_update_bits().
- ASoC: codecs: wm8955: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8955 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8955 driver to use the
generic register cache in its private functions.
- ASoC: codecs: Add missing control_type initialization
Some codec drivers do not initialize the control_type field in their private
device struct, but still use it when calling snd_soc_codec_set_cache_io.
This patch fixes the issue by properly initializing it in the drivers probe
functions.
- ASoC: fix deemphasis control in wm8904/55/60 codecs
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
- ASoC: wm8955: fix resource reclaim in wm8955_register error path
This patch fixes the error path in wm8955_register to properly free resources.

SoC Codec WM8960

- ASoC: Move widgets from DAPM context to snd_soc_card
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
- ASoC: codecs: Add missing control_type initialization
Some codec drivers do not initialize the control_type field in their private
device struct, but still use it when calling snd_soc_codec_set_cache_io.
This patch fixes the issue by properly initializing it in the drivers probe
functions.
- ASoC: fix deemphasis control in wm8904/55/60 codecs
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
- ASoC: Automatically manage ALC coefficients for WM8960
- ASoC: Automatically manage DAC deemphasis rate for WM8960
- ASoC: Remove current WM8960 deemphasis control
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.
- ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option
The SYSCLK source is automatically managed when configuring the PLL.

SoC Codec WM8961

- ASoC: wm8961 - clear WM8961_MCLKDIV bit for freq <= 16500000
MCLKDIV bit of Register 04h Clocking1:
0 : Divide by 1
1 : Divide by 2
Thus in the case of freq <= 16500000, we should clear MCLKDIV bit.
- ASoC: wm8961 - clear WM8961_DACSLOPE bit for normal mode
DACSLOPE bit of Register 06h ADC and DAC Control 2:
0: Normal mode
1: Sloping stop-band mode
Thus in the case of normal mode, we should clear DACSLOPE bit.
- ASoC: Remove unused driver data from WM8961 probe
- ASoC: wm8961: fix resource reclaim in wm8961_register error path
This patch fixes the error path in wm8961_register to properly free resources.

SoC Codec WM8962

- ASoC: One more x86 typo fix
- ASoC: Fix typo in x86 workaround
- ASoC: ifdef out trace points from modules for x86
No idea why this works on ARM but not x86.
- ASoC: Trace Wolfson jack detection IRQs
Add jack detection interrupt trace to Wolfson CODEC drivers.
- ASoC: codecs: wm8962: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm8962 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm8962 driver to use the
generic register cache in its private functions.
- ASoC: Correct WM8962 interrupt mask register read
Fix mismerge from the out of tree BSP where this support was developed.
- ASoC: Fix incorrect kfree in wm8962_probe error path
We allocated memory for wm8962 in wm8962_i2c_probe,
and will free the memory in either wm8962_i2c_probe error path
or wm8962_i2c_remove.
Thus we should not call kfree(wm8962) in wm8962_probe, otherwise
we have double free of wm8962.
- ASoC: Reset WM8962 with default ID value
The value makes no odds and it makes life easier with caches.
- ASoC: Use pm_wakeup_event() in WM8962 jack detection
Ensure that the system does not suspend while we process a WM8962 jack
event by using pm_wakeup_event() to block the suspend while we're waiting
for the jack to settle. Use a slightly longer timeout than the jack waits
to allow for other stuff to take over and delays in scheduling.
- ASoC: Check return value of strict_strtoul() in WM8962
strict_strtoul() has been made __must_check so do so.
- ASoC: Add WM8962 microphone bias control
We unconditionally require SYSCLK since while only microphone detection
specifically requires SYSCLK any actual use case would enable it via
some other means but microphone detection may have nothing active other
than the bias itself.
- ASoC: Compress WM8962 register access map
Reduce the source code size still futher by only specifying non-zero
rows in the WM8962 access map.
- ASoC: Shrink code size for WM8962 register defaults table
Dramatically reduce the code size for the WM8962 register defaults table
by switching to explicitly initialise only defined registers, relying on
static defaulting to zero for the overwelming bulk of the register map.
Similar treatement for the register access table will come later and will
produce a similarly dramatic code size shrink.
- ASoC: Put WM8962 GPIO2/3 into GPIO mode when configuring platform data too
GPIO2 and GPIO3 on the WM8962 are MFPs and need to be put into GPIO mode
before the GPIO block can be used to control them. We're already doing
this when used via gpiolib, factor out the code for use when setting static
configurations via platform data as well.
- ASoC: Implement interrupt based jack detection
Allow microphone detection on WM8962 to be performed using the interrupt
signal, allowing the detection of both microphone presence and button
presses with a signal singal from the CODEC to CPU. Currently a 250ms
debounce time is applied to both short circuit and presence detection,
this has not been optimised.
- ASoC: Add support for WM8962 GPIO outputs
The WM8962 features five GPIOs, add support for controlling their output
state via gpiolib.
- ASoC: Relax restrictions on WM8962 BCLK configuration
The restrictions on configuring BCLK are overly cautious, other constraints
in the system should ensure that reconfiguration is not possible when the
device is sufficiently active to be unable to support reclocking.
- ASoC: Provide microphone bias configuration for WM8962
Add the widget for MICBIAS power control and allow configuration of the
microphone bias setup via the platform data for the WM8962. When
microphone status signals are brought out to GPIO this should be
sufficient to enable microphone detection.
- ASoC: Mark WM8962 Additional Control 4 register as volatile
There are some status bits for microphone detection in here.
- ASoC: Initial WM8962 IRQ support
Provide an initial hookup for interrupts on the WM8962. Currently we simply
report error status via log messages if an IRQ is provided for the device.
- ASoC: Lowercase WM8962 CODEC name
For consistency.
- ASoC: Fix masking of WM8962 FLL1 register
When configuring the FLL we preserve the FLL enable configuration in order
to allow us to reenable the FLL after configuration but we do not clear
the other bits in the register, causing old configuration to be preserved.
- ASoC: Fix merge issue with WM8962 control addition
Let's not add the core controls twice.
- ASoC: Fix off-by-one bug in WM8962 register cache size configuration
This is a simple off-by-one bug, the size of the register cache is
incorrectly set to the maximum register index. Fix it by adding one.
- ASoC: Update WM8962 to build with multi-component
No notable changes, currently build tested only.
- ASoC: Add initial WM8962 CODEC driver
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.

SoC Codec WM8971

- ASoC: codecs: Add missing control_type initialization
Some codec drivers do not initialize the control_type field in their private
device struct, but still use it when calling snd_soc_codec_set_cache_io.
This patch fixes the issue by properly initializing it in the drivers probe
functions.
- ASoC: Remove version display from WM8971 driver

SoC Codec WM8974

- ASoC: Remove unused WM8974 private data
- ASoC: wm8974: fix a memory leak if another WM8974 is registered
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).
- ASoC: Correct inversion of speaker mixer PCM switch
Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>

SoC Codec WM8978

- ASoC: Fix WM8978/migor driver name conflict
Standardise on 'wm8978' as the name for the CODEC.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- ASoC: wm8978: fix a memory leak if a wm8978_register fail
There is a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>

SoC Codec WM8985

- ASoC: checking kzalloc() for IS_ERR() instead of NULL
There is a typo here that got copy and pasted to several probe
functions. kzalloc() returns NULL on allocation failures and not an
ERR_PTR.
- ASoC: WM8985: Use the correct macro for the kcontrol defined
Since we are using custom get/put handlers
use SOC_ENUM_SINGLE_EXT_DECL instead of the original SOC_ENUM_SINGLE_DECL
macro.
- ASoC: Fix a compile warning for printk format
sound/soc/codecs/wm8985.c: In function 'wm8985_hw_params':
sound/soc/codecs/wm8985.c:731:2: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
Actually the variable is fine as int.
- ASoC: WM8985: Remove unneeded declaration.
We are not using the private data in this function, so get rid of it.
- ASoC: WM8985: Initial driver
The WM8985 is a low power, high quality, feature-rich stereo
CODEC designed for portable multimedia applications that
require low power consumption and high quality audio.

SoC Codec WM8988

- ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
snd_soc_unregister_codec is called twice if snd_soc_register_dai fail.

SoC Codec WM8990

- ASoC: WM8990: msleep() takes milliseconds not jiffies
- ASoC: Remove version display from WM8990
It's not needed and the version number never gets updated anyway.
- ASoC: Fix dB scales for WM8990
These should be regular, not linear.

SoC Codec WM8991

- ASoC: WM8991: Add initial WM8991 driver
The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for
handsets rich in multimedia features such as GPS, mobile TV, digital audio
playback and gaming.
This driver was originally written by Graeme Gregory and has been maintained
out of tree by Mark Brown and Dimitris Papastamos.

SoC Codec WM8993/4

- ASoC: Handle low measured DC offsets for wm_hubs devices
The DC servo codes are actually signed numbers so need to be treated as
such.
- ASoC: Explicitly clear WM8993 ramp controls on power down
This helps ensure that the ramp logic is reset when powering back up.
- ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain
SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit.
- ASoC: Use DC servo startup mode when not doing DCS correction
Devices which do not have a DCS correction applied can use the explicit
DC servo startup mode for optimal startup performance. This most
immediately affects the WM8958.
- ASoC: Remove unnecessary semicolons
- ASoC: Store DC offset correction for wm_hubs devices in class W mode
Providing the analogue configuration of the output path remains the same
the DC offset corrected by the DC servo will remain identical so we can
skip the callibration, reducing the startup time for the headphone output.
Implement this for the wm_hubs devices as has been done for several other
CODECs.
Don't do this if we have any analogue paths enabled since offsets may be
being introduced by the analogue paths which could vary outside the
control of the driver.
- ASoC: Update WARN uses in wm_hubs
Add missing newlines.
- ASoC: Explicitly disable DC servo on WM hubs headphone powerdown
- ASoC: Reorder power down sequence for WM hubs devices
Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.
- ASoC: Add additional WM hubs DC servo trace
Log the values we're getting back from the DC servo and the values we
write to it.

SoC Codec WM8994

- ASoC: WM8994: fix wrong value in tristate function
fix wrong value in wm8994_set_tristate func. when updating reg bits,
it should use "value", not "reg".
- ASoC: Allow user-specified WM8958 multiband compressor configurations
The paramters of the WM8958 multiband compressor can be tuned by the
user for their system using a graphical configuration tool on the host.
Allow the user to specify a set of such paramters in platform data and
select between them at runtime.
- ASoC: Yet more x86 tracepoint workarounds
- ASoC: Trace Wolfson jack detection IRQs
Add jack detection interrupt trace to Wolfson CODEC drivers.
- ASoC: Fix WM8994/58 3D stereo control definitions
Cut'n'paste in the register names.
- ASoC: Fix widgets for WM8994/58 AIF2 source control
The compiler really ought to have been warning about unreferenced
variables...
- ASoC: Implement WM8994/58 DAC and ADC oversampling control
The oversampling rate of the DAC and ADC can be controlled to optimise
for either low power consumption or maximum performance.
- ASoC: Add ADC high pass filter support to WM8994
- ASoC: Support WM8994 mono AIF configurations
The WM8994 supports mono signals - enable this in the driver. With DSP
mode an automatic data channel selector is available, activate this
when in mono mode.
- ASoC: When disabling WM8994 FLL force a source selection
When we disable the WM8994 FLL code path sharing means that we end up
writing out a configuration. Currently this is the currently active
input and output frequency (which causes snd_soc_update_bits() to
suppress actual writes both immediately and in the common case where
we reenable the same configuration later) but we allow machine drivers
to pass through a source of zero. Since the register values written
are one less than the source constants this causes corruption of other
bitfields in the register.
Fix this by using the most recently configured FLL source when none is
provided.
- ASoC: Tune performance of WM8958 revision A
Update some of the default configuration for the device to improve
the performance.
- ASoC: Correct event flags for WM8958 AIF DACs
We need a post notification as we need to shut down the MBC after the
data stops flowing rather than before.
- ASoC: Implement runtime PM for WM8994/58
This allows us to communicate our power management state back to the
parent device, allowing it to do a full power down when the device is
idle.
- ASoC: Enable rbtree compression for WM8994/58 register cache
The WM8994 and WM8958 register map is relatively sparse so benefits
from compression. The rbtree compression gives better results than
LZO for both memory and CPU consumption on a map as sparse as this.
- ASoC: Convert WM8994 to use soc-cache.c cache functions
In the process we convert the driver to read registers one at a time
when initialising the cache. This has the effect of working around
limitations in the sizes of I2C transactions which can be done by some
CPUs. Due to the sparseness of the register map the overhead from this
should be minimual unless I2C transactions are very expensive to start.
- ASoC: Add WM8958 microphone detection support
The WM8958 contains an advanced accessory detection feature which allows
detection of up to seven different impedence levels on the microphone
bias output, including detection of video outputs. Since some of the
more involved accessory interfaces may involve noticable interactions
with external components a simple detection scheme is provided by
default with the option to provide custom handling of accessory detect.
- ASoC: Add WM8958 Multi-band compressor support
The WM8958 features a multi-band compressor which can be enabled on
any of the AIF inputs. The MBC allows different gains to be applied to
differnt audio bands, providing an improvement in perceived loudness
of the signal by avoiding overdriving the output transducers. This
patch enables support for the MBC.
- ASoC: Implement support for enhanced AIF3 on WM8958
Additional audio routing options are available on the WM8958 audio
interface 3. Add support for these.
- ASoC: Initial WM8958 audio configuration
The WM8958 is a WM8994 derivative. This patch merely ensures that some
revision specific configuration for WM8994 is not enabled on WM8994,
additional patches will add support for the new features introduced on
the WM8958.
- ASoC: wm8994 - fix memory leaks
- ASoC: Restore WM8994 volatile and readable register operations
They went AWOL during the multi-component merge.
- ASoC: Move WM8994 read/write access data into separate file
Makes the WM8994 driver file itself substantially smaller.
- ASoC: Provide ADC left/right channel source selection on WM8994
Allow the application to choose if the ADC data presented on the left
and right channels is sourced from the internal left or right channel.
This allows a mono recording to be presented as stereo on the external
bus.
- ASoC: Remove register write trace from WM8994
We now have trace in the ASoC core so we don't need to our own trace in
the driver.
- ASoC: Fix register cache setup WM8994 for multi-component
During the multi-component conversion the WM8994 register cache init
got lost.
- ASoC: Store DC offset correction for wm_hubs devices in class W mode
Providing the analogue configuration of the output path remains the same
the DC offset corrected by the DC servo will remain identical so we can
skip the callibration, reducing the startup time for the headphone output.
Implement this for the wm_hubs devices as has been done for several other
CODECs.
Don't do this if we have any analogue paths enabled since offsets may be
being introduced by the analogue paths which could vary outside the
control of the driver.
- ASoC: Remove unused vol field from WM8994 access mask table
Remove unused vol from struct access_mask
Reduces object size ~3kb.
$ size sound/soc/codecs/wm8994.o*
text data bss dec hex filename
40727 4384 4480 49591 c1b7 sound/soc/codecs/wm8994.o.new
43879 4384 4480 52743 ce07 sound/soc/codecs/wm8994.o.old
- ASoC: Restore DAI ID specification for WM8994
WM8994 relies on the DAIs having IDs that match the AIF numbers.
- ALSA: Fix Linux 2.6 merge issues
- ASoC: Fix cut'n'paste comment in WM8994
- ASoC: Added a missing 32-bit PCM format, to the wm8994 codec.
- ASoC: Implement WM8994 DAC data source muxes
Allow selection of the channel used for input to the AIFnDAC signals.
This isn't integrated into DAPM since we treat the data as a single
mono channel until just beyond this selection so it ends up having
no visible effect on the routing.
- ASoC: Log WM8994 separate ADC LRCLKs every time we configure
This makes it that little bit easier to spot the diagnostics in the
logs.
- ASoC: Only tweak WM8994 chip configuration on devices up to rev D
Any subsequent revisions will have these configuration changes applied
by default.
- ASoC: Optimise DSP performance for WM8994
Change the chip defaults to optimise performance of some of the DSP
functionality.
- ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.
- ASoC: Implement WM8994 AIF1ADC2 paths
- ASoC: Implement WM8994 OPCLK support
The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK. The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.
- ASoC: Include WM8994 GPIO and interrupt registers in codec_reg
Very handy for debug.
- ASoC: Don't restart unconfigured WM8994 FLLs
If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.
- ASoC: Add register write logging for WM8994
- ASoC: Warn on low WM8994 AIFCLK
- ASoC: Allow unspecified source when stopping WM8994 FLLs
- ASoC: Tone down debugging for WM8994 class W
It's a little verbose during path changes.
- ASoC: Set full range of WM8994 FLL Fratio values
Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.
- ASoC: Support FLL input clock selection on WM8994
The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.

SoC Codec WM8995

- ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
In the wm8995_set_tristate() function when updating the register
bits use the value and not the register index as the value argument.
- ASoC: WM8995: Add regulator handling code
- ASoC: WM8995: Remember to flush the cache on resume
- ASoC: Fix section mismatch in wm8995.c
__devinitconst can't be used for data referred in driver struct.
- ASoC: Remove WM8995 write sequencer bitfield definitions
They're very verbose and extremely repetitive so bulk up the kernel more
than is ideal. If required we can readd with WRITE_SEQUENCER_n type
definitions that cover the entire register bank in a few defines.
- ASoC: Add initial WM8995 driver
The WM8995 is a digital audio hub CODEC designed for smartphones.
The current driver supports most of the basic functionality of the
WM8995.

SoC Codec WM9081

- ASoC: Optimise WM9081 FLL performance
Tune the FLL gain for optimal performance according to evaluation
results.
- ASoC: wm9081: fix resource reclaim in wm9081_register error path
This patch fixes the error path in wm9081_register to properly free resources.

SoC Codec WM9090

- ASoC: codecs: wm9090: Fix register cache incoherency
The multi-component patch(commit f0fba2ad1) moved the allocation of the
register cache from the driver to the ASoC core. Most drivers where adjusted to
this, but the wm9090 driver still uses its own register cache for its
private functions, while functions from the ASoC core use the generic cache.
Thus we end up with two from each other incoherent caches, which can lead to
undefined behaviour.
This patch fixes the issue by changing the wm9090 driver to use the
generic register cache in its private functions.
- ASoC: sound/wm9090: add missing __devexit marker
This fixes the following warning:
sound/soc/codecs/wm9090.c:668:12: warning: 'wm9090_i2c_remove' defined but not used
- ASoC: Add WM9090 amplifier driver
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.

SoC Codec WM9712

- ASoC: Staticise WM9712 DAI list

SoC Codec alc5621/2/3

- ASoC: Fix inconsistent meaning of default case while checking alc5623->id
In alc5623_i2c_probe(),
the default case for checking alc5623->id behaves the same as case 0x23.
However, In alc5623_probe() the default case for checking alc5623->id
becomes to be the same as case 0x21.
This makes the meaning of default case inconsistent.
Since we have checked codec id in alc5623_i2c_probe() by comparing
vid2 with id->driver_data, it is not possible to run into the default case now.
In case we may add more supported devices to alc5623_i2c_table in the future,
this patch changes the default case return -EINVAL to let people know that
they should not run into this case. They should also add a new case accordingly
for the new id.
- ASoC: Fix a memory leak in alc5623_i2c_probe error path
- ASoC: add support for alc562[123] codecs
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.

SoC DaVinci

- ASoC: da8xx/omap-l1xx: match codec_name with i2c ids
The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c
is not matching with the i2c ids in the board file. Without this fix the
soundcard does not get detected on da850/omap-l138/am18x evm.
Tested-by: Dan Sharon <dansharon@nanometrics.ca>
- ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
- ASoC: davinci-vcif - fix a memory leak
- ASoC: davinci: fixes for multi-component
Multi-component commit f0fba2ad broke a few things which this patch should
fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be
good if those with access to other Davinci boards could test.
--
The multi-component commit put the initialisation of
snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the
McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c).
The initialisation had to be moved from the probe function in these drivers
because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver.
Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in
davinci-pcm.c) before hw_params is called. I have moved the initialisation to
a new snd_soc_dai_ops.startup function in each of these drivers. This fix
indicates that all platforms that use davinci-pcm must have been broken and
need to test with this fix.
--
The multi-component commit also changed the McBSP driver name from
"davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board
level references to the driver name. This change is understandable, as there
is a similarly named "davinci-mcasp" driver in davinci-mcasp.c.
There is probably no 'correct' name for this driver. The DM6446 datasheet
calls it the "ASP" and describes it as a "specialised McBSP". The DM355
datasheet calls it the "ASP" and describes it as a "specialised ASP". The
DM365 datasheet calls it the "McBSP". Rather than fix this problem by
reverting to "davinci-asp", I've elected to avoid future confusion with the
"davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also
consistent with the names of the functions in the driver. There are other
fixes required, so it was never going to be as simple as a revert anyway.
--
The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has
2 ports), so I've changed the the id of the platform_device from 0 to -1.
--
In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link
structure with the DM355 EVM as they use different cpu DAI names (the DM355
has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port).
This also means that the 2 boards need different snd_soc_card structures.
--
The codec_name entries in davinci-evm.c didn't match the i2c ids in the board
files. I have only checked and fixed the details of the names used for the
McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should
check the others.
- sound: fixed typos
- ASoC: davinci-mcasp.c: Return error code in failure
In this code, 0 is returned on failure, even though other
failures return -ENOMEM or other similar values.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@a@
identifier alloc;
identifier ret;
constant C;
expression x;
@@
x = alloc(...);
if (x == NULL) { <+... \(ret = -C; \| return -C; \) ...+> }
@@
identifier f, a.alloc;
expression ret;
expression x,e1,e2,e3;
@@
ret = 0
... when != ret = e1
*x = alloc(...)
... when != ret = e2
if (x == NULL) { ... when != ret = e3
return ret;
}
// </smpl>
- ALSA: ASoc: DaVinci Delay start of ASP to trigger
Since only 4 mainline ASoC codecs support the trigger
callback, we cannot rely upon them stopping the frame clock
if they are master and must assume it is running even if the
sound is paused. Thus we cannot start the ASP until the trigger
method.
- ASoC: davinci: let platform data define edma queue numbers
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.
This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.
platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.
Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.
Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.
This patch has been tested on DM644x and OMAP-L138 EVMs.
- ASoC: davinci: check kzalloc() result (typo)
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).
- ASoC: Remove unnecessary casts of private_data
- ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)
i2s_accurate_sck switch can be used to have a better approximate
sampling frequency.
The clock is an externally visible bit clock and it is named
i2s continuous serial clock (I2S_SCK).
The trade off is between more accurate clock (fast clock)
and less accurate clock (slow clock).
The waveform will be not symmetric.
Probably it is possible to get a better algorithm for calculating
the divider, trying to keep a slower clock as possible.
This patch has been developed against the
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
- ASoC: DaVinci: Added selection of clk input pin for McBSP
When McBSP peripheral gets the clock from an external pin,
there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
and MCBSP_CLKS.
evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
hardware connection and I use MCBSP_CLKS, so I have added
this possibility.
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm)
- ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
Added two clocking options for dm365 McBSP peripheral when used
with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
from external pin and generates frame sync).
A slave clock management can be important when the external codec needs
the system clock and the bit clock synchronized (tested with uda1345).
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
- ASoC: DaVinci: Fix McASP hardware FIFO configuration
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf
Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)
During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.
https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).
The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.

SoC Dynamic Audio Power Management

- ASoC: don't pass the string as the format arguemtn for dev_info()
- ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
This new type is a virtual version of snd_soc_dapm_mux. It is used
when a backing register value is not necessary for deciding which
input path to connect. A simple virtual enumeration control e.g.
SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used
to choose which path to connect.
The snd_soc_dapm_virt_mux type ensures that during the initial
path setup, the first (which is also the default) input path will
be connected.
- ASoC: Do DAPM control updates in the middle of DAPM sequences
Attempt to minimise audible effects from mixer and mux updates by
implementing the actual register changes between powering down widgets
that have become unused and powering up widgets that are newly used.
This means that we're making the change with the minimum set of widgets
powered, that the input path is connected when we're powering up widgets
(so things like DC offset correction can run with their signal active)
and that we bring things down to cold before switching away. Since
hardware tends to be designed for the power on/off case more than for
dynamic reconfiguration this should minimise pops and clicks during
reconfiguration while active.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: Extend DAPM to handle power changes on cross-device paths
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.
This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.
DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.
- ASoC: Move widgets from DAPM context to snd_soc_card
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
- ASoC: Move DAPM paths from DAPM context to snd_soc_card
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
- ASoC: Remove unused DAPM_DOUBLE control types
There are no users of these and it's not clear what they would do given
the mono flow modelling which DAPM does. If need arises we can add them
again.
- ASoC: dapm: Add output driver widget
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.
Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
- ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38
Fix "ASoC: Fix bias power down of non-DAPM codec" for 3.6.37 will cause a
build error when merging into ASoC for-2.6.38. Fix the issue by doing a
change that commit ce6120c "ASoC: Decouple DAPM from CODECs" would do.
- ASoC: Add post-CODEC bias level callback for machine driver
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
- ASoC: Fix bias power down of non-DAPM codec
Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.
Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.
- ASoC: Remove cyclic dependency between soc.h and soc-dapm.h/soc-dai.h
There is no need anymore to include soc.h in soc-dapm.h and soc-dai.h as
drivers are converted to include only soc.h.
Thanks to Lars-Peter Clausen <lars@metafoo.de> for pointing out the issue.
- ASoC: Use DAPM context rather than CODEC when constructing sequences
DAPM widgets may be associated with non-CODEC devices so compare based
on the DAPM context rather than the CODEC pointer.
- ASoC: Add optional name_prefix for codec kcontrol, widget and route names
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.
This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.
Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.
static const struct snd_soc_dapm_route map0[] = {
{"Spk", NULL, "MONO"},
};
static const struct snd_soc_dapm_route map1[] = {
{"Vibra", NULL, "foo MONO"},
};
static struct snd_soc_prefix_map codec_prefix[] = {
{
.dev_name = "codec.2",
.name_prefix = "foo",
},
};
static struct snd_soc_card card = {
...
.prefix_map = codec_prefix,
.num_prefixes = ARRAY_SIZE(codec_prefix),
};
- ASoC: Add DAPM trace events
Trace events for DAPM allow us to monitor the performance and behaviour
of DAPM with logging which can be built into the kernel permanantly, is
more suited to automated analysis and display and less likely to suffer
interference from other logging activity.
Currently trace events are generated for:
- Start and stop of DAPM processing
- Start and stop of bias level changes
- Power decisions for widgets
- Widget event execution start and stop
giving some view as to what is happening and where latencies occur.
Actual changes in widget power can be seen via the register write trace in
soc-core.
- ASoC: Factor out boiler plate for DAPM event generation
Make the DAPM sequence execution look a bit nicer by factoring out the
code to invoke an event into a single function since it's all the same
pretty much.
- ASoC: Convert pop_dbg to use dev_info
Prints from pop_dbg are enabled when dapm_pop_time != 0. Convert it to
use dev_info so that parent device of DAPM context is printed.
- ASoC: Update DAPM debug and error prints
Switch printk and pr_ prints to dev_ variants. It is helpful to see
parent device of DAPM context especially when there are multiple DAPM
contexts (codecs currently).
This is mostly simple conversion. Exceptions are in snd_soc_dapm_set_pin
that prints also pin state, uniform "dapm: unknown pin" error prints from
snd_soc_dapm_set_pin, snd_soc_dapm_force_enable_pin and
snd_soc_dapm_ignore_suspend, and pop_dbg which is converted by an another
patch.
- ASoC: Move pop time from DAPM context to sound card
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
- ASoC: Decouple DAPM from CODECs
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
- ASoC: Fix dapm_seq_compare() for multi-component
Ensure that we keep all widget powerups in DAPM sequence by making
the CODEC the last thing we compare on rather than the first thing.
Also fix the fact that we're currently comparing the widget pointers
rather than the CODEC pointers when we do the substraction so we
won't get stable results.
- ASoC: Add event variants of the AIF widgets
- ASoC: multi-component - ASoC Multi-Component Support
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Some TLV320aic23 and Cirrus platform fixes.
TI CODEC and OMAP fixes
Samsung platform and misc fixes :-
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
MPC8610 and PPC fixes.
i.MX fixes and some core fixes.
J4740 platform fixes:-
- ASoC: Add SND_SOC_DAPM_PRE_POST_PMD event
Some systems codecs need to configure some registers before and after
powering down some of their part. As a convenience add a macro for that.
- ASoC: Support leaving paths enabled over system suspend
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: Refactor DAPM suspend handling
Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.
Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: Remove unused DAPM suspend flag
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>

SoC EP93XX

- Add the build-stub for soc/ep93xx
- ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.
I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3
Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:
Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.
Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.
So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.
In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.
2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.
DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...
If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).
In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.
- ASoC: EP93xx: sampling rate range extended
Changes to both I2S and PCM code:
- Rates list extended up to 96kHz, it's tested on EDB9302 and works for both capture and
playback.
- ASoC: simone: fix resource leak in simone_init error path
Fix the error path to properly free allocated resources.
- ASoC: ep93xx: add Simplemachines Sim.One AC97 audio support
Add AC97 audio support for Simplemachines Sim.One board.
- ASoC: add ep93xx AC97 audio driver
Add support for AC97 controllers found in Cirrus Logic EP93xx family SoCs.
- ASoC: EP93xx: Add Snapper CL15 i2s audio support
Add support for i2s audio on Bluewater Systems Snapper CL15 module
- ASoC: ep93xx i2s audio driver
Add ep93xx i2s audio driver

SoC FSI SH7724

- ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMI
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.
- ASoC: fsi: modify compile error
- ASoC: multi-component - ASoC Multi-Component Support
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Some TLV320aic23 and Cirrus platform fixes.
TI CODEC and OMAP fixes
Samsung platform and misc fixes :-
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
MPC8610 and PPC fixes.
i.MX fixes and some core fixes.
J4740 platform fixes:-
- ASoC: fsi: Add new funtion for SPDIF
- ASoC: fsi: Add specified ID for soc-audio
Specified ID is necessary, when some codecs are used with FSI.
- ASoC: fsi: Fixup for master mode
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
- ASoC: fsi: modify format area definition on flags
There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence number

SoC Freescale

- ASoC: Add missing dev_set_drvdata in p1022_ds_probe
Otherwise, calling dev_get_drvdata in p1022_ds_remove returns NULL.
- ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL.
- ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path
Add missing platform_device_put() if platform_device_add() failed.
- ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path
Add missing platform_device_put() if platform_device_add() failed.
- ASoC: MPC5200: Eliminate duplicate include of of_device.h
Eliminate duplicate #include <linux/of_device.h> from
sound/soc/fsl/mpc5200_dma.c
- ASoC: Return proper error if snd_soc_register_dais fails in psc_i2s_of_probe
- ASoC: fsl - fix build error in pcm030-audio-fabric.c
Fix build error:-
sound/soc/fsl/pcm030-audio-fabric.c:27:33: fatal error:
sound/soc-of-simple.h: No such file or directory
- ASoC: pl022_ds.c: Add of_node_put to avoid memory leak
Add a call to of_node_put in the error handling code following a call to
of_parse_phandle.
This patch also moves the existing call to of_node_put tothe end of the
error handling code, to make it possible to jump to of_node_put without
doing the other cleanup operations. These appear to be disjoint
operations, so the ordering doesn't matter.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
local idexpression x;
expression E,E1,E2;
statement S;
@@
*x =
(of_find_node_by_path
|of_find_node_by_name
|of_find_node_by_phandle
|of_get_parent
|of_get_next_parent
|of_get_next_child
|of_find_compatible_node
|of_match_node
|of_find_node_by_type
|of_find_node_with_property
|of_find_matching_node
|of_parse_phandle
)(...);
...
if (x == NULL) S
<... when != x = E
*if (...) {
... when != of_node_put(x)
when != if (...) { ... of_node_put(x); ... }
(
return <+...x...+>;
|
* return ...;
)
}
...>
(
E2 = x;
|
of_node_put(x);
)
// </smpl>
- ASoC: mpc8610: replace of_device with platform_device
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC8610 HPCD audio drivers (fsl_ssi, fsl_dma,
and mpc8610_hpcd) accordingly.
Also add a #include for slab.h, which is now needed for kmalloc and kfree.
- ASoC: add support for the Freescale P1022 DS reference board
The Freescale P1022 is a dual-core e500-based SOC with multimedia capabilities,
specifically the same SSI audio controller on the MPC8610. The P1022 DS
reference board includes a P1022 and a Wolfson Microelectronics WM8776
codec.
- ASoC: add support for separate codec DAIs to the fsl_dma driver
Some codecs have separate DAIs for playback and capture, so the DMA driver
should allocate a DMA buffer only for the streams that are valid when the
driver is opened.
- sound: asoc/multi-component: fsl: add support for variable SSI FIFO depth
Add code that programs the DMA and SSI controllers differently based on the
FIFO depth of the SSI.
The SSI devices on the MPC8610 and the P1022 are identical in every way except
one: the transmit and receive FIFO depth. On the MPC8610, the depth is eight.
On the P1022, it's fifteen. The device tree nodes for the SSI include a
"fsl,fifo-depth" property that specifies the FIFO depth.
- sound: asoc/multi-component: fsl: add support for disabled SSI nodes
Add support for adding "status = disabled" to an SSI node to incidate that it
is not wired on the board. This replaces the not-so-intuitive previous method
of omitting a codec-handle property.
- sound: asoc/multi-component: fsl: fix exit and error paths in DMA and SSI drivers
The error handling code in the OF probe function of the SSI driver is not
freeing all resources correctly.
Since the machine driver no longer calls the DMA driver to provide information
about the SSI, we don't need to keep a list of DMA objects any more. In
addition, the fsl_soc_dma_remove() function is incorrectly removing *all*
DMA objects when it should only remove one.
- sound: asoc/multi-component: fsl: add support for 36-bit physical addresses
Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.
The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture. This function is just a front-end for
dma_alloc_coherent(). Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.
- sound: powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs. Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.
- sound/soc: mpc5200_psc_ac97: Use gpio pins for cold reset
Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)
From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
- of/device: Replace struct of_device with struct platform_device
of_device is just an alias for platform_device, so remove it entirely. Also
replace to_of_device() with to_platform_device() and update comment blocks.
This patch was initially generated from the following semantic patch, and then
edited by hand to pick up the bits that coccinelle didn't catch.
@@
@@
-struct of_device
+struct platform_device
Reviewed-by: David S. Miller <davem@davemloft.net>
- fix typos concerning "initiali[zs]e"
- powerpc/5200: Fix build error in sound code.
Compiling in the MPC5200 sound drivers results in the following build error:
sound/soc/fsl/mpc5200_psc_ac97.o: In function `to_psc_dma_stream':
mpc5200_psc_ac97.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
sound/soc/fsl/efika-audio-fabric.o: In function `to_psc_dma_stream':
efika-audio-fabric.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
make[3]: *** [sound/soc/fsl/built-in.o] Error 1
make[2]: *** [sound/soc/fsl] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
This patch fixes it by declaring the inline function in the header file to
also be a static.
Tested-by: John Hilmar Linkhorst <John.Linkhorst@rwth-aachen.de>
- ASoC: Remove unused header from MPC5200 PSC driver
The header contains an extern that isn't used by anything. Remove.
- of: Remove duplicate fields from of_platform_driver
.name, .match_table and .owner are duplicated in both of_platform_driver
and device_driver. This patch is a removes the extra copies from struct
of_platform_driver and converts all users to the device_driver members.
This patch is a pretty mechanical change. The usage model doesn't change
and if any drivers have been missed, or if anything has been fixed up
incorrectly, then it will fail with a compile time error, and the fixup
will be trivial. This patch looks big and scary because it touches so
many files, but it should be pretty safe.
- of: Always use 'struct device.of_node' to get device node pointer.
The following structure elements duplicate the information in
'struct device.of_node' and so are being eliminated. This patch
makes all readers of these elements use device.of_node instead.
(struct of_device *)->node
(struct dev_archdata *)->prom_node (sparc)
(struct dev_archdata *)->of_node (powerpc & microblaze)

SoC Ingenic JZ4740

- Add soc/jz4740 build stub
- ASoC: Fix compile breakage in jz4740.c and smartq_wm8987.c
Commit ce6120c require that soc-dapm.h cannot be included before soc.h but
these two drivers were not checked. Fix them by including only soc.h as it
includes soc-dapm.h.
- ASoC: Multi-component: JZ4740: QI_LB60 board fixes
This patch contains two small fixes for the sound board driver for the qi_lb60
introduced by the multi-component patches:
* Remove unnecessary includes: Those includes where only used to get the
definitions for the DAI devices and are thus not needed anymore.
* Fix a typo.
- ASoC: JZ4740: Add qi_lb60 board driver
This patch adds ASoC support for the qi_lb60 board.
- ASoC: Add JZ4740 codec driver
This patch adds support for the JZ4740 internal codec.
- ASoC: Add JZ4740 ASoC support
This patch adds ASoC support for JZ4740 SoCs I2S module.

SoC Intel Medfield MID platform

- add missing files for soc/mid-x86 and soc/tegra
- Add build stubs for tegra and mid-x86 ASoC drivers
- sound: ASoC sst v2: Add medfield machine driver
This patch adds the medfield machine driver
Machine driver glues the sn95031 codec driver and sst_platform driver to form the asoc sound driver
- ASoC: sst_platform porting sst dsp driver interface as per latest in Greg's staging tree
The interface between sst platform driver and intel sst dsp driver
have been changed in Greg's staging tree - next branch
This patch adds the interface changes compatible with the new interface
in Greg's staging tree
- ASoC: sst_platform created helper functions
Few funtions can be modularized in this driver to make them look cleaner
like managing the stream status with locks and filling pcm parameters.
This patch adds helper functions to do the same.
- ASoC: sst platform - fix the style inconsistency
this patch fixes the style inconsistency by removing empty space in MODULE_ALIAS
also removes a redundant return statement
- ASoC: sst v2: Add mid platform driver
This patch adds the platform driver for mid asoc drivers. This platfrom
driver sends commands to sst dsp engine driver for the dai operations.
For this purpose it depends on intel_sst driver which is currently in
staging tree
- ASoC: Add dependency on INTEL_SCU_IPC for Intel MID drivers
- ASoC: mid-x86 - remove the flag in makefile
previous commit e62255f2adf2 introduced DDEBUG flag in Makefile
This causes all debug statemenets to be ON.
This patch removes this flag
- ASoC: sst v2: Add makefiles and kconfigs changes
This patch adds the makefile and kconfig changes for mid asoc drivers:
platform and machine driver which are introduced in 2 preceeding patches

SoC Layer

- Regenerated soc-core.patch
- Fix soc-core build with 2.6.26 or older kernels
- ASoC: Update name of debugfs root symbol to snd_soc_
Everything else is using snd_soc_ so we should use it here too.
- ASoC: core: Remove dapm_sync call from soc_post_component_init
snd_soc_dapm_new_widgets will call dapm_power_widgets at
the end, so there is no need to call snd_soc_dapm_sync
after snd_soc_dapm_new_widgets.
Reviewed-by: Jarkko Nikula <jhnikula@gmail.com>
- ASoC: WM8991: Add initial WM8991 driver
The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for
handsets rich in multimedia features such as GPS, mobile TV, digital audio
playback and gaming.
This driver was originally written by Graeme Gregory and has been maintained
out of tree by Mark Brown and Dimitris Papastamos.
- ASoC: soc core allow machine driver to register the card
The machine driver can't register the card directly and need to do this thru
soc-audio device creation
This patch allows the register and unregister card to be directly called by
machine drivers
- ASoC: soc core add inline to handle card list initialzation
Currently the soc_probe initializes the card hence it does the card list
initialzation. But if machines directly register the card they would need to
do these steps, so putting them as inline would save lot of code
This patch adds an inline to do list initialzation
- ASoC: soc core move the card debugfs initialization
The card debugfs initialization is done in soc_probe but would be better if it
is done when the card in registered
This patch moves the debugfs initialization to register_card()
- ASoC: soc core move card cleanup from soc_remove()
In soc_remove() the card resources are cleaned up.
This can also be done in card_unregister()
This patch move this cleanup into a new function and calls it from
card_unregister. This paves way for further work to allow card registartion
from machine.
- ASoC: Automatically assign the default readable()/volatile() functions
Ensure that all calls to readable_register()/volatile_register() go via
the snd_soc_codec function pointers.
If the default register access table has been given but no functions
for handling readable()/volatile() registers, use the default ones provided
by soc-cache.
- ASoC: Update users of readable_register()/volatile_register()
- ASoC: soc-cache: Add support for default readable()/volatile() functions
For common scenarios, device drivers can provide a table of all the
registers that are at least either readable/writable/volatile. The idea
is that if a register lookup fails, all of its read/write/vol members
will be zero and will be treated as default. This also reduces the
size of the register access array.
- ASoC: soc-cache: Add reg_size as a member to snd_soc_codec
Simplify the use of reg_size, by calculating it once and storing it in
the codec structure for later reference. The value of reg_size is
reg_cache_size * reg_word_size.
- ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache
With the addition of the cache fallback functionality, it is necessary
to ensure that if the register defaults cache was marked as __devinitconst
and the LZO compression is not compiled in the kernel, the fallback to
flat compression will still use a copy of the defaults cache.
- ASoC: Fix indentation in soc_remove()
- ASoC: soc-core: Simplify compress_type overriding functionality
- ASoC: soc-cache: Clean up the cache manipulation code
Use Takashi's clean up code to make the cache manipulation code more
readable.
- ASoC: let snd_soc_update_bits() return an error code
Update snd_soc_update_bits() so that it returns a negative error code if the
the read or write operation fails.
Note that currently, a lot of the lower-level read functions have an unsigned
integer return type (and some of them even try to return a negative number),
but this code still appears to work in those cases.
An examination of the code shows that all current callers are compatible with
this change.
- ASoC: tegra: Kconfig and Makefile
- ASoC: Export debugfs root dentry
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.
Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.
- ASoC: soc-core: Add support for NULL default register caches
The infrastructure for handling NULL default register maps is already
included in the soc-cache code, just ensure that we don't try to dereference
a NULL pointer while accessing a NULL register cache.
- ASoC: Taint the kernel if debugfs is used to write directly to CODECs
Since direct register writes may confuse the drivers and are supposed
to be used only in diagnostic situations discourage their use in
production by tainting the kernel when we do a write.
- ASoC: sst v2: Add makefiles and kconfigs changes
This patch adds the makefile and kconfig changes for mid asoc drivers:
platform and machine driver which are introduced in 2 preceeding patches
- ASoC: sst v2: Add sn95031 codec driver
This patch adds the sn95031 asoc codec driver. This driver currently
supports only playback. Capture and jack detection to be added later
- ASoC: WL1273 FM radio: Fix breakage with MFD API changes
These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.
Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.
- ASoC: DMIC codec: Adding a generic DMIC codec
This codec is to be used by the DMIC driver to
control the DMIC codec. This driver will be used on future
implementations of the DMIC driver to support codec specific
features.
At this time, the codec driver just registers the codec DAI.
- ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync()
The size of the lzo syncing bitmap was incorrectly set to the size
of the cache times the word size, however, the correct size is the
size of the cache.
- ASoC: Fix the device references to codec and platform drivers
The soc-core takes the platform and codec driver reference during probe. Few of
these references are not released during remove. This cause the platform and
codec driver module unload to fail.
This patch fixes by the taking only one reference to platform and codec module
during probe and releases them correctly during remove. This allows load/unload
properly
- ASoC: Fix AC'97 registration unwind
soc_unregister_ac97_dai_link() takes a CODEC as an argument, not a
rtd like the registration function, so give it what it's looking for.
- ASoC: Fix snd_soc_instantiate_card error path
Properly free the resources in the case of snd_card_register failure
and soc_register_ac97_dai_link failure.
- ASoC: Add initial WM8995 driver
The WM8995 is a digital audio hub CODEC designed for smartphones.
The current driver supports most of the basic functionality of the
WM8995.
- ASoC: Make LZO cache compression optional
Make LZO cache compression optional as it pulls in the kernel wide LZO
implementation and rbtree compression is generally more efficient for
typical register maps, especially in terms of CPU performance.
- ASoC: If we can't find a cache compression type default to flat
This makes it easier to make cache types build time configurable as we
don't have a hard dependency on a given cache being built in.
- ASoC: Extend DAPM to handle power changes on cross-device paths
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.
This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.
DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.
- ASoC: Move widgets from DAPM context to snd_soc_card
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
- ASoC: Move DAPM paths from DAPM context to snd_soc_card
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
- ASoC: soc-cache: A few minor stylistic changes
Remove redundant parentheses/spaces in the use of the sizeof
operator.
- ASoC: Remove unnecessary structure definitions
This patch removes some legacy structure definitions which are not using
in current ASoC drivers.
- ASoC: soc-core: Fix null pointer dereference
In case the codec driver did not provide a read/write function,
codec->driver->read|write will be NULL. Ensure that we use the one
specified in codec->read|write to avoid oopsing when we access
the debugfs entries. This is achieved by using snd_soc_read() and
snd_soc_write().
- ASoC: soc-core: Remove useless inline function construct
There is no need to mark this function as inline. Inline functions
usually are small and concise functions that benefit from not needing
to set up a stack frame and undergo a call/ret sequence upon each
invocation.
- ASoC: soc-core: Replace use of strncpy() with strlcpy()
By using strncpy() if the source string does not have a null byte in the
first n bytes, then the destination string is not null-terminated.
This can be fixed in a two-step process by manually null-terminating the
array after the use of strncpy() or by using strlcpy().
- ASoC: Merge common code in DAI link and auxiliary codec probing/removal
Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs"
added much of code that can be shared with DAI link codec probing/removal.
Merge now this common code into new soc_probe_codec, soc_remove_codec and
soc_post_component_init functions.
Error prints in these functions are converted to use dev_err and to print
the error code.
- ASoC: soc-cache: Add optional cache name member to snd_soc_cache_ops
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.
Remove redundant newline in source code.
- ASoC: Add trace events for jack detection
As jack detection can trigger DAPM and the latency in debouncing can create
confusing windows in operation provide some trace events which will hopefully
help in diagnostics. The soc-jack core traces all reports that it gets and
the resulting notifications to upper layers. An event for jack IRQs is also
provided for instrumentation of debounce, and used in the GPIO jack code.
- ASoC: Add post-CODEC bias level callback for machine driver
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
- ASoC: Constify struct snd_soc_codec_driver
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.
- ASoC: soc-core: Allow machine drivers to override compress_type
This patch allows machine drivers to override the compression type
provided by the codec driver.
- ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary. This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.
- ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf
The snd_soc_codec_conf struct now holds codec specific configuration
information.
A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.
In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead. In all other cases the one
set in the snd_soc_codec_conf struct takes effect.
- ASoC: Change the base value of compress_type
Ensure that the base value of compress_type starts at 1 so that
we know whether the machine driver has provided a compress_type
for overriding the codec supplied one.
- ASoC: Add compress_type as a member to snd_soc_codec
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with. The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.
Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.
- ASoC: Don't oops in soc_probe_aux_dev in case of missing codec
Blind copy of codec finding algorithm from soc_bind_dai_link does not work
in soc_probe_aux_dev if matching codec name is not found. In that case the
code falls through and tries to start the probing procedure with invalid
codec pointer.
Fix this and add an error print showing the codec name that cannot be found.
- ASoC: Move active copy of CODEC read and write into runtime structure
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.
- ASoC: Make the DAI ops constant in the DAI structure
Neither drivers nor the core should be fiddling with the actual ops
structure at runtime.
- ASoC: Apostrophe patrol in soc-core.c
Silly little grammar nit but it bugs the hell out of me.
- ASoC: Fix build failure in soc-core.c
Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs"
causes a build failure in soc-core.c: soc_probe_aux_dev since code tries to
access non-existing struct snd_soc_dapm_context and struct snd_soc_card
members.
Root cause for this was a double accident. Author sent the RFC patch from
top of another patch set and the RFC got committed. Fix the build failure
by removing the code line that depends on that another patch set.
- ASoC: Add support for optional auxiliary dailess codecs
This makes possible to register auxiliary dailess codecs in a machine
driver. Term dailess is used here for amplifiers and codecs without DAI or
DAI being unused.
Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs
are probed after initializing the DAI links. There are no major differences
between DAI link codecs and dailess codecs in ASoC core point of view. DAPM
handles them equally and sysfs and debugfs directories for dailess codecs
are similar except the pmdown_time node is not created.
Only suspend and resume functions are modified to traverse all probed codecs
instead of DAI link codecs.
Example below shows a dailess codec registration.
struct snd_soc_aux_dev foo_aux_dev[] = {
{
.name = "Amp",
.codec_name = "codec.2",
.init = foo_init2,
},
};
static struct snd_soc_card card = {
...
.aux_dev = foo_aux_dev,
.num_aux_devs = ARRAY_SIZE(foo_aux_dev),
};
- ASoC: soc-cache: Fix memory overflow in LZO initialization
The bitmap_zero() nbits argument was improperly set to reg_size
but the underlying buffer was bmp_size long. This caused the memset
to zero past the end of the allocated buffer and into the kernel heap
causing strange kernel crashes sometimes by overwriting critical
kernel structures.
- ASoC: Fix resource leak if soc_register_ac97_dai_link failed
Properly free the resources in the case of soc_register_ac97_dai_link failure.
- ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
As the comments of snd_soc_instantiate_cards() said,
snd_soc_instantiate_cards() must be called with client_mutex.
- ASoC: Samsung: Rename from s3c24xx to samsung
Finally, move the 's3c24xx' directory to 'samsung'
- ASoC: soc-cache: Ensure consistent cache naming
- ASoC: soc-cache: Add error checking in the *_cache_sync functions
Ensure that we report any errors encountered during reads/writes
in the cache syncing functions.
Remove redundant newline in the source code.
- ASoC: Add initial WM8737 driver
The WM8737 is a low power, flexible stereo ADC designed for portable
applications. This driver supports most of the functionality of the
WM8737, though some features such as the ALC are not yet implemented.
- ASoC: Remove cyclic dependency between soc.h and soc-dapm.h/soc-dai.h
There is no need anymore to include soc.h in soc-dapm.h and soc-dai.h as
drivers are converted to include only soc.h.
Thanks to Lars-Peter Clausen <lars@metafoo.de> for pointing out the issue.
- ASoC: Add optional name_prefix for codec kcontrol, widget and route names
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.
This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.
Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.
static const struct snd_soc_dapm_route map0[] = {
{"Spk", NULL, "MONO"},
};
static const struct snd_soc_dapm_route map1[] = {
{"Vibra", NULL, "foo MONO"},
};
static struct snd_soc_prefix_map codec_prefix[] = {
{
.dev_name = "codec.2",
.name_prefix = "foo",
},
};
static struct snd_soc_card card = {
...
.prefix_map = codec_prefix,
.num_prefixes = ARRAY_SIZE(codec_prefix),
};
- ASoC: Prevent system suspend while debouncing wakeup capable GPIO jacks
If the device associated with a GPIO jack is wakeup capable then disable
suspend while we're debouncing the jack so that we skip suspends that race
with the jack.
Note that currently the GPIO based jack has a CODEC associated with it
which we're using right now. These jacks should be reparented against the
card itself and this code adjusted.
- ASoC: soc-cache: Add support for rbtree based register caching
This patch adds support for rbtree compression when storing the
register cache. It does this by not adding any uninitialized registers
(those whose value is 0). If any of those registers is written
with a nonzero value they get added into the rbtree.
Consider a sample device with a large sparse register map. The
register indices are between [0, 0x31ff]. An array of 12800 registers
is thus created each of which is 2 bytes. This results in a 25kB
region. This array normally lives outside soc-core, normally in the
driver itself. The original soc-core code would kmemdup this region
resulting in 50kB total memory. When using the rbtree compression
technique and __devinitconst on the original array the figures are
as follows. For this typical device, you might have 100 initialized
registers, that is registers that are nonzero by default. We build
an rbtree with 100 nodes, each of which is 24 bytes. This results
in ~2kB of memory. Assuming that the target arch can freeup the
memory used by the initial __devinitconst array, we end up using
about ~2kB bytes of actual memory. The memory footprint will increase
as uninitialized registers get written and thus new nodes created in
the rbtree. In practice, most of those registers are never changed.
If the target arch can't freeup the __devinitconst array, we end up
using a total of ~27kB. The difference between the rbtree and the LZO
caching techniques, is that if using the LZO technique the size of
the cache will increase slower as more uninitialized registers get
changed.
- ASoC: soc-cache: Add support for LZO register caching
This patch adds support for LZO compression when storing the register
cache. The initial register defaults cache is marked as __devinitconst
and the only change required for a driver to use LZO compression is
to set the compress_type member in codec->driver to SND_SOC_LZO_COMPRESSION.
For a typical device whose register map would normally occupy 25kB or 50kB
by using the LZO compression technique, one can get down to ~5-7kB. There
might be a performance penalty associated with each individual read/write
due to decompressing/compressing the underlying cache, however that should not
be noticeable. These memory benefits depend on whether the target architecture
can get rid of the memory occupied by the original register defaults cache
which is marked as __devinitconst. Nevertheless there will be some memory
gain even if the target architecture can't get rid of the original register
map, this should be around ~30-32kB instead of 50kB.
- ASoC: soc-cache: Add support for flat register caching
This patch introduces the new caching API and migrates the
old caching interface into the new one. The flat register caching
technique does not use compression at all and it is equivalent to
the old caching technique. One can still access codec->reg_cache
directly but this is not advised as that will not be portable
across different caching strategies.
None of the existing drivers need to be changed to adapt to this
caching technique. There should be no noticeable overhead associated
with using the new caching API.
- ASoC: Add trace events for ASoC register read/write
The trace subsystem provides a convenient way of instrumenting the kernel
which can be left on all the time with extremely low impact on the system
unlike prints to the kernel log which can be very spammy. Begin adding
support for instrumenting ASoC via this interface by adding trace for the
register access primitives.
- ASoC: Fix compile error if CONFIG_DEBUG_FS is not configured
Add soc_init_card_debugfs and soc_cleanup_card_debugfs functions to fix below error.
CC sound/soc/soc-core.o
sound/soc/soc-core.c: In function 'soc_probe':
sound/soc/soc-core.c:1689: error: implicit declaration of function 'soc_init_card_debugfs'
sound/soc/soc-core.c: In function 'soc_remove':
sound/soc/soc-core.c:1718: error: implicit declaration of function 'soc_cleanup_card_debugfs'
make[2]: *** [sound/soc/soc-core.o] Error 1
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: soc-cache: Use BUG_ON() for unsupported hw_read() calls
Instead of dereferencing a NULL function pointer and falling apart
use BUG_ON() for any unimplemented hw_read() calls.
- ASoC: WM8770: Initial driver
The WM8770 is a high performance, multi-channel audio
codec. The WM8770 is ideal for surround sound processing
applications for home hi-fi, automotive and other audio
visual equipment.
- ASoC: Move pop time from DAPM context to sound card
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
- ASoC: Move codec debugfs directories under parent card directory
Make use of sound card debugfs directory and move codec directories under
the parent card debugfs directory.
debugfs/asoc/{codec dir} -> debugfs/asoc/{card->name}/{codec dir}.
- ASoC: Add sound card directory under debugfs/asoc/
There will be need to have sound card specific debugfs entries. This patch
introduces a new debugfs/asoc/{card->name}/ directory but does not add yet
any entries there.
- ASoC: Decouple DAPM from CODECs
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
- ASoC: soc-cache: Remove unnecessary debugging info
No need to print the register-value pair again, as we've already hooked
snd_soc_write() for that matter.
- ASoC: Push snd_soc_write() and snd_soc_read() into the source file
Facilitating adding trace type stuff. For a first pass add some dev_dbg()
statements into them.
- ASoC: Convert soc-jack code to use request_any_context_irq()
Allow the standard soc-jack GPIO based jack handling to handle the use of
GPIOs which may sleep (such as those on GPIO expanders) by converting the
code to use request_any_context_irq().
- ASoC: Fix snd_soc_register_dais error handling
kzalloc for dai may fail at any iteration of the for loop,
thus properly unregister already registered DAIs before return error.
The error handling code in snd_soc_register_dais() already ensure all the DAIs
are unregistered before return error, we can remove the error handling code
to unregister DAIs in snd_soc_register_codec().
- ASoC: Fix SND_SOC_ALL_CODECS typo for alc5623
Include alc5623.c in SND_SOC_ALL_CODECS when dependencies are met.
- ASoC: add support for alc562[123] codecs
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.
- ASoC: Include cx20442 to SND_SOC_ALL_CODECS
- ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740
Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met.
- ASoC: Check return value of struct_strtoul() in pmdown_time_set()
strict_strtoul() has just been made must check so do so.
- ASoC: soc-core: Fix codec->name memory leak
Ensure that the codec->name is freed when unregistering the codec.
- ASoC: Fix I2C component device id number creation
Use bitwise AND instead of logical AND when masking.
- ASoC: Restore MAX98088 CODEC driver
This reverts commit f6765502f8daae3d237a394889276c8987f3e299 and adds
the missing include file.
- Revert "ASoC: Add max98088 CODEC driver"
This reverts commit e65d255a5a610ce9e185e0df855250706aca7c21 since it
does not build.
- llseek: automatically add .llseek fop
All file_operations should get a .llseek operation so we can make
nonseekable_open the default for future file operations without a
.llseek pointer.
The three cases that we can automatically detect are no_llseek, seq_lseek
and default_llseek. For cases where we can we can automatically prove that
the file offset is always ignored, we use noop_llseek, which maintains
the current behavior of not returning an error from a seek.
New drivers should normally not use noop_llseek but instead use no_llseek
and call nonseekable_open at open time. Existing drivers can be converted
to do the same when the maintainer knows for certain that no user code
relies on calling seek on the device file.
The generated code is often incorrectly indented and right now contains
comments that clarify for each added line why a specific variant was
chosen. In the version that gets submitted upstream, the comments will
be gone and I will manually fix the indentation, because there does not
seem to be a way to do that using coccinelle.
Some amount of new code is currently sitting in linux-next that should get
the same modifications, which I will do at the end of the merge window.
Many thanks to Julia Lawall for helping me learn to write a semantic
patch that does all this.
===== begin semantic patch =====
// This adds an llseek= method to all file operations,
// as a preparation for making no_llseek the default.
//
// The rules are
// - use no_llseek explicitly if we do nonseekable_open
// - use seq_lseek for sequential files
// - use default_llseek if we know we access f_pos
// - use noop_llseek if we know we don't access f_pos,
// but we still want to allow users to call lseek
//
@ open1 exists @
identifier nested_open;
@@
nested_open(...)
{
<+...
nonseekable_open(...)
...+>
}
@ open exists@
identifier open_f;
identifier i, f;
identifier open1.nested_open;
@@
int open_f(struct inode *i, struct file *f)
{
<+...
(
nonseekable_open(...)
|
nested_open(...)
)
...+>
}
@ read disable optional_qualifier exists @
identifier read_f;
identifier f, p, s, off;
type ssize_t, size_t, loff_t;
expression E;
identifier func;
@@
ssize_t read_f(struct file *f, char *p, size_t s, loff_t *off)
{
<+...
(
*off = E
|
*off += E
|
func(..., off, ...)
|
E = *off
)
...+>
}
@ read_no_fpos disable optional_qualifier exists @
identifier read_f;
identifier f, p, s, off;
type ssize_t, size_t, loff_t;
@@
ssize_t read_f(struct file *f, char *p, size_t s, loff_t *off)
{
... when != off
}
@ write @
identifier write_f;
identifier f, p, s, off;
type ssize_t, size_t, loff_t;
expression E;
identifier func;
@@
ssize_t write_f(struct file *f, const char *p, size_t s, loff_t *off)
{
<+...
(
*off = E
|
*off += E
|
func(..., off, ...)
|
E = *off
)
...+>
}
@ write_no_fpos @
identifier write_f;
identifier f, p, s, off;
type ssize_t, size_t, loff_t;
@@
ssize_t write_f(struct file *f, const char *p, size_t s, loff_t *off)
{
... when != off
}
@ fops0 @
identifier fops;
@@
struct file_operations fops = {
...
};
@ has_llseek depends on fops0 @
identifier fops0.fops;
identifier llseek_f;
@@
struct file_operations fops = {
...
.llseek = llseek_f,
...
};
@ has_read depends on fops0 @
identifier fops0.fops;
identifier read_f;
@@
struct file_operations fops = {
...
.read = read_f,
...
};
@ has_write depends on fops0 @
identifier fops0.fops;
identifier write_f;
@@
struct file_operations fops = {
...
.write = write_f,
...
};
@ has_open depends on fops0 @
identifier fops0.fops;
identifier open_f;
@@
struct file_operations fops = {
...
.open = open_f,
...
};
// use no_llseek if we call nonseekable_open
////////////////////////////////////////////
@ nonseekable1 depends on !has_llseek && has_open @
identifier fops0.fops;
identifier nso ~= "nonseekable_open";
@@
struct file_operations fops = {
... .open = nso, ...
+.llseek = no_llseek, /* nonseekable */
};
@ nonseekable2 depends on !has_llseek @
identifier fops0.fops;
identifier open.open_f;
@@
struct file_operations fops = {
... .open = open_f, ...
+.llseek = no_llseek, /* open uses nonseekable */
};
// use seq_lseek for sequential files
/////////////////////////////////////
@ seq depends on !has_llseek @
identifier fops0.fops;
identifier sr ~= "seq_read";
@@
struct file_operations fops = {
... .read = sr, ...
+.llseek = seq_lseek, /* we have seq_read */
};
// use default_llseek if there is a readdir
///////////////////////////////////////////
@ fops1 depends on !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier readdir_e;
@@
// any other fop is used that changes pos
struct file_operations fops = {
... .readdir = readdir_e, ...
+.llseek = default_llseek, /* readdir is present */
};
// use default_llseek if at least one of read/write touches f_pos
/////////////////////////////////////////////////////////////////
@ fops2 depends on !fops1 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier read.read_f;
@@
// read fops use offset
struct file_operations fops = {
... .read = read_f, ...
+.llseek = default_llseek, /* read accesses f_pos */
};
@ fops3 depends on !fops1 && !fops2 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier write.write_f;
@@
// write fops use offset
struct file_operations fops = {
... .write = write_f, ...
+ .llseek = default_llseek, /* write accesses f_pos */
};
// Use noop_llseek if neither read nor write accesses f_pos
///////////////////////////////////////////////////////////
@ fops4 depends on !fops1 && !fops2 && !fops3 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier read_no_fpos.read_f;
identifier write_no_fpos.write_f;
@@
// write fops use offset
struct file_operations fops = {
...
.write = write_f,
.read = read_f,
...
+.llseek = noop_llseek, /* read and write both use no f_pos */
};
@ depends on has_write && !has_read && !fops1 && !fops2 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier write_no_fpos.write_f;
@@
struct file_operations fops = {
... .write = write_f, ...
+.llseek = noop_llseek, /* write uses no f_pos */
};
@ depends on has_read && !has_write && !fops1 && !fops2 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
identifier read_no_fpos.read_f;
@@
struct file_operations fops = {
... .read = read_f, ...
+.llseek = noop_llseek, /* read uses no f_pos */
};
@ depends on !has_read && !has_write && !fops1 && !fops2 && !has_llseek && !nonseekable1 && !nonseekable2 && !seq @
identifier fops0.fops;
@@
struct file_operations fops = {
...
+.llseek = noop_llseek, /* no read or write fn */
};
===== End semantic patch =====
- ASoC: Add max98088 CODEC driver
This patch adds the MAX98088 CODEC driver.
- ASoC: Pay attention to driver supplied DAI IDs
The driver can specify a DAI ID number so use that.
- ASoC: Check list debugfs files for PAGE_SIZE overflow
- ASoC: Move soc-core module init next to functon definition
- ASoC: don't register AC97 devices twice
With generic AC97 ASoC glue driver (codec/ac97.c), we get following warning when
the device is registered (slightly stripped the backtrace):
kobject (c5a863e8): tried to init an initialized object, something is seriously
wrong.
[<c00254fc>] (unwind_backtrace+0x0/0xec)
[<c014fad0>] (kobject_init+0x38/0x70)
[<c0171e94>] (device_initialize+0x20/0x70)
[<c017267c>] (device_register+0xc/0x18)
[<bf20db70>] (snd_soc_instantiate_cards+0x924/0xacc [snd_soc_core])
[<bf20e0d0>] (snd_soc_register_platform+0x16c/0x198 [snd_soc_core])
[<c0175304>] (platform_drv_probe+0x18/0x1c)
[<c0174454>] (driver_probe_device+0xb0/0x16c)
[<c017456c>] (__driver_attach+0x5c/0x7c)
[<c0173cec>] (bus_for_each_dev+0x48/0x78)
[<c0173600>] (bus_add_driver+0x98/0x214)
[<c0174834>] (driver_register+0xa4/0x130)
[<c001f410>] (do_one_initcall+0xd0/0x1a4)
[<c0062ddc>] (sys_init_module+0x12b0/0x1454)
This happens because the generic AC97 glue driver creates its codec->ac97 via
calling snd_ac97_mixer(). snd_ac97_mixer() provides own version of
snd_device.register which handles the device registration when
snd_card_register() is called.
To avoid registering the AC97 device twice, we add a new flag to the
snd_soc_codec: ac97_created which tells whether the AC97 device was created by
SoC subsystem.
- ASoC: Use delayed work for debounce of GPIO based jacks
Rather than block the workqueue by sleeping to do the debounce use delayed
work to implement the debounce time. This should also means that we extend
the debounce time on each new bounce, potentially allowing shorter debounce
times for clean insertions.
- ASoC: soc-cache: Add spi_write support for all I/O types
Ensure that all drivers that use SPI and I2C will work properly
by providing SPI write functions for all different I/O types.
- ASoC: WM8804: Initial driver
The WM8804 is a high performance consumer mode S/PDIF transceiver with
support for 1 received channel and 1 transmitted channel.
- ASoC: Fix incorrect parameter to snd_soc_codec_volatile_register
We need to pass the register index and not the register value.
This patch depends on my previous patch "ASoC: Delegate to hw
specific read for volatile registers".
- ASoC: Delegate to hw specific read for volatile registers
Ensure that reads on volatile registers will correctly delegate
to the bus specific read function.
- ASoC: adapt multi-componentism again
Go back to the new world order.
(Also fixed indentation.)
- ASoC: Return -1 instead of -EINVAL to ensure consistency
The code can't really cope with I/O errors, so it would be better
to be consistent throughout all cache functions and return -1 instead
of -EINVAL.
The return value of snd_soc_read(...) is mostly checked in the probe
function and nowhere else.
- ASoC: Report error codes for card DAI instantiation failures
Also clean up the error print a bit.
- ASoC: Remove extra rtd->dev.init_name assignment in soc_probe_dai_link
rtd->dev.init_name is set twice in soc_probe_dai_link. I removed the first
assignement from dai_link->stream_name since then there won't be sysfs name
changes and usually dai_link->name seems to fit anyway better for a sysfs
directory name.
- ASoC: Report error code when failing to add controls
Helps with diagnostics.
- ASoC: Add platform listing to debugfs
List registered platforms in debugfs to improve debugability of machine
drivers.
- ASoC: Add DAI list to debugfs
Allow the user to inspect the list of registered DAIs at runtime to
improve diagnostics for machine driver setup.
- ASoC: Add debugfs listing of registered CODECs
Help with diagnostics for machine driver setup by listing all the
registered CODECs in debugfs.
- ASoC: WM8985: Initial driver
The WM8985 is a low power, high quality, feature-rich stereo
CODEC designed for portable multimedia applications that
require low power consumption and high quality audio.
- ASoC: Swap bias level enumeration
Swapping the bias level enumeration is only meant to help debugging. It is
easier if number 0 means bias off and bigger number means bigger bias level.
- ASoC: Add build infrastructure for WL1273
The Makefile and Kconfig updates for WL1273 appear to have been mising
from the patch posted, add them.
- ASoC: Don't call DAI registration for CODECs with no DAI
Otherwise we generate worrying (but benign) warnings for amps.
- ASoC: add 88pm860x codec driver
Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it
also supports headset/mic/hook/short detection.
- ASoC: core - fix build warning on x86_64
Output size_t type as a "%Zu" to avoid warnings.
- ASoC: Remove redundant device name from debugfs directory
Since the core now includes deduplication in the name of CODEC
devices there's no need to add extra for the debugfs directory name.
- ASoC: Set up debugfs only once per CODEC
Since the debugfs directory is current per CODEC we should only init
it when the CODEC is initialised, otherwise we end up with errors
being generated when an attempt is made to add duplicate debugfs
entries.
Since most of this stuff is actually for the card we should refactor
but this can come later.
- ASoC: Automatically determine control_data for soc-cache users
Since the provision of a struct device for the CODEC is now mandatory
we can use container_of() to locate the struct i2c_client and struct
spi_device for relevant devices, removing the need to manually set it
in each driver.
A further patch will automate selection of the control type based on
the bus_type of the struct device, further reducing the amount of
driver code required.
- ASoC: multi-component - ASoC Multi-Component Support
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Some TLV320aic23 and Cirrus platform fixes.
TI CODEC and OMAP fixes
Samsung platform and misc fixes :-
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
MPC8610 and PPC fixes.
i.MX fixes and some core fixes.
J4740 platform fixes:-
- ASoC: Add initial WM8962 CODEC driver
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
- ASoC: Fix multi-componentism
Spot the build testing.
- ASoC: Fix soc-cache buffer overflow bug
Make sure we stay within the cache boundaries when updating the
register cache.
- ASoC: soc-core: fix debugfs_pop_time file permissions
I think this is a typo, debugfs_pop_time should not be executable.
- ASoC: register cache should be 1 byte aligned for 1 byte long register
- ASoC: Initial WM8741 CODEC driver
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz. The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.
TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.
- fix typos concerning "initiali[zs]e"
- ASoC: Select wm_hubs automatically for WM8994
Otherwise all machine drivers need to do so.
- ASoC: Handle read failures in codec_reg
When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.
- ASoC: Fix sorting of Makefile and Kconfig
- ASoC: Add JZ4740 codec driver
This patch adds support for the JZ4740 internal codec.
- ASoC: Add JZ4740 ASoC support
This patch adds ASoC support for JZ4740 SoCs I2S module.
- ASoC: Fix sorting of DA7210 entries in Kconfig
- ASoC: Fix overflow bug in SOC_DOUBLE_R_SX_TLV
When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.
The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.
Solution: Mask the result so that it "wraps around", emulating
sign-extension.
- ASoC: Pay attention to write errors in volsw_2r_sx
- ASoC: ep93xx i2s audio driver
Add ep93xx i2s audio driver
- ASoC: kirkwood: Add i2s support
This patch enables support for the i2s controller available on kirkwood
platforms
- ASoC: Add driver for cs42l51
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.
- ASoC: NUC900/audio: add nuc900 audio driver support
Add support for NUC900 AC97
- ASoC: Add SOC_DOUBLE_R_SX_TLV control
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
- ASoC: core: Fix for the volume limiting when invert is in use
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.
Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
- ASoC: Allow DAI links to be kept active over suspend
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: Support leaving paths enabled over system suspend
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: Refactor DAPM suspend handling
Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.
Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
- ASoC: core: Support for limiting the volume
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)
If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:
snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
- ASoC: Add debug output tracing all cache register writes
- ASoC: Add WM9090 amplifier driver
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
- ASoC: Allow reporting of NULL jacks
Follow the core jack implementation and allow reporting on the status
of NULL jacks, avoiding the need to check in detection implementations.
- ASoC: Add indirection for CODEC private data
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

SoC Marvell Kirkwood

- Add kirkwood and nuc900 build stubs
- ASoC: Add support for OpenRD Ultimate
OpenRD Ultimate & Client are similar machines so enable OpenRD client sound
support on Ultimate too
Tested-by: Robas Teodor <teodor.robas@gmail.com>
- ASoC: kirkwood: Add audio support to hp t5325 thin clients
This patch is adding support for hp t5325 thin clients.
There's a alc5623 codec connected to the i2s interface.
- ASoC: Change my mail address
Like other coworkers, I'm about leave Mandriva/Edge-It so I'm changing
my mail address to use my personal one.
- ASoC: kirkwood: add alias to pcm module
Allow snd-soc-kirkwood autoloading by adding an alias.
- ASoC: OpenRD Client : Fix naming breakage due to multicomponent support
multicomponent support added/changed some device name but added some typos,
breaking existing OpenRD Client support.
- ASoC: kirkwood-i2s: Handle mute/unmute playback/record
The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.
- ASoC: Fix kirkwood i2s mono playback
Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.
- ASoC: patch for the useless 'break' removal in kirkwood
This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.
- ASoC: kirkwood: Add audio support to openrd client platforms
This patch is adding support for openrd client platforms. It's using
the cs42l51 codec and has one mic and one speaker plugs.
- ASoC: kirkwood: Add i2s support
This patch enables support for the i2s controller available on kirkwood
platforms

SoC NVIDIA Tegra

- add missing files for soc/mid-x86 and soc/tegra
- Add build stubs for tegra and mid-x86 ASoC drivers
- ASoC: tegra: Add DAPM widgets/routes for Harmony
With this change, I can capture from a microphone plugged into the
mic jack on Harmony (after unmuting Left Input PGA, and maybe turning
up the gain there too).
- ASoC: tegra: Harmony machine support
- ASoC: tegra: Add tegra-pcm driver
This provides an ASoC platform driver that manages Tegra's APB DMA
controller.
- ASoC: tegra: s/IS_ERR_OR_NULL/IS_ERR/ for clk_get_sys
A recent discussion on linux-arm-kernel noted that the value returned by
clk_get_sys is an opaque token, and not strictly a pointer; it is
meaningful only to the clock API, clients should not dereference the value,
and the clock API must accept any non-IS_ERR value it returned.
Hence, only IS_ERR is appropriate to interpret the result, not
IS_ERR_OR_NULL.
I checked that clk_get_sys in both ASoC's for-next and Tegra's for-next
do behave as described; NULL is not returned in the case of error.
- ASoC: tegra: Machine utility code
Many portions of Tegra ASoC machine drivers will be similar or identical.
To avoid cut/paste, this file will act as a repository for all that common
code. For now, it solely includes code to reprogram the audio PLL for
44.1KHz- vs. 48KHz-based sample rates.
- ASoC: Update name of debugfs root symbol to snd_soc_
Everything else is using snd_soc_ so we should use it here too.
- ASoC: tegra: Add tegra-i2s driver
This provides an ASoC DAI interface for Tegra's I2S controller.
- ASoC: tegra: Add tegra-das driver
The DAS (Digital Audio Switch) is a mux/crossbar which sits between
the DACs (Digital Audio Controllers) and the DAPs (Digital Audio
Ports). Audio data may be routed between DACs and DAPs in various
combinations as required by board design/application.
- ASoC: tegra: Remove TEGRA_I2S_AUDIO from Kconfig
That config variable doesn't exist in the mainline kernel, and hence
the dependency shouldn't either.
In linux-tegra-2.6.36, the dependency did exist to avoid a conflict with
the old non-ALSA Tegra I2S driver. However, this isn't and won't be
upstreamed.
- ASoC: tegra: Kconfig and Makefile

SoC Nuvoton NUC900

- Add kirkwood and nuc900 build stubs
- ASoC: Fix missing spin_unlock_irqrestore
In nuc900_dma_hw_params(), if snd_pcm_lib_malloc_pages failed
it returns without calling spin_unlock_irqrestore().
Since snd_pcm_lib_malloc_pages() does not touch struct nuc900_audio,
we don't need to hold the lock while calling snd_pcm_lib_malloc_pages().
Fix it by moving spin_lock_irqsave() down to after snd_pcm_lib_malloc_pages().
In nuc900_dma_prepare(), spin_unlock_irqrestore() is missing in the error path.
Fix it by removing the return in default case.
- ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
I think this unneededd !! operations just reduce the readability.
- ASoC: Fix compile error for nuc900-pcm.c
This patch fixes below error:
CC sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_open':
sound/soc/nuc900/nuc900-pcm.c:267: error: 'nuc900_ac97_data' undeclared (first use in this function)
sound/soc/nuc900/nuc900-pcm.c:267: error: (Each undeclared identifier is reported only once
sound/soc/nuc900/nuc900-pcm.c:267: error: for each function it appears in.)
sound/soc/nuc900/nuc900-pcm.c: At top level:
sound/soc/nuc900/nuc900-pcm.c:337: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-pcm.c:354: error: 'nuc900_soc_platform_probe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-pcm.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
This patch fixes below compile warning:
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
- ASoC: Fix compile error for nuc900-ac97.c
Fix below compile error by add a missing ';'.
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:318: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-ac97.c:405: error: 'nuc900_ac97_drvprobe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-ac97.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: nuc900-ac97: fix a memory leak
- ASoC: remove duplicated include for nuc900
Remove duplicated include.
- ASoC: NUC900: patch for fix build error
This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.
- ASoC: nuc900: patch for modifing the ac97 delays to minimum
This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.
I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.
- ASoC: nuc900: fix a typo and rename the header file
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.
- ASoC: nuc900: fix a wait loop bug
The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.
We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.
- ASoC: nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.
There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.
- ASoC: NUC900/audio: add nuc900 audio driver support
Add support for NUC900 AC97

SoC PXA2xx 88PM860x Tavor EVB3

- ASoC: add tavorevb3 machine driver for 88pm860x

SoC PXA2xx Aeronix Zipit Z2

- ASoC: PXA: Fix codec address on Zipit Z2
WM8750 address is 0x1b, not 0x1a. Without this fix ALSA detects no sound
cards on Zipit
- ASoC: PXA: Fix jack detection on Zipit Z2
Fix jack detection on Zipit Z2, otherwise it
disables headphones output when jack is connected

SoC PXA2xx Corgi

- ASoC: correct link specifications for corgi, poodle and spitz
ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.
- ASoC: Lock the CODEC in PXA external jack controls
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
- ASoC: Fix a few more PXA build errors
Dead pxa2xx-pcm.h includes and a missing ,

SoC PXA2xx E740

- ASoC: e740_wm9705 - free gpio in e740_exit()
In e740_init(), we call gpio_request() for
GPIO_E740_MIC_ON, GPIO_E740_AMP_ON and GPIO_E740_WM9705_nAVDD2.
We should free the these gpio accordingly in e740_exit().

SoC PXA2xx EM-X270

- ASoC: Fix a few more PXA build errors
Dead pxa2xx-pcm.h includes and a missing ,

SoC PXA2xx Poodle

- ASoC: correct link specifications for corgi, poodle and spitz
ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.
- ASoC: Lock the CODEC in PXA external jack controls
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
- ASoC: pxa2xx-i2s is the proper name of the I2S DAI, not pxa-i2s.
- ASoC: Fix a few more PXA build errors
Dead pxa2xx-pcm.h includes and a missing ,

SoC PXA2xx Spitz

- ASoC: correct link specifications for corgi, poodle and spitz
ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.
- ASoC: Lock the CODEC in PXA external jack controls
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
- ALSA: Fix Linux 2.6 merge issues
- [ARM] pxa/spitz: Correctly register WM8750
This patch registers the WM8750 codec on a proper place on the SPITZ machine
after the WM8750 driver was converted to new API.

SoC PXA2xx Tosa

- ASoC: fix the building issue of missing codec field in 'struct snd_soc_card'

SoC PXA2xx saarb

- ASoC: add saarb machine driver for 88pm860x
88PM860x codec is used in Marvell saarb development board. 88PM860x codec
is used as master mode for SSP communication. Only I2S format is supported.

SoC S6000

- ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
- s6105-ipcam: fix compilation
When the s6105-ipcam ASoC driver had been converted to the
multi-component API, a single reference to a former structure
element remained, blocking successful compilation.
- s6000-pcm: fix compilation
s6000_soc_platform has lost its forward declaration and there no
longer is a name element in it, so use a string constant when
calling request_irq.
- s6000-i2s: fix compilation
A semicolon was missing.
- ASoC: s6000: use resource_size for {request/release}_mem_region and ioremap
The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.

SoC SH7760 AC97

- ASoC: sh: fsi-ak4642: fixup platform device id
- ASoC: fix migor audio build
Commit 6d803ba736abb5e122dede70a4720e4843dd6df4 "ARM: 6483/1: arm & sh:
factorised duplicated clkdev.c" broke compilation of migor audio. Use the
correct header to fix the problem.
- ASoC: sh: fsi-da7210: remove unnecessary format settings
- ASoC: sh: fsi: remove runtime register check from fsi_master_xxx
Current FSI driver was checking register range on fsi_master_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
- ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk
There was a strange part where fsi->xxx had been used
for fsi_master_xxx function instead of master->xxx in current FSI.
This patch modify it.
- ASoC: sh: fsi: modify improper dependent
FSI-AK4642 and FSI-DA7210 are depend on I2C, not I2C_SH_MOBILE
- ASoC: sh: fsi: Add over/under run counter
Current FSI driver had printed under/over run error
if status register have its error bit.
But runtime print cause the next error
because print out is slow.
This patch add error counter and print error when sound stop
- ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger
- ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx
Current FSI driver was checking register range on fsi_reg_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
- ASoC: sh: fsi: remove fsi_master_write
- ASoC: sh: fsi: clean up SPDIF defines
- ASoC: sh: fsi-ak4642: midify card name
- ASoC: sh: fsi-ak4642: tidyup unnecessary variables
- ASoC: sh: fsi-ak4642: Add FSI port and ak464x selection
Current FSI-Ak4642 device had niche settings which were
FSI2-A-AK4643 and FSI-A-AK4642.
This patch add platform_device_id which can control
FSI/FSI2, PortA/PortB, AK4642/AK4643 from platform data.
- ASoC: Do not include soc-dai.h
There is no need to include soc-dai.h since soc.h includes it. Convert
drivers to include only soc.h.
- ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove
We call snd_soc_register_dais() in sh4_soc_dai_probe(),
thus we should call snd_soc_unregister_dais() in sh4_soc_dai_remove().
Otherwise, we got "too many arguments to function 'snd_soc_unregister_dai'"
error message.
- ARM: mach-shmobile: ap4evb: FSI clock use proper process for ak4642
Current AP4 FSI didn't use set_rate for ak4642,
and used dummy rate when init.
And FSI driver was modified to always call set_rate.
The user which are using FSI set_rate is only AP4 now.
- ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMI
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.
- sound: sh: ctrl_in/outX to __raw_read/writeX conversion.
The ctrl_xxx routines are deprecated, switch over to the __raw_xxx
versions.
- ASoC: fsi: simultaneous playback/recorde support
Current FSI driver had not cared about simultaneous
playback/capture on same port.
This patch add new fsi_stream struct to care it,
- ASoC: Add fsi_is_play function
- ASoC: fsi: Add new macro and shift for PortA/B In/Out
Some FSI register have similar bit array for PortA/B and In/Out.
This patch add new macro and shift for it
- ASoC: fsi: avoid un-necessary status read
- ASoC: fsi: remove un-necessary variable from fsi_dai_startup
- ASoC: fsi: Add fsi_get_frame_width function
It is not so important for now.
But will be used in future.
- sound/soc/sh/siu: Fixed undefined dma_length of scatter gather list
Without this patch, an undefined/random sg->dma_length is used and
the sound will be played/captured wrongly.
- ASoC: fsi-hdmi: remove unneeded header
This patch solve below report from Guennadi.
But I didn't remove #include <sound/sh_fsi.h>.
Because it have FSI_PORT_B define which is used on this file.
> +#include <linux/platform_device.h>
> +#include <sound/sh_fsi.h>
> +#include <video/sh_mobile_hdmi.h>
Now that everything is done with strings - do you still need these
headers?
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- ASoC: sh: fix build error: terminate the platform device ID list
Platform driver ID table must be zero-element terminated.
- ASoC: fsi: merge fsi_data_push/pop to fsi_fifo_data_ctrl
Current FSI driver had data push/pop functions.
But the main operation of these 2 were very similar.
This mean it is possible to merge these to 1 function.
- ASoC: fsi: modify variable name to easy to understand
Current FSI driver is using
data-length / width / number / offset for variables.
But it was a very confusing name.
This patch rename them to easy to understand,
and add new functions for it.
- ASoC: Fix WM8978/migor driver name conflict
Standardise on 'wm8978' as the name for the CODEC.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- ASoC: fsi: Add fsi_dma_soft_push/pop function
- ASoC: fsi: modify noisy comment out
- ASoC: fsi: Add fsi_dma_get_area
- ASoC: fix SIU driver breakage, occurred during the multi-component transition
This patch fixes multiple bugs and a typo, occurred during the multi-
component transition.
- ASoC: fix compile breakage of the sh/siu driver
The sh/siu ASoC driver doesn't compile because of a function defined static in
the source and extern in a header. Remove the unneeded declaration in the
header.
- ASoC: Add device table to SH FSI driver
[Modified to move the location of the table]
- ASoC: fsi codecs: Update card name field
- ASoC: fsi codecs: modify menu attribute on Kconfig
Current SND_FSI_xxx menu attributes were bool,
but it should be tristate.
This patch solve below report from Guennadi
"bool" means, if someone is linking the whole ASoC into the kernel, they
will not be able to build this as a module. Not a big deal, but you're
stealing some freedom from the user.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- ASoC: fsi-ak4642: modify platform_name
- ASoC: fsi-codec: Add FSI - HDMI support
- ASoC: fsi-ak4642/fsi-da7210: modify dai link settings for card detect
This patch modify dai link
- platform_name: sh_fsi/sh_fsi2 are used for FSI driver
- codec_name: ak4642/ak4643 are used for ak4642 driver
This is quick hack. I should modify it more wisely in future
- ASoC: fix clkdev API usage in sh/migor.c
The clkdev API doesn't use .name and .id members of struct clk for clock
lookup. Instead clocks should be added to a lookup list. Without this patch
audio om the Migo-R board fails silently.
- ASoC: fsi: Add new funtion for SPDIF
- ASoC: fsi: remove device id check
Current FSI driver id is not only 0
- ASoC: fsi: remove unnecessary clock processing
- ASoC: fsi: Add specified ID for soc-audio
Specified ID is necessary, when some codecs are used with FSI.
- ASoC: fsi: Fixup for master mode
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
- ASoC: fsi: Add pr_err for noticing unsupported access
This patch didn't use dev_err,
because it is difficult to get struct device here.
- ASoC: fsi: Change struct fsi_regs to fsi_core
Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.
- ASoC: fsi: remove noisy CR_FMT macro
- ASoC: fsi: remove un-used variable on fsi_dai_startup
- ASoC: fsi: fixup wrong value setting order of TDM
channel size should be set before setting register value
- ASoC: fsi: fixup clock inversion operation
Clock inversion should be specified by each flags bit.
- ASoC: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.
Consequently when I2C is not set, the compilation fails [1]
This patch fixes this issues, by adding a depencdency on the related HW-
controller.
- ASoC: header cleanup for FSI-DA7210
- ASoC: header cleanup for FSI-AK4642
- ASoC: header cleanup for FSI
- ASoC: fix uninitialised variable in siu_dai.c
- sh: define DMA slaves per CPU type, remove now redundant header
Now that DMA slave IDs are only used used in platform specific code and have
become opaque cookies for the rest of the code, we can make the, CPU specific
too.

SoC Texas Instruments OMAP

- ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
- ASoC: mcbsp: Add McBSP support for OMAP4
This patch adds McBSP support for the OMAP4 CPU
- ASoC: sdp4430: Add Jack support
Use jack framework to enable detection for the headset microphone
and stereo output in the sdp4430.
- ASoC: sdp4430: Enable FM stereo pins
Add FM stereo pins to the machine driver and add them as a
dapm widget.
- ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.
Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.
Thanks to Tony Lindgren <tony@atomide.com> for noticing.
- ASoC: Fix resource reclaim for osk5912
In current implementation, there are resources leak in the error path.
This patch properly reclaims the allocated resources in the error path.
Also adds a missing clk_put in osk_soc_exit.
- ASoC: Return proper error for omap3pandora_soc_init
Return PTR_ERR(omap3pandora_dac_reg) instead of 0 if regulator_get failed.
- ASoC: OMAP: fix OMAP1 compilation problem
In the new code introduced with commit cf4c87abe238ec17cd0255b4e21abd949d7f811e,
"OMAP: McBSP: implement McBSP CLKR and FSR signal muxing via mach-omap2/mcbsp.c",
the way omap1 build is supposed to bypass omap2 specific functionality doesn't
optimize out all omap2 specific stuff. This breaks linking phase for omap1
machines, giving "undefined reference to `omap2_mcbsp1_mux_clkr_src'"
and "undefined reference to `omap2_mcbsp1_mux_fsr_src'" errors. Fix it.
Created and tested against linux-2.6.37-rc1.
- OMAP: McBSP: implement functional clock switching via clock framework
Previously the OMAP McBSP ASoC driver implemented CLKS switching by
using omap_ctrl_{read,write}l() directly. This is against policy; the OMAP
System Control Module functions are not intended to be exported to drivers.
These symbols are no longer exported, so as a result, the OMAP McBSP ASoC
driver does not build as a module.
Resolve the CLKS clock changing portion of this problem by creating a
clock parent changing function that lives in
arch/arm/mach-omap2/mcbsp.c, and modify the ASoC driver to use it.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, this symbol must be exported for use by
sound/soc/omap/omap-mcbsp.c.
Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/* and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are. These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversions.
- OMAP: McBSP: implement McBSP CLKR and FSR signal muxing via mach-omap2/mcbsp.c
The OMAP ASoC McBSP code implemented CLKR and FSR signal muxing via
direct System Control Module writes on OMAP2+. This required the
omap_ctrl_{read,write}l() functions to be exported, which is against
policy: the only code that should call those functions directly is
OMAP core code, not device drivers. omap_ctrl_{read,write}*() are no
longer exported, so the driver no longer builds as a module.
Fix the pinmuxing part of the problem by removing calls to
omap_ctrl_{read,write}l() from the OMAP ASoC McBSP code and
implementing signal muxing functions in arch/arm/mach-omap2/mcbsp.c.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, these symbols must be exported for use by
sound/soc/omap/omap-mcbsp.c.
Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/*, and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are. These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversion.
- ASoC: OMAP4: MCPDM: Remove unnecessary include of plat/control.h
Commit 346a5c890 (OMAP: control: move plat-omap/control.h
to mach-omap2/control.h) in the linux-omap tree removed
plat/control.h and most of its callers. This one slipped
through - breaking the build as below when
CONFIG_SND_OMAP_SOC_MCPDM is defined. Fix this.
CC sound/soc/omap/omap-mcpdm.o
sound/soc/omap/omap-mcpdm.c:35: fatal error: plat/control.h: No such file or directory
compilation terminated.
make[3]: *** [sound/soc/omap/omap-mcpdm.o] Error 1
make[2]: *** [sound/soc/omap] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: omap: Remove needless prints from machine drivers
It is currently completely normal to execute these machine drivers code on
different boards if the kernel includes support for multiple boards so no
error message should be printed if the machine_is_xxx does not match with
the machine driver.
Therefore remove these pr_err and pr_debug prints in those cases.
- sound: OMAP: McBSP: Fix static function warning
This patch fixes sparse warning due non declaration of static function
sound/soc/omap/omap-mcbsp.c:783:5: warning: symbol 'omap_mcbsp_st_info_volsw' was not declared. Should it be static?
- ASoC: omap: rx51: Use gpio_set_value_cansleep for speaker amp control
Speaker amplifier is controlled by TWL4030 GPIO which may sleep. Therefore
use gpio_set_value_cansleep to get rid of runtime warning that is introduced
after the commit 9c4ba94 and to get a stack trace if ever executing this
code in atomic context.
- ASoC: Use a more adequate name for the CX20442 codec DAI
In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.
Use a better name, "cx20442-voice", as suggested by Liam Girdwood.
- ASoC: omap-mcbsp: Remove period size constraint in THRESHOLD mode
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.
Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)
- ASoC: omap-mcbsp: Support for sDMA packet mode
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.
The patch extends the original THRESHOLD mode by doing the
following:
if (period_words <= max_threshold)
Current THRESHOLD mode configuration
Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size
With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.
- ASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_params
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.
- ASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_params
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.
- ASoC: RX-51: Add basic jack detection
This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only
SND_JACK_VIDEOOUT type is reported. More types could be reported after
getting more audio features supported and necessary drivers integrated for
implementing automated accessory detection.
- ASoC: RX-51: Add Jack Function kcontrol
Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used
as headphone, headset or audio-video connector. This patch implements the
control 'Jack Function' which is used to select the desired function.
At the moment only TV-out without audio is supported.
- ASoC: pandora: fix CLKX polarity
After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.
Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
- ASoC: omap-mcbsp: Place correct constraints for streams
OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer
This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
- ASoC: omap-mcbsp: Save, and use wlen for threshold configuration
Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
- ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
Fix build warning about unused ops and add ops
to the sdp4430 DAI link.
- ASoC: SDP4430: Add support for Earphone speaker
Enable earphone speaker in sdp4430 machine driver.
- ASoC: SDP4430: Add sdp4430 machine driver
Add ASoC support for TI SDP4430.
- OMAP: McBSP: Add 32-bit mode support
This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.
- ASoC: omap: Add basic audio support for Nokia RX-51/N900
This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html
- ASoC: zoom2 - update DAPM pins
Remove bogus twl4030 pins
- ASoC: pandora - update DAPM pins
Remove bogus TWL4030 pins.

Soc PXA2xx Imote 2

- ASoC: pxa2xx-i2s is the proper name of the I2S DAI, not pxa-i2s.

Soc PXA2xx Magician

- ASoC: Lock the CODEC in PXA external jack controls
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
- ASoC: pxa2xx-i2s is the proper name of the I2S DAI, not pxa-i2s.
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