Detailed SoC changes v1.0.24 v1.0.25

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Detailed SoC changelog between 1.0.17 and 1.0.25 releases

alsa-driver

SoC PXA2xx Core

- ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700
SND_SOC_RAUMFELD selects SND_SOC_CS4270 which needs CONFIG_I2C,
and also selects SND_SOC_AK4104 which needs SPI_MASTER.
Thus make SND_SOC_RAUMFELD depend on I2C && SPI_MASTER.
Add depend on SPI_MASTER to fix below build error if CONFIG_SPI_MASTER
is not selected.
LD .tmp_vmlinux1
sound/built-in.o: In function `ak4104_spi_write':
last.c:(.text+0x290cc): undefined reference to `spi_sync'
sound/built-in.o: In function `ak4104_probe':
last.c:(.text+0x292a0): undefined reference to `spi_write_then_read'
sound/built-in.o: In function `ak4104_spi_probe':
last.c:(.text+0x29398): undefined reference to `spi_setup'
sound/built-in.o: In function `ak4104_init':
last.c:(.init.text+0x4ec): undefined reference to `spi_register_driver'
make: *** [.tmp_vmlinux1] Error 1
Add depend on I2C to fix below build error if CONFIG_I2C is not selected:
CC sound/soc/codecs/cs4270.o
sound/soc/codecs/cs4270.c: In function 'cs4270_i2c_probe':
sound/soc/codecs/cs4270.c:657: error: implicit declaration of function 'i2c_smbus_read_byte_data'
sound/soc/codecs/cs4270.c: In function 'cs4270_init':
sound/soc/codecs/cs4270.c:730: error: implicit declaration of function 'i2c_add_driver'
sound/soc/codecs/cs4270.c: In function 'cs4270_exit':
sound/soc/codecs/cs4270.c:736: error: implicit declaration of function 'i2c_del_driver'
make[3]: *** [sound/soc/codecs/cs4270.o] Error 1
make[2]: *** [sound/soc/codecs] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
SND_PXA2XX_SOC_HX4700 selects SND_SOC_AK4641 which needs CONFIG_I2C.
Thus make SND_PXA2XX_SOC_HX4700 depend on I2C.
Add depend on I2C to fix below build error if CONFIG_I2C is not selected:
CC sound/soc/codecs/ak4641.o
sound/soc/codecs/ak4641.c: In function 'ak4641_modinit':
sound/soc/codecs/ak4641.c:646: error: implicit declaration of function 'i2c_add_driver'
sound/soc/codecs/ak4641.c: In function 'ak4641_exit':
sound/soc/codecs/ak4641.c:656: error: implicit declaration of function 'i2c_del_driver'
make[3]: *** [sound/soc/codecs/ak4641.o] Error 1
make[2]: *** [sound/soc/codecs] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X
In saarb_pm860x_init() and evb3_pm860x_init(), we call
pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn
calls pm860x_set_bits().
Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X.
This patch fixes below build error if CONFIG_MFD_88PM860X is not configured.
LD .tmp_vmlinux1
sound/built-in.o: In function `pm860x_write_reg_cache':
last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write'
sound/built-in.o: In function `pm860x_set_bias_level':
last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits'
last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write'
last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write'
sound/built-in.o: In function `pm860x_read_reg_cache':
last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read'
sound/built-in.o: In function `pm860x_probe':
last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read'
sound/built-in.o: In function `pm860x_codec_handler':
last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read'
last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read'
sound/built-in.o: In function `pm860x_mic_jack_detect':
last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits'
sound/built-in.o: In function `pm860x_hs_jack_detect':
last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits'
last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits'
last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits'
make: *** [.tmp_vmlinux1] Error 1
- ASoC: pxa2xx-pcm: remove unused variable 'dai'
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'
- ASoC: pxa-ssp: Correct check for stream presence
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
- ASoC: add iPAQ hx4700 machine driver
AK4641 connected via I2S and I2C, jack detection via GPIO.
- ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
pxa2xx_pcm_hw_free frees dma channel and sets prtd->dma_ch to -1,
but does not set prtd->params to NULL, so if pxa2xx_pcm_hw_params will
be called immediately, it leaves prtd->dma_ch initialized with -1,
and it results in oops in __pxa2xx_pcm_prepare. This bug is triggered
via SDL.
This patch adds check for prtd->dma_ch to __pxa2xx_pcm_prepare and
cleans prtd->params, so now it works properly.

SoC Audio for Freecale i.MX1x i.MX2x CPUs

- ASoC: Fix DMA channel leak in imx-pcm-dma-mx2 driver.
"snd_imx_pcm_hw_params" callback can be called
several times by the user (i.e. OSS emulation)
leading to a DMA channel leak.
- ASoC: imx: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Add missed MODULE_LICENSE("GPL") for imx-pcm-fiq
This driver can be built as module and the file header indicates that
the driver is published under the GPL.
Thus add MODULE_LICENSE("GPL") for it.
- ASoC: Convert imx directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C
SND_SOC_MX27VIS_AIC32X4 selects SND_SOC_TLV320AIC32X4,
but SND_SOC_TLV320AIC32X4 needs CONFIG_I2C.
So we need to make SND_SOC_MX27VIS_AIC32X4 depend on I2C.
otherwise I got below build error if CONFIG_I2C is not selected.
CC sound/soc/codecs/tlv320aic32x4.o
sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_read':
sound/soc/codecs/tlv320aic32x4.c:323: error: implicit declaration of function 'i2c_smbus_read_byte_data'
sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_probe':
sound/soc/codecs/tlv320aic32x4.c:641: error: 'i2c_master_send' undeclared (first use in this function)
sound/soc/codecs/tlv320aic32x4.c:641: error: (Each undeclared identifier is reported only once
sound/soc/codecs/tlv320aic32x4.c:641: error: for each function it appears in.)
sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_modinit':
sound/soc/codecs/tlv320aic32x4.c:763: error: implicit declaration of function 'i2c_add_driver'
sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_exit':
sound/soc/codecs/tlv320aic32x4.c:774: error: implicit declaration of function 'i2c_del_driver'
make[3]: *** [sound/soc/codecs/tlv320aic32x4.o] Error 1
make[2]: *** [sound/soc/codecs] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: imx: Remove unused variable 'dai'
- ASoC: imx: Fix build warning of unused 'card' variable
Fixes the following warning:
CC sound/soc/imx/imx-pcm-fiq.o
sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new':
sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card'
CC sound/soc/imx/imx-pcm-dma-mx2.o
- ASoC: Remove unused function declaration in imx-ssi.h
These functions are removed in commit f0fba2ad
"ASoC: multi-component - ASoC Multi-Component Support".
Let's remove the leftover function declaration in header file.
- ASoC: imx: eukrea_tlv320 needs i2c
Add a missing dependency that is required for random configurations.
- ASoC: imx: use more robust checking of available streams
Replace the channels_min check with a check for the relevant substream
being present. Suggested here [1] when mxs implemented the
audio-support.
[1] http://www.spinics.net/lists/arm-kernel/msg133010.html
- ASoC: imx-ssi: use dma_writecombine consistently
If the channel is allocated as writecombine, then mmaping it should also
use writecombine. Also, add a proper device for the call. Ported from a
similar fix for mach-mxs.
- ARM i.MX dma: Fix burstsize settings
dmaengine expects the maxburst parameter in words, not bytes.
The imxdma driver and its users do this wrong. Fix this.
As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work
with imx-dma. This broke the driver with imx-sdma support which
correctly takes the maxburst parameter in words. This patch
puts the sdma based sound back to work.
- ASoC: imx: add missing module informations
- add some modules aliases
- add module license to avoid tainted kernel when loading the imx-pcm-audio
driver
- ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry
SND_MXC_SOC_SSI looks to be unused, so kill it.
- ASoC: imx: remove superfluous code in imx-ssi.c
Checking if IMX_SSI_DMA is set and then set it again is useless.
- ASoC: imx: fix burstsize for DMA
SSI counts in words, the DMA engine in bytes. (Wrong) factor got removed
in bf974a0 (ASoC i.MX: switch to new DMA api).
- ASoC: imx: set watermarks for mx2-dma
They got accidently removed by f0fba2a (ASoC: multi-component - ASoC
Multi-Component Support). Reintroduce them and get rid of the
superfluous defines because the fiq-driver has its own hardcoded values.
- ASoC: Add machine driver for Visstrim_M10 board.
Visstrim_M10 boards have an external tlcv320aic3205 codec
attached to SSI1. This driver glues together both interfaces.
External amplifier is not supported in this first version.
- ASoC: Fix burstsize and DSP_B format problems in imx-ssi.
When choosing IMX_DMA flag, burtsizes are set to its default
value (0) which leads to driver malfunction. Change them to 4.
DSP_B interface needs additional flag to match DSP_B formats
as described in several codecs as wm8741 and aic3205.
- ASoC: eukrea-tlv320: add MBIMXSD51 support
- eukrea-tlv320: fix platform_name
commit f0fba2ad1b6b53d5360125c41953b7afcd6deff0 included a mistake
on the name of the platform in the snd_soc_dai_link structure.

SoC Audio for TXx9

- ASoC: txx9: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert txx9 directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: txx9: Add __exit_p at necessary place
We have __exit annotation for txx9aclc_generic_remove(),
thus add __devexit_p to wrap it.
- ASoC: Fix txx9aclc.c build
552d1ef6b5a98d7b95959d5b139071e3c90cebf1 [ASoC: core - Optimise and refactor
pcm_new() to pass only rtd] breaks compilation of txx9aclc.c:
CC [M] sound/soc/txx9/txx9aclc.o
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_new':
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: error: 'card' undeclared (first use in this function)
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: note: each undeclared identifier is reported only once for each function it appears in
make[5]: *** [sound/soc/txx9/txx9aclc.o] Error 1
Fixed by providing a definition for card.

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

- ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC"
introduces below build warnings:
drivers/misc/Kconfig:212:error: recursive dependency detected!
drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC
sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731
sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC
SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation.
This patch remove the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above
warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC
need to depend on ATMEL_SSC.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: check for substream not channels_min in pcm engines
This is a follow up on 53dea36c70c1857 which fixes the other affected
pcm engines.
Description from 53dea36c70c1857:
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Without this patch I was seeing null-pointer dereferenc in atmel-pcm.
- ASoC: Fix build dependency for SND_ATMEL_SOC_SSC
Make SND_ATMEL_SOC_SSC select ATMEL_SSC to fix below build errors:
LD .tmp_vmlinux1
sound/built-in.o: In function `atmel_ssc_remove':
sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free'
sound/built-in.o: In function `atmel_ssc_probe':
sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request'
sound/built-in.o: In function `atmel_ssc_set_audio':
sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request'
sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free'
make: *** [.tmp_vmlinux1] Error 1
- ASoC: atmel: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert atmel directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Constify snd_soc_dai_ops structs
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.
The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>
- ASoC: drop support for PlayPaq with WM8510
SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b
("[ALSA] Revised AT32 ASoC Patch"). That support depends on
BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got
added in v2.6.27. It still doesn't. It has never been possible to even
build this driver. Drop it.
- ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
OMAP drivers are omitted as I know Peter already has patches for them.
- ASoC: playpaq_wm8510: Return proper error if clk_get fails
Return proper error instead of 0 if clk_get fails.
- sound: sound/atmel_ssc_dai: add a missing space to an error message
- ASoC: core - Optimise and refactor pcm_new() to pass only rtd
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
- ASoC: Add context parameter to card DAPM callbacks
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
- ASoC: atmel_ssc: Don't try to free ssc if request failed
We should only call ssc_free() when ssc_request() succeeds or bad
things will happen.
- ASoC: atmel_ssc_dai: fix ssc error path
We do not have to free a resource that is not allocated yet.
- ASoC: trivial: typo in atmel_pcm_dma_params strucutre comment
- ASoC: trivial: typo in debug comment
- ASoC: sam9g20_wm8731: use the proper SYSCKL value
at91sam9g20 is providing master clock to wm8731: not using a crystal but an
external MCLK. We can avoid conflict and save power using WM8731_SYSCLK_MCLK as
we do not need oscillator to be powered.
- ASoC: Remove -codec from WM8731 driver name

SoC Audio for the Samsung chips

- ASoC: Fix idma build after update for channel count check
- ASoC: Add trivial pm_runtime usage to Samsung DAI drivers
Currently this won't actually do anything but using this will help the
core SoC code track when the system is idle.
- ASoC: samsung: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Remove export of s3c_pcm_dai
We don't need to export s3c_pcm_dai after multi-component patch.
Thus remove export of s3c_pcm_dai and make it static.
- ASoC: Complete initialisation before registering Samsung PCM DAI
Otherwise there's a race where the DAI might get used without everything
having been set up.
- ASoC: Staticise asoc_idma_platform
- ASoC: Raise Speyside audio system clock rate to 512fs
To support advanced system functionality for additional components; the
actively used clocks will remain the same for current components. Also
factor the rate out to a single #define while we're at it.
- ASoC: Fix a typo in s3c24xx_simtec_tlv320aic23 driver
Fix a typo introduced by commit e00c3f55
"ASoC: Convert Samsung directory to module_platform_driver".
This fixes the build error:
CC sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_init':
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function)
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: (Each undeclared identifier is reported only once
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: for each function it appears in.)
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_exit':
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function)
make[3]: *** [sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
I think we had better naming it with *driver, thus I change
it to simtec_audio_tlv320aic23_driver.
- ASoC: Convert Samsung I2S driver to devm_kzalloc()
- ASoC: Make SND_SOC_LITTLEMILL select MFD_WM8994
SND_SOC_LITTLEMILL selects SND_SOC_WM8994, but SND_SOC_WM8994 needs MFD_WM8994.
Thus we need to select MFD_WM8994 to fix below build error:
LD .tmp_vmlinux1
sound/built-in.o: In function `wm8994_write':
sound/soc/codecs/wm8994.c:201: undefined reference to `wm8994_reg_write'
sound/built-in.o: In function `wm8994_read':
sound/soc/codecs/wm8994.c:222: undefined reference to `wm8994_reg_read'
sound/built-in.o: In function `wm8994_resume':
sound/soc/codecs/wm8994.c:2847: undefined reference to `wm8994_reg_read'
sound/built-in.o: In function `wm8994_codec_probe':
sound/soc/codecs/wm8994.c:3501: undefined reference to `wm8994_reg_read'
sound/soc/codecs/wm8994.c:3660: undefined reference to `wm8994_reg_read'
sound/soc/codecs/wm8994.c:3672: undefined reference to `wm8994_reg_read'
sound/built-in.o: In function `wm8958_dsp2_fw':
sound/soc/codecs/wm8958-dsp2.c:154: undefined reference to `wm8994_bulk_write'
make: *** [.tmp_vmlinux1] Error 1
- ASoC: Map microphones on Littlemill
Littlemill has one analogue microphone on the board (connected to IN1LN)
and an array of four DMICs connected to both DMICDAT lines. The biases
can be selected by jumpers but pick the default jumper fit.
- ASoC: Add WM8958 based headset detection on Littlemill
The board supports CODECs that won't work with this but the CODEC driver
will check to see if it's running on the right chip for us.
- ASoC: Rename Speyside WM8962 to Tobermory
All the other machine drivers for non-default configurations are named
after the relevant audio module so do so for Tobermory also.
- ASoC: Fix __iomem annotation for IDMA registers
We always store the register address as __iomem but pass it around as a
plain void * which upsets sparse.
- ASoC: Convert smdk_wm8994pcm to use module_platform_driver()
Use the module_platform_driver() macro which makes
the code smaller and a bit simpler.
- ASoC: Add basic 1277-EV1 Littlemill audio driver
The Littlemill audio card supports a number of pluggable miniboards,
normally for the WM8994 family of devices. As all these devices look
mostly the same from an external configuration point of view and are
runtime enumerable we can write a standard machine driver which will
work out of the box with any of them. Start doing that with the bare
bones of a driver, only supporting AIF1.
Future patches will flesh this out to be more fully featured.
- ASoC: Convert Samsung directory to module_platform_driver
Saves some boilerplate code.
- ASoC: Add fully_routed flag to Speyside machines
- ASoC: Add Lowland machine driver
The Lowland platform is based on the Cragganmore system like Speyside but
uses the WM5100 audio CODEC.
- ASoC: Include linux/module.h for smdk2443_wm9710
Include linux/module.h to fix below build error:
CC sound/soc/samsung/smdk2443_wm9710.o
sound/soc/samsung/smdk2443_wm9710.c:64: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/smdk2443_wm9710.c:64: warning: data definition has no type or storage class
sound/soc/samsung/smdk2443_wm9710.c:64: warning: type defaults to 'int' in declaration of 'MODULE_AUTHOR'
sound/soc/samsung/smdk2443_wm9710.c:64: warning: function declaration isn't a prototype
sound/soc/samsung/smdk2443_wm9710.c:65: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/smdk2443_wm9710.c:65: warning: data definition has no type or storage class
sound/soc/samsung/smdk2443_wm9710.c:65: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION'
sound/soc/samsung/smdk2443_wm9710.c:65: warning: function declaration isn't a prototype
sound/soc/samsung/smdk2443_wm9710.c:66: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/smdk2443_wm9710.c:66: warning: data definition has no type or storage class
sound/soc/samsung/smdk2443_wm9710.c:66: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE'
sound/soc/samsung/smdk2443_wm9710.c:66: warning: function declaration isn't a prototype
make[3]: *** [sound/soc/samsung/smdk2443_wm9710.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: Fix a typo in jive_wm8750
Fix a typo in jive_wm8750 that introduces below build error.
Also removes an unused err variable.
CC sound/soc/samsung/jive_wm8750.o
sound/soc/samsung/jive_wm8750.c: In function 'jive_wm8750_init':
sound/soc/samsung/jive_wm8750.c:104: warning: unused variable 'err'
sound/soc/samsung/jive_wm8750.c: At top level:
sound/soc/samsung/jive_wm8750.c:134: error: unknown field 'dapm_widgtets' specified in initializer
sound/soc/samsung/jive_wm8750.c:134: warning: initialization from incompatible pointer type
make[3]: *** [sound/soc/samsung/jive_wm8750.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: Correct name of Speyside Main Speaker widget
- ASoC: SAMSUNG: Fix build error
This patch adds <linux/modules.h> to fix following build errors.
sound/soc/codecs/wm8994.c: In function 'wm8994_readable':
sound/soc/codecs/wm8994.c:58: warning: unused variable 'wm8994'
sound/soc/samsung/smdk_wm8994.c:176: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/smdk_wm8994.c:176: warning: data definition has no type or storage class
sound/soc/samsung/smdk_wm8994.c:176: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION'
sound/soc/samsung/smdk_wm8994.c:176: warning: function declaration isn't a prototype
sound/soc/samsung/smdk_wm8994.c:177: error: expected declaration specifiers or '...' before string constant
- ASoC: Flush Samsung DMA on free
Ever since it was written the Samsung DMA driver has had a TODO in the
hw_free() function wondering if we need to flush the DMA buffers. Up until
now the answer has been no but with the recent improvements Boojin has
done to the DMA infrastructure for the Samsung port the answer has changed
to yes for at least S3C6410 systems.
If we don't then when we next prepare() the channel the API will get
confused trying to run callbacks on the transfers hanging around from the
previous time the stream was open and oops.
- ASoC: Samsung: Update DMA interface
This patch adds to support the DMA PL330 driver that uses
DMA generic API. Samsung sound driver uses DMA generic API
if architecture supports it. Otherwise, use samsung specific
S3C-PL330 API driver to transfer PCM data.
[kgene.kim@samsung.com: removed useless variable]
- ASoC: Convert Goni to data based DAPM init
- ASoC: Convert Jive to table based init
- ASoC: Convert SMDK WM8580 to table based DAPM init
- ASoC: Convert SmartQ to table based init
- ASoC: Convert RX1950 to table based init
- ASoC: Convert H1940 to table based init
- ASoC: Convert Simtec machines to table based DAPM init
- ASoC: samsung: s3c-i2s-v2.c needs module.h
Include <linux/module.h> to fix below build error:
CC sound/soc/samsung/s3c-i2s-v2.o
sound/soc/samsung/s3c-i2s-v2.c:573: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:573: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:573: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:638: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:638: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:638: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:677: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:677: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:677: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c: In function 's3c_i2sv2_register_dai':
sound/soc/samsung/s3c-i2s-v2.c:736: warning: initialization discards qualifiers from pointer target type
sound/soc/samsung/s3c-i2s-v2.c: At top level:
sound/soc/samsung/s3c-i2s-v2.c:754: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:754: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:754: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:756: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/s3c-i2s-v2.c:756: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:756: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE'
sound/soc/samsung/s3c-i2s-v2.c:756: warning: function declaration isn't a prototype
make[3]: *** [sound/soc/samsung/s3c-i2s-v2.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: samsung: WM8994 depends on MFD_WM8994
Any driver that selects SND_SOC_WM8994 should also make sure that
MFD_WM8994 is set, since the codec relies on the mfd code:
sound/built-in.o: In function `wm8994_read':
last.c:(.text+0x20160): undefined reference to `wm8994_reg_read'
sound/built-in.o: In function `wm8994_write':
last.c:(.text+0x20e68): undefined reference to `wm8994_reg_write'
This solves the problem by selecting the MFD driver directly
and adding extra 'depends on' statements to make sure that we
respect the dependencies of that driver.
- ASoC: Staticise simtec_audio_resume()
It is exported via resume callback of struct dev_pm_ops rather than referenced
directly and so should be staticised.
- ASoC: Staticise samsung_spdif_dai
- ASoC: samsung: Add __devexit_p at necessary places
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the confi
options."
- ASoC: Use dai_fmt in speyside_wm8962
- ASoC: Add DMIC control to Speyside WM8962 board
- ASoC: Add support for on-board analogue microphones on Speyside WM8962
- ASoC: Convert WM8962 MICBIAS to a supply widget
A supply widget is generally clearer than a MICBIAS widget and a mic bias
is just a type of supply so use a supply widget for the MICBIAS. This also
avoids confusion with the routing when connected to multiple inputs.
- ASoC: Support a wider range of sample rates on Speyside WM8962
As we've only got one audio interface and it is symmetric we can just set
SYSCLK based on the sample rate requested by the application layer. Provide
a default so bypass paths work before audio playback.
- ASoC: Add line loads to the list of supported detections for Speyside
- ASoC: samsung: Fix checking return value of clk_get
clk_get() returns a pointer to the struct clk or an ERR_PTR().
This patch also use PTR_ERR() for return value.
- ASoC: SAMSUNG: Add Kconfig to support SMDK4212
This patch adds Kconfig to support SMDK4212.
SMDK4212 is based on samsung exynos4212 SoC.
And WM8994 is used for audio codec.
- ASoC: Add Springbank I/O card to Speyside Kconfig
- ASoC: Ensure we only run Speyside WM8962 bias level callbacks once
We get called once per DAPM context but only need to run once. When DAPM
was serialized this was a series of noops but now it can run in parallel
we need to take proper care.
- ASoC: Run Speyside WM8962 at 512fs
Ensure we have access to all the advanced DSP functinality offered by the
WM8962 by running the system clock at 512fs.
- ASoC: rx1950: Fix compilation error due to missing header
Add linux/types.h to fix this compilation error:
In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0,
from arch/arm/mach-s3c2410/include/mach/gpio.h:27,
from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5,
from include/linux/gpio.h:18,
from sound/soc/samsung/rx1950_uda1380.c:20:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function)
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here
arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’
- ASoC: h1940: Fix compilation error due to missing header
Add linux/types.h to fix this compilation error:
In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0,
from arch/arm/mach-s3c2410/include/mach/gpio.h:27,
from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5,
from include/linux/gpio.h:18,
from sound/soc/samsung/rx1950_uda1380.c:20:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’
arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function)
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant
arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here
arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level:
arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’
arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’
- ASoC: Allow userspace control of Speyside headphone output
In order to facilitate the widest range of use cases (especially things
like speakerphone) allow the headphone output to be enabled and disabled
by the application layer.
- ASoC: Update SMDKs for WM8580 -codec removal
- ASoC: SAMSUNG: Add I2S0 internal dma driver
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.
[Reapplied after dependencies propagated through in 3.1-rc1. --broonie]
- ASoC: Fix warning in Speyside WM8962
- ASoC: Fix binding of WM8750 on Jive
The I2C address is misformatted and would never match.
- ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
This reverts commit d7c3e9525ac8e898f1156a1f3a7c5038f6560186 as it does
not currently build due to missing dependencies in the Samsung tree.
- ASoC: SAMSUNG: Add I2S0 internal dma driver
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.
- ASoC: SAMSUNG: Modify I2S driver to support idma
Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c
- ASoC: Improve error reporting in Speyside WM8962 driver
- ASoC: SAMSUNG: Add idma related register definition
This patch add idma related register definitions to support idma.
- ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.
- ASoC: SAMSUNG: Move I2S common register definition
I2S registers can be used for control idma.
Previously, register is defined in i2s.c.
For sharing the registers, It is moved to i2s-regs.h
- ASoC: SAMSUNG: Add WM8994 PCM Machine driver
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.
- ASoC: SMDKV310: Enable SPDIF device
- ASoC: Fix mismerge of Speyside set_bias_level_post()
- ASoC: Support Speyside build variants with WM8962 fitted
- ASoC: Manage Speyside system clocking only in bias management
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.
- ASoC: Update speyside audio driver for hardware revision 2
Revision 2 of the Speyside platform supplies a 32kHz clock on MCLK2 rather
than MCLK1.
- ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
If DMA active status should be checked, I2SCON register should be referenced.
In this patch, Fix the incorrect referencing of I2SCON register.
Reported-by : Lakkyung Jung <lakkyung.jung@samsung.com>
- ASoC: Don't specify the DMA driver for Speyside baseband link
- ASoC: Mark Speyside widgets as ignoring suspend
Allow audio paths through the Speyside system to be kept active while the
system is suspended (for example, when on a voice call) by marking all the
external widgets and the DAI link to the WM1250-EV1 baseband module as
ignoring suspend.
- ASoC: Add stub baseband link on Speyside
Demonstrate the connection of a baseband to the system. We add a DAI for
the link to the baseband. This will become visible to the application
layer - audio should be started from the application layer using an
application such as this:
http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c
which starts up audio as for CPU based playback and record up to the point
where data is streamed.
Due to non-availability of baseband simulation hardware we reuse the
configuration for the CPU link with the CODEC acting as clock master,
allowing signals to be observed with a scope. A more standard system
would have separate configuration for the baseband with its own ops
structure and operations. Normally the baseband would be clock master
as the baseband audio will be synchronised to the external telephony
network.
- ASoC: Add pin switches for fixed analogue inputs and outputs on Speyside
Pin switches enable direct control of the DAPM state from userspace,
enabling simple enabling and disabling of the path. This is especially
useful for outputs such as the speaker which are composed of several
physical devices as it allows them to be controlled as a group.
- ASoC: Add Speyside headset jack detection support
Speyside makes use of support the WM8915 has for detecting the polarity
of the microphone and ground connections on headsets, using a GPIO to
control the polarity of the ground connection and switching between the
two microphone bias supplies available on the device in order to do so.
As a result of this the detection support is more involved than for most
other CODECs, using a callback to configure the current polarity of the
jack and translate this into the board-specific connections required for
the current scenario.
On Android some additional work is required to hook this up to the
application layer as the Android HeadsetObserver monitors a custom
drivers/switch API rather than the standard Linux APIs. This can be
done by either updating HeadsetObserver or modifying the ALSA core to
report via drivers/switch as well.
- ASoC: Support the sub speaker driver on Speyside
Speyside includes a WM9081 configured as an external speaker driver taking
an analogue input from HPOUT2 on the WM8915 on the system. Add support for
this to the driver, using a prefix of "Sub" for the WM9081 controls to
ensure we avoid collisions with controls on the WM8915.
- ASoC: Optimise clock management for WM8915 Speyside
Dynamically enable and disable the FLL on the WM8915, configuring the
system clock to 256fs for 48kHz when the device is active but reverting
to using the input 32.768kHz clock directly at other times to support
features such as jack detection with minimal power consumption.
- ASoC: Add basic widgets for WM8915 Speyside
Provide widgets for the basic widgets connected directly to the WM8915
on Speyside - the headphones, speaker, digital and analogue microphones.
For the outputs this is just documentation, for the inputs this ensures
that the relevant microphone biases are enabled when they are in use.
- ASoC: Remove to support sound for S5P6442
According to removing ARCH_S5P6442, we don't need to support
sound for S5P6442.
- ASoC: Don't specify the DMA driver for Goni baseband link
- ASoC: Don't specify the DMA driver for OpenMoko baseband link
- ASoC: Fix CODEC DAI names for Goni
Immediately after sending the last fix I realised that the CODEC DAI names
also don't correspond to the WM8994 driver. Update the DAI names to match.
- ASoC: Fix CODEC name in Goni
This was typoed at some point in the multi-component merge, though the
driver was added along with that.
- ASoC: Initial audio support for Speyside on Cragganmore 6410
This is minimal code required to get audio out of the Speyside audio
subsystem on the Wolfson Cragganmore 6410 reference platform. It sets
up the link between the CPU and AIF1 of the WM8915 on the system,
enabling audio playback via the headphone and speaker outputs of the
device (which require no further configuration except runtime). It
allows verification of basic functionality of the system.
- ASoC: SAMSUNG: Add WM8580 PCM Machine driver
This patch add WM8580 PCM machine driver to support PCM audio
on SMDKC110, SMDKV210, SMDK6450, SMDK6440 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKC110, SMDKV210, SMDK6450
- ASoC: SAMSUNG: Fix the inverted clocks handling for pcm driver
Fix the inverted clocks handling for pcm cpu driver.
By using SND_SOC_DAIFMT_NB_NF, Audio noise can be generated on SMDK.
- ASoC: mini2440: Fix uda134x codec problem.
ASoC audio for mini2440 platform in current kenrel doesn't work.
First problem is samsung_asoc_dma device is missing in initialization.
Next problem is with codec. Codec is initialized but never probed
because no platform_device exist for codec driver. It leads to errors
during codec binding to asoc dai. Next problem was platform data which
was passed from board to asoc main driver but not passed to codec when
called codec_soc_probe().
Following patch should fix issues. But not sure if in correct way.
Please review.
- ASoC: Change dependency of ARCH_EXYNOS4
This patch changes dependency of ARCH_EXYNOS4 from ARCH_S5PV310
according to the change of ARCH name, EXYNOS4.
- ASoC: Samsung: Merge neo1937_wm8753 and neo1973_gta02_wm8753 sound board driver
The neo1973(GTA01) and neo1973_gta02(GTA02) have a very similar audio hardware
setup. They both use the same codec with the same routing to the gsm modem and
bluetooth chip. But they do use different AMPs though and there are some minor
differences in the speaker setup.
As a result most of the code of those two drivers is identical.
So from a maintenance point of view it makes sense to merge them into a single
driver. It also reduces the size of kernel images supporting both the GTA01 and
GTA02.
As a side-effect of this merge the GTA01 for example gains support for routing
audio to and from the bluetooth DAI.
- ASoC: neo1973_wm8753: Remove scenario management code.
It has been proven to be inflexible to do scenario management in kernel space.
Since actual neo1973 board support has not been merged in mainline and this
patch has been in the neo1973 tree for some time now it should be safe to remove
this functionality without breaking existing userspace.
- ASoC: remove one to many l's in the word
The patch below removes an extra "l" in the word.
- ASoC: neo1973_wm8753 audio support does not require scoop
This driver does not use any of the functionality provided by the scoop
hardware. Remove the unneeded header.

SoC Blackfin

- Add missing soc/* stub files
- ASoC: check for substream not channels_min in pcm engines
This is a follow up on 53dea36c70c1857 which fixes the other affected
pcm engines.
Description from 53dea36c70c1857:
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Without this patch I was seeing null-pointer dereferenc in atmel-pcm.
- ASoC: blackfin: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert blackfin directory to module_platform_driver
Factor out some boilerplate code.
- blackfin: add module.h to files implicitly expecting to use it.
Its presence was implicit everywhere, but we are aiming to fix that,
so call out the users explicitly.
- ASoC: Staticise bf5xx_pcm_i2s_new()
It is not used outside this driver so no need to make the symbol global.
- ASoC: Staticise bf5xx_pcm_ac97_new()
It is not used outside this driver so no need to make the symbol global.
- ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe
Fix below build warning:
sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type
- ASoC: Blackfin: bf5xx-ad193x: Fix codec device name
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
- ASoC: Blackfin: ADAU1373 eval board support
Add a machine driver to support the EVAL-ADAU1373 board connected to a
Analog Devices BF5XX evaluation board.
- ASoC: ad193x: fix system clock
system clock is 24.576MHz instead of 12.288MHz
- ASoC: Blackfin: Add machine driver for EVAL-ADAV80X boards
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.
- ASoC: Blackfin: allow SPI for SSM2602 parts
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.
- ASoC: Blackfin: Add bf5xx-adau1701 machine driver
Add a machine driver to support the ADAU1701 SigmaDSP processors on
Analog Devices BF5XX evaluation boards.
- ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver
- ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
- ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
- ASoC: Blackfin: push down SPORT settings from global variables
Now that we have multi-component support, take the time to unify the
SPORT implementations a bit and make the setup dynamic. This kills
off the global sport_handle which was shared across all the Blackfin
machine drivers. The pin management aspect is off loaded to platform
resources, and now multiple SPORTs can be instantiated simultaneously.
- ASoC: Blackfin: standardize machine driver names
Some machine drivers were using "bf5xx-", others were using "bf5xx_",
while others were using "bfin-". Further, some were using the same
name in the transport layer which makes it hard to use different codecs
at the same time. So standardize all of them to "bfin-" and make sure
they are name spaced according to their driver name.
- ASoC: Blackfin: drop "-codec" from codec names
The recent multi-component patch incorrectly added "-codec" suffixes to
parts which are not MFD. Drop the suffix from the machine drivers too.
- ASoC: Blackfin: add ad193x sysclk configuration
- ASoC: Blackfin I2S: add 8-bit sample support

SoC Codec 88PM860x

- Add missing soc/* stub files
- ASoC: Convert 88pm860x-codec to devm_kzalloc()
- ASoC: Convert 88pm860x-codec to table based DAPM and control init
- ASoC: Convert CODEC drivers to module_platform_driver
Factors out a bit of boilerplate.
- ASoC: Include delay.h in 88pm860x
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: 88pm860x-codec - reset the codec correctly
Reset the codec according to the Audio power-up delay errata for the 88PM8607.
- ASoC: 88pm860x-codec - Allow independent use of both I2S playback and capture
Introduce a I2S CLK supply so playback and capture can operate independently.
- ASoC: s/w->kcontrols/w->kcontrol_news/g
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.

SoC Codec AC97

- ASoC: Drop unused state parameter from CODEC suspend callback
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
- ASoC: Convert CODEC drivers to module_platform_driver
Factors out a bit of boilerplate.

SoC Codec AD1836

- ASoC: Convert ad1836 to devm_kzalloc()
- ASoC: Drop unused state parameter from CODEC suspend callback
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
- ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET
According to the datasheet:
The BIT[5:4] of ADC Control Register 2 is to control the word width.
00 = 25 Bits
01 = 20 Bits
10 = 16 Bits
11 = Invalid
Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4.
- ASoC: AD1836: rename suspend/resume funcs
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.
- ASoC: AD1836: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.
- ASoC: AD1836: fix intermixed tab/space indentation
- ASoC: AD1836: drop unnecessary spi register check
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us. So drop the redundant checking/message.
- ASoC: AD1836: clean up comment headers
- ASoC: AD1836: Fix build error
Commit f97d0c6d5f94 ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes it.
- ASoC: AD1836: Add input gain control for ADC2
The AD1836 has a PGA for its second ADC. This patch adds a control for
adjusting the the gain of the PGA.
- ASoC: AD1836: Remove unused fields from private struct
The control_type field is never used, so it can be removed. The
control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: AD1836: Add AD1835/AD1837/AD1838/AD1839 support
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver. The main difference between those devices
is the number of DACs and ADCs.
This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.
The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.
- ASoC: AD1836: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
- ASoC: AD1836: Add ADC/DAC controls helper macros
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.
- ASoC: AD1836: Fix setting the PCM format

SoC Codec AD1938

- Add missing soc/* stub files

SoC Codec AD193X

- ASoC: Convert ad193x to devm_kzalloc()
- ASoC: ad193x: Convert to direct regmap API usage
- ASoC: ad193x: Use snd_soc_update_bits where appropriate
We can reduce the code size here a bit by using snd_soc_update_bits instead of
open-coding the read-modify-write cycle. The conversion done in this patch is
not completely straightforward and some minor code restructuring has been
incorporated to further reduce the code size.
- ASoC: ad193x: Add sysclk DAPM supply
Add a DAPM supply widget for the internal sysclk, so it can be disabled
automatically when not needed.
- ASoC: ad193x: Remove non-functional DAPM route controls
DAPM route controls only take effect on paths where the sink is a mixer or a
mux, furthermore the control must be a control assigned to the mixer or mux.
- ASoC: ad193x: Make enum items const char * const
- ASoC: ad193x: Provide dB ranges for the volume controls
- ASoC: ad193x: Use table based DAPM and controls setup
- ASoC: ad193x: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: ad193x: Fix define of AD193X_PLL_INPUT_MASK
Current code defines AD193X_PLL_INPUT_MASK as (~0x6) which is quite
different from other MASK defines.
To make it consistent with other mask defines, define AD193X_PLL_INPUT_MASK
as 0x6 and change the code accordingly.
I think this change improves the readability.
- ASoC: ad193x: Setup regmap read and write flag masks for SPI
Currently register read-back for the ad193x is broken, because it expects bit 0
of the upper byte to be set to indicate a read operation, while the regmap
default for SPI is to use bit 7.
This patch also addresses another oddity of the device. There are SPI and I2C
versions of this codec. In both cases the registers are 8-bit wide and numbered
from 0x0 to 0x10, but in the SPI case there is also a so called
'global address' which is prefixed in-front of the register address. The global
address mimics I2C behaviour and includes a static device address the and the
read/write flag. This basically extends the register address to an 16-bit value
numbered from 0x800 to 0x810. These are the register numbers which are
currently used by the driver. This works, because I2C will ignore the upper
8 bits of the register, but it is still a bit confusing, as there are no such
register numbers in the I2C case.
The approach taken by this patch is to number the registers from 0x00 to 0x10
and encode the global address for SPI mode into the read and write flag masks.
- ASoC: ad193x: remove cache support
asoc cache layer can't support this kind of spi registers well.
remove cache support and read/write registers directly
- ASoC: ad193x: fix dac word len setting
dac word len value should left shift before setting
- ASoC: ad193x: fix registers definition
fix dac word len mask and adc tdm fmt shift value
- ASoC: ad193x: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.
- ASoC: ad193x: tweak style to match other codecs
Rename the snd_soc_control_type field from "bus_type" to "control_type",
and drop the now unused "control_data" field.

SoC Codec AD1980

- ASoC: Convert CODEC drivers to module_platform_driver
Factors out a bit of boilerplate.
- ASoC: Drop exporting ad1980_dai
ad1980_dai is not used outside this driver,
thus drop exporting it.
- ASoC: ad1980: Return proper error if vendor id mismatch
Return -ENODEV instead of 0 if vendor id mismatch.
- ASoC: ad1980: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.

SoC Codec AD73311

- ASoC: Convert CODEC drivers to module_platform_driver
Factors out a bit of boilerplate.
- ASoC: ad73311: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.

SoC Codec ADAU1373

- ASoC: Convert adau1373 to devm_kzalloc()
- ASoC: Drop unused state parameter from CODEC suspend callback
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
- ASoC: Cleanup duplicated const
Commit 85e7652("ASoC: Constify snd_soc_dai_ops structs") accidentally
introduced a few duplicated consts. This patch cleans it up.
- ASoC: adau1373: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the array.
- ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS
In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false.
- ASoC: Add ADAU1373 codec support
This patch adds support for the Analog Devices ADAU1373 audio codec.

SoC Codec ADAU1701

- ASoC: Convert adau1701 to devm_kzalloc()
- ASoC: Move SigmaDSP firmware loader to ASoC
It has been pointed out previously, that the firmware subsystem is not the right
place for the SigmaDSP firmware loader. Furthermore the SigmaDSP is currently
only used in audio products and we are aiming for better integration into the
ASoC framework in the future, with support for ALSA controls for firmware
parameters and support dynamic power management as well. So the natural choice
for the SigmaDSP firmware loader is the ASoC subsystem.
- ASoC: Cleanup duplicated const
Commit 85e7652("ASoC: Constify snd_soc_dai_ops structs") accidentally
introduced a few duplicated consts. This patch cleans it up.
- ASoC: adau1701: Fix prototype for adau1701_set_sysclk
- ASoC: adau1701: Initialize codec->control_data before using it
Currently codec->control_data is not initialized before calling
process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE).
- ASoC: adau1701: signedness bug in adau1701_write()
"ret" is supposed to be signed here. The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.
- ASoC: Add ADAU1701 codec driver
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.

SoC Codec ADAV80x

- ASoC: Allow source specification for CODEC level sysclk
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
- ASoC: Report an error for unknown adav80x formats
Not only fixes error handling but also some uninitialized variable
warnings.
- ASoC: Add ADAV80x codec driver
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.

SoC Codec AK4104

- ASoC: Convert ak4104 to devm_kzalloc()
- ASoC: Fix reg_word_size for ak4104
According to the register map in datasheet, the registers are 8 bit.
- ASoC: Remove module probe announcements from CODEC drivers

SoC Codec AK4535

- ASoC: Convert ak4535 to devm_kzalloc()
- ASoC: Remove unneeded platform_device.h inclusions from CODECs
They've not been needed for a long time if they were ever required.
- ASoC: ak4535: fixup cache register table
ak4535_reg should be 8bit, but cache table is defined as 16bit.
- ASoC: ak4535: convert to soc-cache
- ASoC: Use data based init for ak4535 DAPM

SoC Codec AK4641

- ASoC: Fix return value of ak4641_pcm_set_dai_fmt()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.
- ASoC: Convert ak4641 to devm_kzalloc()
- ASoC: Remove unneeded platform_device.h inclusions from CODECs
They've not been needed for a long time if they were ever required.
- ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget
- ASoC: ak4641: Remove unused codec field from struct ak4641_priv
- ASoC: Staticize ak4641_dai
- ASoC: Asahi Kasei AK4641 codec driver
A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800
among others.

SoC Codec AK4642

- ASoC: Convert ak4642 to devm_kzalloc()
- ASoC: Remove unneeded platform_device.h inclusions from CODECs
They've not been needed for a long time if they were ever required.
- ASoC: Remove driver versioning from ak4642
It's never been updated so it can't be that useful and it makes the
driver needlessly chatty.
- ASoC: ak4642: add ak4648 support
- ASoC: ak4642: add Line out support
- ASoC: ak4642: add headphone mute switch control
- ASoC: ak4642: add DAPM support for HeadPhone Output
- ASoC: ak4642: add ak4642_set_bias_level()
- ASoC: ak4642: ak4642 was tested
ak4642 was tested by ms7724se board
- ASoC: ak4642: fixup cache register table
ak4642 register was 8bit, but cache table was defined as 16bit.
ak4642 doesn't work correctry without this patch.
- ASoC: ak4642: convert to soc-cache
- ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
mask didn't cover update-data
- ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support
This patch support LEFT_J / I2S only for now

SoC Codec AK4671

- ASoC: Convert ak4671 to devm_kzalloc()
- ASoC: ak4671: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Remove unneeded hw_write initialisation in ak4671
It is not required now.
- ASoC: Remove unused "control_data" field of struct ak4671_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Use data based init for ak4671 DAPM

SoC Codec CQ0093 Voice

- mfd: Use mfd cell platform_data for davinci cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
- asoc: davinci_voicecodec: use mfd_data instead of driver_data
Use mfd_data for passing information from mfd drivers to soc
clients. The mfd_cell's driver_data field is being phased out.
Clients that were using driver_data now access .mfd_data
via mfd_get_data().
- ASoC: DaVinci: fix kernel panic due to uninitialized platform_data
This patch fixes the Kernel panic issue on accessing davinci_vc in
cq93vc_probe function. struct davinci_vc is part of platform device's
private driver data(codec->dev->p->driver_data) and this is populated
by DaVinci Voice Codec MFD driver.

SoC Codec CS4270

- ASoC: Convert cs4270 to devm_kzalloc()
- ASoC: cs4720: use snd_soc_cache_sync()
Replace the manual register restore mechanism in cs4270.c and call the
generic snd_soc_cache_sync() handler instead.
This factors code out in favour of core facilities and also fixes a
bus confusion that is most probably caused by intermixing i2c-regmap
functions and i2c_smbus_* accessors.
Reported-and-tested-by: Sven Neumann <s.neumann@raumfeld.com>
- ASoC: Return early with -EINVAL if invalid dai format is detected
- ASoC: Remove unused "control_data" field of struct cs4270_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Fix comment in cs4270 codec driver
The comment does not reflect reality anymore since the multi-component
monster patch landed. Things are matched by names now, and not by
exporting and referencing a struct. Fix it to avoid confusion.
- ASoC: Constify i2c_device_id table
- ASoC: Remove module probe announcements from CODEC drivers

SoC Codec CS42L51

- ASoC: Convert cs42l51 to devm_kzalloc()
- ASoC: Convert cs42l51 to table based DAPM and control init
- ASoC: cs42l51: Fix off-by-one for reg_cache_size
Just checking the code in cs42l51_fill_cache():
The cache pointer points to codec->reg_cache + 1.
I think it is because CS42L51_FIRSTREG is 0x01,
so codec->reg_cache[0] is not used here.
Then we read CS42L51_NUMREGS bytes to cache.
So we need reg_cache_size to be CS42L51_NUMREGS + 1.
- ASoC: Avoid a redundant read in cs42l51_pdn_event
snd_soc_update_bits already does read-modify-write,
no need to read the register before calling snd_soc_update_bits.
- ASoC: Return early with -EINVAL if invalid dai format is detected
- ASoC: Remove unused "control_data" field of struct cs42l51_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

SoC Codec CS42L73

- ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt
What we want is to clear BIT[5:4](PCM_MODE_MASK) and BIT[3](PCM_BIT_ORDER) bits,
but current code clears BIT[2:0].
- ASoC: Convert CS42L73 to devm_kzalloc()
- ASoC: Staticise and constify cs42l73_reg_defaults
It's not exported and doesn't need to change.
- ASoC: cs42l73: Make inv and format to be unsigned int
Fix below smatch warning:
sound/soc/codecs/cs42l73.c +1030 cs42l73_set_dai_fmt(53) error: inv is never equal to 1024 (wrong type 0 - 255).
sound/soc/codecs/cs42l73.c +1032 cs42l73_set_dai_fmt(55) error: inv is never equal to 768 (wrong type 0 - 255).
sound/soc/codecs/cs42l73.c +1036 cs42l73_set_dai_fmt(59) error: inv is never equal to 1024 (wrong type 0 - 255).
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
- ASoC: cs42l73: Unify the way to define bits of register
Current code defines some bits with left shift to the proper bit defined in
datasheet, but some don't.
Unify the definition with proper left shift and adjust the code accordingly.
- ASoC: Remove redundant regcache_sync call in cs42l73_resume
It's done in cs42l73_set_bias_level when the dapm.bias_level is switching
from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY.
- ASoC: cs42l73: Show correct revision id
- ASoC: cs42l73: Return proper error code if device id mismatch
Return -ENODEV instead of 0 if device id mismatch.
- ASoC: Add support for CS42L73 codec
This patch adds support for the Cirrus Logic CS42L73 low power stereo
codec.

SoC Codec CX20442

- ASoC: cx20442: add bias control over a platform provided regulator
Now that a regulator device for controlling the codec chip reset state
over a platform agnostic regulator API is available on the only board
using this driver so far, extend the driver with a bias control function
which will request virtual power to the codec chip from that virtual
regulator, and will supersede the present implementation existing at the
sound card level.
Thanks to the regulator sharing mechanism, both the old (the sound card)
and the new (the codec) implementations should coexist smoothly until
the sound card file is updated. For this to work as expected, update the
sound card .set_bias_level callback to not touch codec->dapm.bias_level.
While extending the cx20442 structure, drop unused control_type member.
Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1:
ams-delta: set up a regulator over the modem reset GPIO pin".
- Revert "ASoC: Update cx20442 for TTY API change"
This reverts commit ed0bd2333cffc3d856db9beb829543c1dfc00982.
Since we reverted the TTY API change, we should revert the ASoC update
to it too.
- ASoC: Update cx20442 for TTY API change
receive_buf() was recently changed to return the number of bytes
received but the cx20442 driver wasn't updated to match the new API.
I don't have any hardware but since we don't actually appears to be
listening to the data at all just report that we accepted all the data
that was offered to us.
- ASoC: Use data based init for cx20442 DAPM
- ASoC: CX20442: fix wrong reg_cache_default content
Content of the CX20442's snd_soc_codec_driver.reg_cache_default pointed
area, introduced with my recent NULL pointer dereferece fix (commit
f019ee5feb344ff0b22b58df4568676295aae14f), occured wrong after further
testing, more thorough than just booting successfully. There are two
problems with it:
1) It should read
(1 << CX20442_TELOUT) | (1 << CX20442_MIC),
not
CX20442_TELOUT | CX20442_MIC.
2) While correctly matching actual codec hardware state on boot when
fixed per 1), a few more code modifications would still be required
to reflect that state not only into register cache, but also force
them into DAPM pins state, otherwise an inconsitency occures which
may prevent further codec state changes from being applied correctly.
As a result, the phone stops ringing after reboot, until someone
picks up the handset for the first time.
Revert that reg_cache_default content to a working, previous de facto
default value of 0, in hope this change can still be accepted as an rc
cycle fix.
Created and tested against linux-2.6.38-rc4
- ASoC: CX20442: fix NULL pointer dereference
The CX20442 codec driver never provided the snd_soc_codec_driver's
.reg_cache_default member. With the latest ASoC framework changes, it
seems to be referred unconditionally, resulting in a NULL pointer
dereference if missing. Provide it.
Created and tested on Amstrad Delta against linux-2.6.38-rc2

SoC Codec Cirrus Logic CS4271

- ASoC: cs4271: Fix wrong mask parameter in some snd_soc_update_bits calls
- ASoC: Remove unused "control_data" field of struct cs4271_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC.
This is done to make capture work properly.
- ASoC: Extend range of supported sample rates for CS4271 CODEC.
Extend range of supported sample rates for CS4271 CODEC.
- ASoC: Constify i2c_device_id table
- ASoC: CS4271: Move Chip Select control out of the CODEC code.
Move Chip Select control out of the CODEC code for CS4271.
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
- ASoC: Staticise non-exported symbols in cs4271
- ASoC: cs4271.c: improve error handling
CS4271 CODEC driver adapted to recently introduced error handling in
snd_soc_update_bits().
Added snd_soc_cache_sync() error handling.
- ASoC: CS4271 codec support
Added support for CS4271 codec to ASoC.

SoC Codec DA7210

- ASoC: da7210: Add support for line input and mic
DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and
a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as
well as INPGA.
- ASoC: Convert da7210 to devm_kzalloc()
- ASoC: Fix duplicate const warnings in da7210.c
- ASoC: da7210: Add support for line out and DAC
DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono).
This patch adds support for gain controls for these three line outs.
It also adds support for overall DAC gain control.
- ASoC: da7210: Add support for DAPM
This patch adds support for DAPM covering all inputs and outputs
as well as ADC and DAC.
- ASoC: da7210: Add support for ALC and Noise suppression
This patch adds controls to set following ALC parameters,
- Max gain, Min gain, Noise gain, Attack rate, Release rate and delay
It also adds a switch to enable/disable noise suppression.
As per DA7210 data sheet, ALC and noise suppression can be enabled
only if certain conditions are met. This condition checks are handled
by simply using "_EXT" version of controls to capture change events.
- ASoC: da7210: Add support for mute and zero cross controls
This patch adds support for below set of controls,
(1) Mute controls for MIC, AUX and ADC
(2) Zero cross controls for head phone, AUX, INPGA and line out
(3) Head phone mode selection - class H or G
It also adds digital_mute() call back.
- ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
This patch add controls for setting cut-off for high pass and voice
filters of ADC and DAC. There are also switches to enable/disable
these filters.
Also removed hard coded, fixed values of these parameters used by
previous version of driver.
- ASoC: da7210: Add support for ADC & DAC equalizers
This patch adds support for ADC and DAC five band equalizers
available on DA7210 codec.
- ASoC: da7210: bugfix for head phone volume control
This patch takes care of reserved bits of headphone volume
register by using correct volume range.
- ASoC: Convert DA7210 to table based DAPM init
- ASoC: da7210: Add support for other DAI word lengths, format and mode
This patchs adds support for following,
(1) DAI 20 and 32 bit word sizes
(2) DAI left and right justified formats
(3) DAI slave mode
- ASoC: da7210: convert to soc-cache

SoC Codec DFBM-CS320 bluethooth

- ASoC: Add driver for the dfbmcs320 bluetooth module
This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used
on the neo1973 boards.
The patch also modifies the neo1937_wm8753 sound board driver to use the new
driver instead of registering the bluetooth DAI manually.
Previously there was a name mismatch between the bluetooth DAI and the bluetooth
DAI link and the sound card was not instantiated, with this patch the issue is
no longer present and sound support works again.

SoC Codec DIT SPDI/F

- ASoC: spdif-dit: Add missing MODULE_*
MODULE_ALIAS is required so that the module will auto-load based on a
platform_device registration in the board file.
While we're at it, add some other MODULE_*.

SoC Codec Freescale SGTL5000

- ASoC: sgtl5000: update author email address
Update MODULE_AUTHOR email address.
- regulator: pass additional of_node to regulator_register()
With device tree support for regulators, its needed that the
regulator_dev->dev device has the right of_node attached.
To be able to do this add an additional parameter to the
regulator_register() api, wherein the dt-adapted driver can
then pass this additional info onto the regulator core.
- ASoC: Convert sgtl5000 to use devm_kzalloc()
Convert sgtl5000 codec driver to use devm_kzalloc().
- ASoC: sgtl5000: Fix voltage units in dev_err message
vdda, vddio and vddd are voltages expressed in milivolts (mV), so use the
proper annotation.
- ASoC: sgtl5000: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the array.
- ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
Ensure all mask bits are clear before setting new value.
- ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
We have defined SGTL5000_LINREG_VDDD_MASK in sgtl5000.h,
use it instead of hardcoded (0x1 << 4) - 1 for the mask.
- ASoC: Set sgtl5000->ldo in ldo_regulator_register
Otherwise calling ldo_regulator_remove() does not unregister regulator
and free memories.
- ASoC: sgtl5000: Fix setting mic bias resistor
According to the datasheet:
CHIP_MIC_CTRL 0x002A
BITS[9:8] BIAS_RESISTOR
0x0 = Powerd off
0x1 = 2.0 kohm
0x2 = 4.0 kohm
0x3 = 8.0 kohm
To set mic bias resistor, we need to update bits[9:8] of
SGTL5000_CHIP_MIC_CTRL register.
- ASoC: sgtl5000: Fix define for SGTL5000_BIAS_R_MASK
According to the datasheet:
CHIP_MIC_CTRL 0x002A
BITS[9:8] BIAS_RESISTOR
0x0 = Powerd off
0x1 = 2.0 kohm
0x2 = 4.0 kohm
0x3 = 8.0 kohm
Thus SGTL5000_BIAS_R_MASK should be defined as 0x0300 instead of 0x0200.
- ASoC: sgtl5000: fix module device table type for sgtl5000_dt_ids
The module device table for of_device_id should use "of" type.
- ASoC: sgtl5000: add device tree probe support
It adds device tree probe support for sgtl5000 driver.
- ASoC: sgtl5000: fix cache handling
Cache handling in this driver is broken. The chip has 16-bit registers, yet the
register numbers also increase by 2 per register, i.e. there are only
even-numbered registers. The cache in this driver, though, simply increments
register numbers, so it does need some mapping as seen in
sgtl5000_restore_regs(), note the '>> 1':
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
That, of course, won't work with snd_soc_update_bits(). (Thus, we won't even
notice the missing register 0x1c in the default regs which shifted all follwing
registers to wrong values.) Noticed on the MX28EVK where enabling the regulators
simply locked up the chip.
Refactor the routines and use a properly sized default_regs array which matches
the register layout of the underlying chip, i.e. create a truly flat cache.
This also saves some code which should make up for the bigger array a little.
When soc-core will somewhen have another cache type which handles a step size,
this conversion will also ease the transition.
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
- ASoC: sgtl5000: guide user when regulator support is needed
Print a hint when the user has a setup where CONFIG_REGULATOR is really
needed to make the driver work.
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
- ASoC: sgtl5000: refactor registering internal ldo
The code for registering the internal ldo was present twice. Turn it
into a function instead. Also, inform the user if LDO is used now.
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
- ASoC: Support !REGULATOR build for sgtl5000
The regulator is optional depending on board design.
- ASoC: sgtl5000: use after free in ldo_regulator_register()
The "ldo" variable was dereferenced after free on the error path.
- ASoC: Staticise non-exported symbols in SGTL5000
- ASoC: remove unnecessary header including in SGTL5000 codec driver
Remove unnecessary headers:
- mach/hardware.h in sgtl5000.c
- linux/i2c.h in sgtl5000.h
- ASoC: Add Freescale SGTL5000 codec support
Add Freescale SGTL5000 codec support.
Supported features:
- line-in and mic input
- headphone and line-out output
- line-in bypass ADC and DAC to headphone
- 16, 20, 24, 32 bit audio
- 8 ~ 96k sample rates

SoC Codec General Digital MICs

- ASoC: DMIC codec - Add input widget
Digital microphones can have some additional elements in their
audio path (like microphone bias). An input widget is required
for digital microphone CODEC driver to allow external connections
in machine drivers.

SoC Codec Ingenic JZ4740

- ASoC: Convert jz4740 to devm_kzalloc()
- ASoC: Include linux/io.h for jz4740 codec
Include linux/io.h to fix below build errors:
CC sound/soc/codecs/jz4740.o
sound/soc/codecs/jz4740.c: In function 'jz4740_codec_read':
sound/soc/codecs/jz4740.c:82: error: implicit declaration of function 'readl'
sound/soc/codecs/jz4740.c: In function 'jz4740_codec_write':
sound/soc/codecs/jz4740.c:92: error: implicit declaration of function 'writel'
sound/soc/codecs/jz4740.c: In function 'jz4740_codec_probe':
sound/soc/codecs/jz4740.c:373: error: implicit declaration of function 'ioremap'
sound/soc/codecs/jz4740.c:373: warning: assignment makes pointer from integer without a cast
sound/soc/codecs/jz4740.c:393: error: implicit declaration of function 'iounmap'
make[3]: *** [sound/soc/codecs/jz4740.o] Error 1
make[2]: *** [sound/soc/codecs] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: codecs: JZ4740: Convert to table based controls and DAPM setup
Use the newly introduced dapm_widgets, dpam_routes and controls fields of the
snd_soc_dai_driver struct to setup controls and DAPM.
- ASoC: codecs: JZ4740: Fix OOPS
Commit ce6120cc(ASoC: Decouple DAPM from CODECs) changed the signature of
snd_soc_dapm_widgets_new to take an pointer to a snd_soc_dapm_context instead of
a snd_soc_codec. The call to snd_soc_dapm_widgets_new in jz4740_codec_dev_probe
was not updated to reflect this change, which results in a compiletime warning
and a runtime OOPS.
Since the core code calls snd_soc_dapm_widgets_new after the codec has been
registered it can be dropped here.

SoC Codec LM4857

- ASoC: Convert lm4857 to devm_kzalloc()
- ASoC: Remove references to linux@wolfsonmicro.com
- ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module
This patch moves the code for the lm4857 AMP from the neo1973_wm8753 sound
board driver to its own module.
The lm4857 is a generic AMP IC and not specific to the neo1973.

SoC Codec MAX98088

- ASoC: Convert max98088 to devm_kzalloc()
- ASoC: max98088 codec: Catch driver bugs for eq channel name
Move the EQ channel names to a separate array and iterate over it in
max98088_get_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.
Also added a BUILD_BUG_ON to ensure that the eq_mode_name and controls
arrays are the same size.
- ASoC: Remove unused "control_data" field of struct max98088_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Remove redundant freq assignment for
max98095->sysclk/max98088->sysclk
Current implementation set max98095->sysclk/max98088->sysclk to freq twice.
Set it once is enough, this patch removes the first assignment in case
we may set invalid clock frequency to max98095->sysclk/max98088->sysclk.
- ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2
Added digital mute function in DAI1 and DAI2.
- ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control
Moved the EX Limiter Mode from dapm widget to control, because it was not
required DAPM route.
- ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls
Fixed invalid register definitions in mixer controls such as left
speaker mixer, left hp mixer and left rec mixer.
- ASoC: Use data based init for max98088 DAPM

SoC Codec MAX98095

- ASoC: Convert max98095 to devm_kzalloc()
- ASoC: max98095: Convert codec->hw_write to snd_soc_write
codec->hw_write is broken now, convert codec->hw_write to snd_soc_write.
The hardware has 2 banks of registers sharing a section in I2C register space.
The 1st bank is the primary one and is cached.
The 2nd bank is for loading coefficients only and they do not need cache.
These coefficients registers are therefore direct writes.
Thus we set cache_bypass flag to deal with this before calling snd_soc_write.
- ASoC: max98095 codec: Catch driver bugs for biquad channel name
Move the biquad channel names to a separate array and iterate over it in
max98095_get_bq_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.
Also added a BUILD_BUG_ON to ensure that the bq_mode_name and controls
arrays are the same size.
- ASoC: Remove unused "control_data" field of struct max98095_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: codecs: Max98095: Fix logging of hardware revision.
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
- ASoC: Max98095: Move existing NULL check before pointer dereference.
Visual inspection shows that max98095_put_eq_enum() and
max98095_put_bq_enum() each have a possible NULL deref of 'pdata'.
This change moves the NULL check above the use.
- ASoC: Add EQ and filter to max98095 CODEC driver
This patch adds the equalizer and biquad filter controls.
- ASoC: Remove redundant freq assignment for
max98095->sysclk/max98088->sysclk
Current implementation set max98095->sysclk/max98088->sysclk to freq twice.
Set it once is enough, this patch removes the first assignment in case
we may set invalid clock frequency to max98095->sysclk/max98088->sysclk.
- ASoC: Add max98095 CODEC driver
This patch adds the MAX98095 CODEC driver.

SoC Codec MAX9850

- ASoC: Convert max9850 to devm_kzalloc()
- ASoC: Convert max9850 to table based DAPM and control init
- ASoC: Fix spacing in MAX8950
- ASoC: Add MAX9850 codec driver
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM

SoC Codec MAX9877

- ASoC: max9877: Update register if either val or val2 is changed
In the case of ((max9877_regs[reg] >> shift) & mask) != val
but ((max9877_regs[reg2] >> shift) & mask) == val2,
current code does not update the registers.
Fix the logic to update registers if either val or val2 is changed.

SoC Codec Philips UDA134x

- ASoC: UDA134x: Remove POWER_OFF_ON_STANDBY define.
Define POWER_OFF_ON_STANDBY cause trobles when trying to get some
sound from codec because code for bias setup was not compiled
(define wasn't defined). This define was removed in commit:
cc3202f5 but again introduced by commit: f0fba2ad1 which then
completely break codec functionality so remove it again.
- ASoC: mini2440: Fix uda134x codec problem.
ASoC audio for mini2440 platform in current kenrel doesn't work.
First problem is samsung_asoc_dma device is missing in initialization.
Next problem is with codec. Codec is initialized but never probed
because no platform_device exist for codec driver. It leads to errors
during codec binding to asoc dai. Next problem was platform data which
was passed from board to asoc main driver but not passed to codec when
called codec_soc_probe().
Following patch should fix issues. But not sure if in correct way.
Please review.

SoC Codec Philips UDA1380

- ASoC: Convert uda1380 to devm_kzalloc()
- ASoC: Convert uda1380 to table based DAPM and control init
- ASoC: uda1380: Convert to gpio_request_one()
Using gpio_request_one can make the error handling simpler.
Also remove a redundant "Failed to issue reset" error message.
We already show the error message in uda1380_reset() error path.
- ASoC: uda1380: Return proper error in uda1380_modinit failure path
Return proper error for uda1380_modinit if i2c_add_driver() fails.

SoC Codec RT5631

- ASoC: Convert rt5631 to devm_kzalloc()
- ASoC: rt5631: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
the last entry from being omitted.
- ASoC: rt5631: Remove unused codec field from struct rt5631_priv
- ASoC: Remove unused function check_vdac_to_outmix from rt5631
- ASoC: Staticise non-exported symbols in rt5631
- ASoC: Staticize rt5631_dai
- ASoC: Add driver for rt5631

SoC Codec SSM2602

- ASoC: Convert ssm2602 to devm_kzalloc()
- ASoC: ssm2602: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: ssm2602: Support setting the oscillator and the clock output state
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.
This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.
- ASoC: ssm2602: Set initial bias level to standby
Set the initial bias level to standby during CODEC probe instead of leaving the
CODEC powered off.
- ASoC: ssm2602: Re-enable oscillator after suspend
Currently the the internal oscillator is powered down when entering BIAS_OFF
state, but not re-enabled when going back to BIAS_STANDBY. As a result the
CODEC will stop working after suspend if the internal oscillator is used to
generate the sysclock signal. This patch fixes it by clearing the appropriate
bit in the power down register when the CODEC is re-enabled.
- ASoC: ssm2602: Do not dereference codec->control_data
The driver assumes that control_data points to the drivers i2c_client struct,
but this is no longer the case since the ASoC core has switched to regmap.
- ASoC: Fix NULL vs. 0 warning in SSM2602
sparse complains if 0 is used as a NULL pointer constant.
- ASoC: SSM2602: Provide dB ranges for the volume controls
Also fix the maximum value for the capture volume control.
- ASoC: SSM2602: Model power supply for the digital core as a DAPM widget
Model the power supply for the digital core as a DAPM_SUPPLY widget. This allows
to cleanup the code a bit.
- ASoC: SSM2602: Add entry for the ssm2603 to the device id table
The SSM2603 is mostly register compatible with the SSM2602 and can be supported
by the current driver without any changes.
- ASoC: SSM2602: Add SSM2604 support
The SSM2604 is basically a lightweight variant of the SSM2602 with a compatible
register layout. Thus we can easily support both devices by the same driver,
by providing a slightly set of controls, widgets and routes.
Compared to the SSM2602 the SSM2604 has no microphone input and no headphone
output.
- ASoC: SSM2602: Fix reg_cache_size
reg_cache_size is supposed to be the number of elements in the register cache,
not the size in bytes.
- ASoC: SSM2602: Do not power the codec up in probe
It is not required to have the codec powered at this stage and DAPM will power
the ADC and DAC down again after probe has run anyway.
Thus we avoid some unnecessary writes by this change.
- ASoC: SSM2602: Fix default register cache
Some of the values in the default register cache did not represent the codecs
state after reset. This patch fixes it.
- ASoC: SSM2602: Remove unused struct and define
Those are leftovers from a pre-multicomponent era.
- ASoC: SSM2602: Remove duplicate control
There are currently two controls which allow selecting the capture source, one
as a normal control, the other as part of a DAPM_MUX widget.
Remove the normal control.
- ASoC: SSM2602: Cleanup coeff handling
Drop unused field from the coeff struct, precalculate the srate register at
compile-time and cleanup up the naming.
- ASoC: SSM2602: Fix 'Mic Boost2' control
The 'Mic Boost2' control's shift was off by one and thus was not working.
- ASoC: SSM2602: Properly annotate i2c probe and remove functions
Annotate the i2c probe and remove functions with __devinit and __devexit.
- ASoC: SSM2602: add SPI support
The ssm2602 codec has a SPI interface as well as I2C, so add the simple
bit of glue to make it usable.
- ASoC: SSM2602: convert to soc-cache
- ASoC: SSM2602: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD. This was incorrectly changed during the multi-component updated.

SoC Codec STA32X

- ASoC: sta32x: Optimize the array work to find rate_min and rate_max
For a given ir and fs, there is at most one possible match for the case
mclk_ratios[ir][j].ratio * fs == freq.
Thus we can break from the inner loop once a match is found.
- ASoC: Convert sta32x to devm_kzalloc()
- ASoC: Staticise non-exported symbols in sta32x
- ASoC: sta32x: add workaround for ESD reset issue
sta32x resets and loses all configuration during ESD test.
Work around by polling the CONFA register once a second
and restore all coeffcients and registers when CONFA
changes unexpectedly.
- ASoC: sta32x: add platform data definition
Add a structure for platform specific configuration and use it,
thereby removing a few FIXMEs which marked hard-coded values.
- ASoC: sta32x: preserve coefficient RAM
The coefficient RAM must be saved in a shadow so it can
be restored when the codec is powered on using
regulator_bulk_enable().
- ASoC: sta32x: Write the register default value to cache for reserved registers
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched.
codec->hw_read is broken now.
Here we use below trick to avoid writing to reserved registers while resume.
Write the register default value to cache for reserved registers,
so the write to the these registers are suppressed by the cache
restore code when it skips writes of default registers.
- ASoC: sta32x: Set reg_cache_default to sta32x_regs
- ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
This patch includes below small fixes:
1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove().
2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(),
as we will call regulator_bulk_free() in sta32x_remove().
3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove.
The i2c core will set the clientdata to NULL.
Johannes Stezenbach <js@sig21.net>
- ASoC: sta32x: shortcut the for loop to get ir and mcs
There is exactly one match or no match at all during the for loop iteration,
thus we can break from the for loop once a match is found.
- ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails
- ASoC: STA32x: Preserve reserved register bits
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched. It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.
- ASoC: STA32x: Add mixer controls for biquad coefficients
The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1. The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).
These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.
- ASoC: add STA32X codec driver
[zonque@gmail.com: transform to new ASoC structure]

SoC Codec STAC9766

- ASoC: Fix reg_cache_size for stac9766
reg_cache_size is supposed to be the number of elements in the register cache,
not the size in bytes.

SoC Codec SigmaDSP Firmare Loader

- ASoC: SigmaDSP: Add regmap support
Add support for loading the SigmaDSP firmware using regmap. This allows us
to transparently use SPI or I2C as the transport protocol on devices which
support them.
For now we keep the old I2C support since we have one user of this which is not
straight forward to convert to regmap, due to variable length registers.
- ASoC: SigmaDSP: Move private structs and functions to C file
Move the structs and functions only used by SigmaDSP firmware loader itself
from the header to the C file.
- ASoC: SigmaDSP: Provide diagnostic error messages
Provide some error messages when loading the firmware fails, so it is possible
to diagnose the reason for the failure.

SoC Codec TI sn95031

- ASoC: Remove needless codec->dapm.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: sn95031: Do not use static variable for channel_index
No reason to use static variable for channel_index.
- ASoC: Drop exporting sn95031_get_mic_bias
sn95031_get_mic_bias() is not used outside this driver
and it is a static function now.
Thus drop exporting sn95031_get_mic_bias.
- ASoC: Staticize sn95031_dais
- ASoC: sn95031: Staticize sn95031_pcm_hw_params
- ASoC: sn95031: Fix the logic to find free channel
In the case of no free channel available,
current implementation returns 0 instead of negative errno.
This patch fixes the logic to return -EINVAL if no free channel available.
- ASoC: sn95031: decorate function with __devexit_p()
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
Fix this issue in codecs sn95031.
- sound: Add delay.h to sound/soc/codecs/sn95031.c
This is further fallout from delay.h removal from asm/apic.h and asm/dma.h:
ca444564a947: x86: Stop including <linux/delay.h> in two asm header files
Which caused this build failure:
sound/soc/codecs/sn95031.c: In function ‘sn95031_get_mic_bias’:
sound/soc/codecs/sn95031.c:153:2: error: implicit declaration of function ‘msleep’ [-Werror=implicit-function-declaration]
LKML-Reference: <20110325152014.297890ec@endymion.delvare>
- ASoC: Use data based init for sn95031 DAPM
- ASoC: sn95031: fix the amic tlv scale
The tlv scale is defined as (min, step, mute). The mute is not supported here so
put the value to 0
- ASoC: sn95031: fix the DMIC path routing
This patch makes the DMIC dynamically connect to TX Mux, earlier code had
erroneously made this as static path
- ASoC: sn95031: make playback rails depend on actual pins they control
This patch makes the codec playback rails (headset and speaker) depend on
actual pins they control. This enables better power management of the codec
- ASoC: mid-x86: Use the soc-jack apis for jack type detection
This patch modifies the mfld_machine to use the new jack apis for adding the
voltage zones for jack type detection. It also modifed TI sn95031 codec driver
to use these new apis
- ASoC: sn95031: Add support for reading mic bias
This patch adds support to read the mic bias voltage
when a jack is inserted. It uses ADC to measure.
- ASoC: sn95031: Add jack support in the codec
This patch adds support for jack detection and reporting in the codec
It however is not fully functional as it doesn't measure adc to figure
out what got inserted which will be added later
- ASoC: sn95031: add capture support
This patch adds the support for capture path in sn95031 codec.
This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs
are connected through a MUX to ADC. The TX paths can be assigned to any of the
ADCs or DMICs.

SoC Codec TLV320AIC23

- ASoC: Convert tlv320aic23 to devm_kzalloc()
- ASoC: tlv320aic23: Clear TLV320AIC23_MS_MASTER bit for slave mode
According to the datasheet:
Digital Audio Interface Format (07h) register:
BIT6: Master/slave mode
0: Slave
1: Master
Current code sets TLV320AIC23_MS_MASTER bit for master mode,
but does not clear it for slave mode.
- ASoC: tlv320aic23: convert to soc-cache
- ASoC: Consolidate use of controls with custom get/put function
Use the macros for controls require custom get/put function.
This is to make sure that the soc_mixer_control is used
consistently among the drivers.
- ASoC: Use data based init for tlv320aic23 DAPM

SoC Codec TLV320AIC26

- ASoC: Convert tlv320aic26 to devm_kzalloc()
- audio: tlv320aic26: fix PLL register configuration
The current PLL configuration code for the tlc320aic26 codec appears to assume a
hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS
API for the calculation.
Tested using a MityDSP-L138 platform providing a 24.576 MHz clock.

SoC Codec TLV320AIC3X

- ASoC: Convert tlv320aic3x to devm_kzalloc()
- ASoC: Remove conditional I2C usage from tlv320aic3x driver
The driver only supports I2C so doesn't need to do things conditionally.
- ASoC: tlv320aic3x: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: tlv320aic3x: Clear BIT_CLK_MASTER and WORD_CLK_MASTER bits for for slave mode
According to the datasheet:
Page0 / Register8: Audio Serial Data interface Control Register A
BIT 7: Bit Clock Directional Control
0: Bit clock is an input (slave mode)
1: Bit clock is an output (master mode)
BIT 6: Word Clock Directional Control
0: Word clock is an input (slave mode)
1: Word clock is an output (master mode)
Current code sets BIT_CLK_MASTER and WORD_CLK_MASTER bits for master mode,
but does not clear these bits for slave mode.
- ASoC: tlv320aic3x: Convert codec->hw_read to snd_soc_read
codec->hw_read is broken now, let's covert to snd_soc_read.
- ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget
We really should be doing this in the core, not in a driver...
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
- ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models
Save model information in driver_data so we can simplify the implementation.
- ASoC: Remove unused "control_data" field of struct aic3x_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: omap: Update e-mail address of Jarkko Nikula
My gmail account got disabled and I'm not going to reopen it.
- ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.
- ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state
TLV320AIC33, TLV320AIC34 and I believe others too in this family have some
hw bugs that cause that analogue and digital VDD supplies remain leaking
up to a few mA of current after certain use cases even the hw blocks inside
codec are driven to off.
Highest leakages occur after using the bypass paths inside codec but it
is possible to get smaller leakages just by toggling mute switches in
unused audio paths (i.e. no DAPM changes) while codec is on due another
active audio path.
While some cases are able to workaroud by making sure that e.g. output mixer
switches are muted before powering down the output stage this doesn't help
all the cases.
Therefore use the software reset command to clear possible leakage currents
since that works in every cases and affects only this codec instance. Only
drawback is that now cache sync is required everytime when codec bias comes
out from bias off state, not only when supply regulators were off.
- ASoC: tlv320aic3x: Don't sync first two registers from register cache
There is no need to sync first two registers from cache to hw after a reset.
First one is used to select page for register access and this driver is
normally accessing page 0 only. Second one does a software reset which is
obviously unneeded after hardware or previous software reset command.
- ASoC: Fix wrong data type access in a few codec drivers
Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol")
changed the control private data type that is passed to snd_soc_cnew when
creating dapm mixer and mux controls. Commit did not update a few codec
drivers that are using their own put callbacks and thus are accessing a
wrong data type.
Tested-by: Stephen Warren <swarren@nvidia.com>

SoC Codec TLV320DAC33

- ASoC: Convert tlv320dac33 to devm_kzalloc()
- ASoC: tlv320dac33: Convert to table based init
- ASoC: tlv320dac33: Add guarding parentheses to macros
Put parentheses around macro argument uses. This avoids pitfalls
for the programmer, where the argument expansion does not give the
expected result, for example:
SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1);
- ASoC: tlv320dac33: Update e-mail address
- ASoC: tlv320dac33: Lower the OSC calibration time
To get correct calibration, we can decrease the time
needed for the OSC to calibrate itself.
With this change we can save ~15ms in the OSC
calibration phase.
- ASoC: tlv320dac33: Move codec power up to DAPM
Move the codec power on (in reg 0x01, bit 4) from
set_bias_level:SND_SOC_BIAS_ON to a DAPM supply.
In this way we can be sure, that all the things within
the codec is powered before the external amp is
going to be enabled.
- ASoC: tlv320dac33: Restore L/R DAC power control register
Register 0x40, 0x41 need to be restored after power up, since
it contains gain related fields, which affects playback volume.
- ASoC: tlv320dac33: Fix inconsistent spinlock usage
The lock is used within the interrupt handler.
Correct the spinlock usage, and use irqsave/irqrestore
flavour of spin_lock/unlock.
- ASoC: tlv320dac33: add MODULE_DEVICE_TABLE
The device table is required to load modules based on modaliases.

SoC Codec TPA6130A2

- ASoC: Convert tpa6130a2 to devm_kzalloc()
- ASoC: tpa6130a2: Remove model_id from platform data
The model_id is no longer needed within the platform_data
for the TPA driver since the model of TPA specified
with the device name (tpa6130a2/tpa6140a2).
Also update rx51 (the only affected user) to use the device name rather
than platform data.
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
- ASoC: tpa6130a2: Model support cleanup
Use the device name and driver_data to identify
the TPA model supported by the driver.
Board files should use either "tpa6130a2" or
"tpa6140a2" as device name to specify the model
in used on the specific board.
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
- ASoC: tpa6130a2: Remove obsolete cleanup for clientdata
The i2c core will clear the clientdata pointer automatically,
we don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
- ASoC: tpa6130a2: Update e-mail address

SoC Codec TWL4030

- ASoC: twl4030: Convert to table based init
- ASoC: Consolidate use of controls with custom get/put function
Use the macros for controls require custom get/put function.
This is to make sure that the soc_mixer_control is used
consistently among the drivers.
- MFD: twl4030-audio: Rename platform data
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver
To avoid breakage, change all depending drivers, files
to use the new types.
- MFD: twl4030-codec -> twl4030-audio: Rename the driver
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
- mfd: Use mfd cell platform_data for twl4030 codec cells platform
bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
- mfd: mfd_cell is now implicitly available to twl4030 drivers
The cell's platform_data is now accessed with a helper function;
change clients to use that, and remove the now-unused data_size.
- mfd: Use mfd cell platform_data for twl4030 codec cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
- ASoC: Fix spacing in MAX8950

SoC Codec TWL6040

- ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
The DL1 PDM interface adds a little gain depending on the output device.
Add a method to retrieve the gain value for machine driver usage.
- ASoC: twl6040: Request core to inline the DAPM sequence
We need to have as less time between McPDM shutdown,
and power down of the DAC on the twl6040 codec as possible.
Request core to ignore the pmdown_time for the playback
stream.
Backround: with the McPDM protocol we are sendning not only
the pure audio stream, but OMAP McPDM also transmits
additional information (for example offset cancellation).
If McPDM is stopped prior to the DAC this information will
be not sent to the codec, which can result noise rendered
by the twl6040 codec.
- ASoC: twl6040: Rename the Earphone Driver event handler
Since the event handler is only used by the Earphone Driver, it is better
to rename it from twl6040_power_mode_event to twl6040_ep_drv_event.
- ASoC: twl6040: Change event ordering for Earphone driver
It is better to switch HS Power Mode (if it was in low power mode) before
we enable the Earpiece driver. The switched off EP driver can filter out
noise coming from the Low Power to High Performance transition on the
HSL DAC.
- ASoC: twl6040: Remove PLL usage restrictions
There is no limitation dictated by outputs or inputs regarding to the
selected PLL (LP/HP).
Remove the checks for this, and allow all path with any PLL configuration.
- ASoC: twl6040: Remove Capture restriction for 17.64MHz sysclk
Capture is supported in all PLL configuration.
- ASoC: twl6040: Workaround for headset DC offset caused pop noise
Both Headset DAC need to be turned on/off at the same time before
any of the output drivers are enabled (HS Left/Right, Earpiece).
Move the HS DAC enable code to sequenced DAPM_SUPPLY, and attach
it to the DACs.
- ASoC: twl6040: Support for vibra output paths
twl6040 have two vibra output drivers.
They can be operated with audio stream coming through
the PDM interface (fifth channel).
The vibra outputs can be controlled via the input/FF
driver as well.
Selection between the two mode is implemented within
the codec driver, the input/FF driver can only operate if
the routing is set to "Input FF".
Changing from "Input FF" to "Audio PDM" mode is protected
as well: The switchin can only be done, if there is no
running effect from the input/FF.
- ASoC: twl6040: Convert to table based init
- ASoC: twl6040: Warn user in twl6040_put_volsw for error case
Let the user know, that the callback has been called with unexpected
register parameter.
- ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code
The ASoC core now have one callback function, which can handle
single, and double register mixer controls.
- ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger
Avoid using the mc->rreg to identify the 2r type of gain control.
Introduce a variable to track this.
This change is needed to avoid breakage with the upcoming volsw volsw_2r
merger.
- ASoC: twl6040: Simplify custom get_volsw callback
The custom get_volsw does not need to call any core get_volsw calls,
since we are returning the shadow values for the gains.
Return -EINVAL in the unlikely event, if the function has been called
for unhandled control. This way we can remove one check in the code.
- ASoC: Consolidate use of controls with custom get/put function
Use the macros for controls require custom get/put function.
This is to make sure that the soc_mixer_control is used
consistently among the drivers.
- ASoC: twl6040: Simplify custom put_volsw callback
Return -EINVAL in the unlikely event, if the function has been called
for unhandled control. This way we can remove one check in the code.
- ASoC: twl6040: Simplify code in out_drv_event for pending work check
Instead of checking, if the work is pending, it is safer to cancel
the pending work, or wait till the scheduled work finishes.
This way we can avoid modifying the variables used by the work
function.
Since we know that no work is pending, we can remove the two additional
checks in POST_PMU, and PRE_PMD for non pending works.
- ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event
None of the driver handled by out_drv_event have it's power
bit shifted by 3.
Remove the case for shift 3, and also add comment for the cases.
- ASoC: twl6040: correct loop counters for HS/HF ramp code
The Headset gain range is 0 - 0xf (4 bit resolution)
The Handsfree gain range is 0 - 0x1d (5 bit resolution,
0x1e, and 0x1f values are invalid)
- ASoC: twl6040: One workqueue should be enough
It is a bit overkill to have three (3) separate
workqueue for a single driver.
We can manage things with one workqueue nicely.
- ASoC: twl6040: No need to change delay during HF ramp
The Handsfree gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain
is -26dB.
- ASoC: twl6040: No need to change delay during HS ramp
The Headset gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x0 raw at
one end of the range, and not in the middle.
- ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data
The delayed_work named 'delayed_work' is for the headset detection,
so move it to the twl6040_jack_data struct.
- ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF
The delayed works for the output can be moved within the
twl6040_output struct (from the twl6040_data) to be better
organized.
- ASoC: twl6040: Combine the custom volsw get, and put functions
We can manage with one set of get, and put function for the gain
controls we need to handle with custom code due to the shadowing
of the register.
For both get, and put function we can call decide based on the
mc->rreg value, if we need to call the volsw, or the vlosw_2r
variant (in 2r case rreg is not 0).
Handling of the shadow values are the same for both type of
controls.
- ASoC: twl6040: Rename pga_event to out_drv_event
This event handler is used with the OUT_DRV widgets.
The name pga_event was misleading.
- ASoC: twl6040: Function to fetch the TRIM values
Provide API to fetch the TRIM values (for machine drivers)
- ASoC: twl6040: Read the TRIM values from the chip
Update the reg_cache with values from chip regarding to TRIM.
- ASoC: twl6040: No need to read the INTID register
Since our irq handler has been called, it is granted, that
the reason was either PLUGINT, or UNPLUGINT.
The INTID register has been checked in the MFD part of
twl6040 driver (twl6040-irq.c).
We have no reason to read from chip again here.
- ASoC/MFD: twl6040: Combine bit definitions for Headset control registers
Use one set of defines for the HS bits, since they are identical in both
control register.
- ASoC: twl6040/sdp4430: Change legacy DAI name
Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to
be more intuitive.
- ASoC: twl6040: Support for AUX L/R output
AUX L/R outputs can be driver from the Handsfree PGA output.
- ASoC: twl6040: Use consistent names for Headset path
Use "Headset XYZ" for user visible controls, while the internal DAPM
widgets can use "HS XYZ".
In this way we can group the Headset related controls in UI
(alsamixer for example).
- ASoC: twl6040: Use consistent names for Handsfree path
Use "Handsfree XYZ" for user visible controls, while the internal DAPM
widgets can use "HF XYZ".
In this way we can group the Handsfree related controls in UI
(alsamixer for example).
- ASoC: twl6040: Earphone path correction
Fix the DAPM routing for the earphone path.
Convert the DAPM_SWITCH_E to DAPM_OUT_DRV_E, so we can have correct
power up, and down sequence for EP.
Introduce mute control (Earphone Playback Switch) for users to
enable/disable the EP path.
Note: the EP does not have it's own dedicated DAC. EP is connected to
HSL DAC.
- ASoC: twl6040: Introduce SW only shadow register
Software only shadow register to be used by the driver.
For example Earpiece path will need this shadow register.
- ASoC: twl6040: Remove strings "NULL" from DAPM route
Replace the string with plain NULL.
- ASoC: twl6040: Fix comments for register names
Change the register name strings in the comments for the
twl6040_reg table, so it is easier to search for specific
register.
This is cosmetic change.
Before we had for example:
TWL6040_REG_HSLCTL as register definition.
At the register table we had:
TWL6040_HSLCTL
Searching for TWL6040_HSLCTL resulted no hits.
While if we look for REG_HSLCTL, we can find the places
the register has been used.
- ASoC: twl6040: Lower the power on gain values at startup
The default gains on outputs/inputs are set to 0dB.
This is fixing the pop noise issue at the first playback, which
caused by the wrong starting point of the ramp code.
The ramp code for the outputs expects the gains to be in
their lowest configuration in order to be effective.
After the playback stops, the ramp code takes care of
ramping down the gains to their minimum.
- ASoC: twl6040: Correct supported number of playback channels
twl6040 supports 5 playback, and 2 capture channels
- ASoC: twl6040: Fix the number of channels for vibra
Only mono audio can be used for vibra (DL4 channel).
- ASoC: twl6040: Use chip defaults in the initial reg_cache
Reset the twl6040_reg array to hold the chip default values.
The only changed values were for the microphone input selection.
Select no input for the microphones in the twl6040_init_chip function.
- ASoC: twl6040: Chip initialization cleanup
There is no need to write to the vio registers at probe time, since most
them either read only, or shared with MFD or not used.
On the other hand it is a good idea to updated the ASICREV register in
the cache at this time.
After power up we need to restore some registers. Clean up the list to
contain only the registers we are going to restore.
- ASoC: twl6040: Add back support for legacy mode
The legacy mode has been accidentaly removed by commit:
ASoC: twl6040: add all ABE DAIs
Add back the twl6040-hifi dai.
- ASoC: twl6040: No need to convert the PLL ID
Since the PLL handling has been simplified, and
rebased on 0, there is no longer need for converting
the PLL ID.
- ASoC: twl6040: Configure PLL only once
Avoid configuring the PLL several times during audio startup.
We can configure the PLL at prepare time with parameters collected
earlier hw_param, and set_dai_sysclk calls.
- ASoC: twl6040: Simplify sample rate constraint handling
We can manage the sample rate constraints without the need
to maintain a variable and a pointer.
This simplifies the handling of the constraint, and makes it
more robust.
- ASoC: twl6040: Move PLL selection to codec driver
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.
- ASoC: twl6040: Use neutral name for power mode text/enum
Change the variable names to be neutral (not refering to HS).
This will ease up the introduction of PLL selection, which
going to use the same enum strings.
- ASoC: twl6040: Do not use wrapper for irq request
The twl6040_request_irq/free_irq inline functions are going
to be removed, so replace them with direct calls.
The irq number is provided by the core driver via resource.
Reviewed-by: Felipe Balbi <balbi@ti.com>
- ASoC: twl6040: Configure ramp step based on platform
Enable ramp down/up step to be configured based on
platform.
- ASoC: twl6040: set default constraints.
Set default sysclk constraints to high performance mode.
- ASoC: twl6040: Remove pll and headset mode dependency
Remove dependency between pll (hppll, lppll) and headset power
mode (low-power, high-performance), as headset power mode can
be used with any pll.
A new control is created to allow headset power mode configuration
from userspace. Changing headset power mode during earpiece related
usecases is not propagated down to the codec as earpiece requires
HS DAC in HP mode.
- ASoC: twl6040: Support other sample rates in constraints.
Add other supported sample rates to LP and HP modes.
- ASoC: twl6040: add all ABE DAIs
Add all DAIs to fully support OMAP4 ABE.
- ASoC: twl6040: Convert into TWL6040 MFD child
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
- MFD: twl4030-audio: Rename platform data
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver
To avoid breakage, change all depending drivers, files
to use the new types.
- ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB.
- ASoC: twl6040 - fix LINEGAIN volume control
Fix the TWL6040 LINEGAIN volume control to match the TRM.
- ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails
- ASoC: Staticise twl6040_hs_jack_report()
It's an internal function so shouldn't be exported (as sparse points
out).

SoC Codec WL1273

- ASoC: Remove unneeded mutex_init in wl1273_probe()
Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support",
snd_soc_register_codec() now does all the codec list and mutex init.
Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more.
- mfd: Use mfd cell platform_data for wl1273 cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
- mfd: mfd_cell is now implicitly available to wl1273 drivers
The cell's platform_data is now accessed with a helper function;
change clients to use that, and remove the now-unused data_size.
- [media] ASoC: WL1273 FM radio: Access I2C IO functions through pointers
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.
Also update the year in the copyright statement.

SoC Codec WM1250-EV1

- ASoC: Convert wm1250-ev1 driver to use devm_kzalloc()
- ASoC: Set idle_bias_off for WM1250 EV1
The WM1250 EV1 is functionally digital in a system (the analogue I/O
is either ground referenced or always powered) so flag it as idle_bias_off.
- ASoC: Add platform data for WM1250 EV1 GPIOs
The WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.
- ASoC: Correct revision display for WM1250-EV1 module
The hardware documentation uses revision numbers starting at 1.
- ASoC: Fix warning in WM1250-EV1 driver
- ASoC: Parse board ID/revision information from WM1250-EV1 board
The WM1250-EV1 board has an ID chip on it, check the board ID and display
the board revision during startup.
- ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT.
- ASoC: Fix mis cherry-pick of wm1250-ev1 driver
- ASoC: Add initial WM1250-EV1 Springbank audio I/O module driver
The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas
reference platform provides a simple audio I/O with an independant clock
domain, intended to simulate cellular modem and bluetooth subsystems
within the platform.
The card supports some limited GPIO based control but this is currently not
implemented.

SoC Codec WM2000

- ASoC: Convert WM2000 into a standard CODEC driver
We've been able to handle external amps for a while now.
- ASoC: Convert wm2000 to use regmap API
The driver wasn't even using the ASoC common code.
- ASoC: Remove unused struct wm2000_setup_data
- ASoC: Convert WM2000 to devm_kzalloc()
- ASoC: Move WM2000 to dev_pm_ops
There's a general move to use dev_pm_ops rather than bus specific functions
in order to facilitate work on the PM core. Do this conversion to WM2000.
The driver ought to be updated to work better in a multi-component model
but the mechanical conversion ensures that we avoid blocking PM core work
until that happens.

SoC Codec WM5100

- ASoC: Enable ASoC register map dump for some regmap CODECs
It's still useful to be able to poke around in the register map at
runtime.
- ASoC: Make WM5100 tone generator widgets signal generators
- ASoC: Use devm_kzalloc() in wm5100
- ASoC: Remove WM5100 DSP memory windows from register default data
They're all volatile so shouldn't have defaults and as we've got pages
into the DSP memory the registers themselves aren't that useful - a
further patch adding support for the DSPs will provide direct diagnostic
access to the DSP memories.
- ASoC: Move WM5100 platform data based setup into I2C probe
Get things configured as early as possible, especially useful for the
GPIOs which might be useful anyway.
- ASoC: Convert WM5100 gpiolib support to direct regmap API usage
- ASoC: Move most WM5100 resource allocation to I2C probe
More standard Linuxish.
- ASoC: Mark WM5100 MISC CONTROL as readable
- ASoC: Need to convert wm5100 cache sync to direct regmap usage too
ASoC knows nothing about the cache now.
- ASoC: Convert wm5100 to direct regmap API usage
- ASoC: Fix return value of wm5100_gpio_direction_out()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.
- ASoC: Update WM5100 accessory detection for revision A
- ASoC: Implement WM5100 accessory detection support
The WM5100 includes an advanced, low power, accessory detect subsystem
capable of detecting both accessory presence and button presses while
the device is in an ultra low power mode. Implement initial support for
this.
- ASoC: Add missing default for WM5100 Clocking 1
- ASoC: Fix typo in 24.576MHz rate in WM5100
- ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path
- ASoC: Add missed BCLK rate to WM5100 driver
Reported-by: Axel Lin <axel.lin@gmail.com>
- ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100
Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping
fully power down connected devices when idle.
- ASoC: Remove needless codec->dapm.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: Add WM5100 driver
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.

SoC Codec WM8350

- ASoC: Convert WM8350 to table based DAPM and control init
- ASoC: Convert WM8350 to devm_kzalloc()
- ASoC: Replace remaining use of *_volsw_2r with *_volsw
The snd_soc_*_volsw_2r functionality has been merged to
*volsw callbacks.
Few places still used the get, or put variant of volsw_2r,
replace those with the corresponding *_volsw.

SoC Codec WM8400

- ASoC: Convert WM8400 to table based DAPM and control init
- ASoC: Convert WM8400 to devm_kzalloc()
- ASoC: Convert wm8400 MICBIAS to a supply widget
- ASoC: wm8400: Fix setting Fout clock divider for FLL Control 4
What we want here is to clear the WM8400_FLL_OUTDIV_MASK bits then
OR with factors.outdiv.
- ASoC: wm8400: Fix wrong bit setting for WM8400_POWER_MANAGEMENT_2
If (fakepower & ((1 << WM8400_INMIXR_PWR) | (1 << WM8400_AINRMUX_PWR)))
is false, we should clear WM8400_AINR_ENA bits instead of WM8400_AINL_ENA.
- mfd: Use mfd cell platform_data for wm8400 cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- asoc: wm8400-codec: Use mfd_data instead of driver_data
Use mfd_data for passing information from mfd drivers to soc
clients. The mfd_cell's driver_data field is being phased out.
Clients that were using driver_data now access .mfd_data
via mfd_get_data().

SoC Codec WM8510

- ASoC: Convert WM8510 to table based DAPM and control init
- ASoC: Remove unused -codec from Wolfson device driver names
Devices that aren't MFDs don't need to distinguish this.
- ASoC: wm8510: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Add device tree binding for WM8510

SoC Codec WM8523

- ASoC: Remove unneeded hw_write initialisation in wm8523
It is not required after commit 8d50e447
"ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs"
- ASoC: Convert WM8523 to table based control and DAPM initialization
- ASoC: Add device tree binding for WM8523
- ASoC: Remove -codec from WM8523 driver name
It's redundant to specify it.

SoC Codec WM8580

- ASoC: Fix return value of wm8580_set_sysclk()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.
- ASoC: Convert WM8580 to table based DAPM and control init
- ASoC: wm8580: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Replace remaining use of *_volsw_2r with *_volsw
The snd_soc_*_volsw_2r functionality has been merged to
*volsw callbacks.
Few places still used the get, or put variant of volsw_2r,
replace those with the corresponding *_volsw.
- ASoC: Add device tree support for WM8580
- ASoC: Remove redundant -codec from WM8580 driver name

SoC Codec WM8711

- ASoC: Convert WM8711 to table based control init
- ASoC: Leave input audio data bit length settings untouched in wm8711_set_dai_fmt
Current implementation in wm8711_set_dai_fmt always clear BIT[3:2]
(the Input Audio Data Bit Length Select) of WM8711_IFACE(07h) register.
Input Audio Data Bit Length Select bits are set by wm8711_hw_params,
we should leave BIT[3:2] untouched in wm8711_set_dai_fmt.
- ASoC: wm8711: Fix wrong mask for setting input audio data bit length select
The Input Audio Data Bit Length Select is controlled by BIT[3:2] of
WM8711_IFACE(07h) register.
Current code incorrectly masks BIT[1:0] which is for Audio Data Format Select.
- ASoC: wm8711: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Remove references to linux@wolfsonmicro.com
- ASoC: wm8711: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: wm8711: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Add device tree binding for WM8711
- ASoC: Remove some more redundant -codecs from driver names
- ASoC: Use data based init for wm8711 DAPM

SoC Codec WM8727

- ASoC: Remove unused -codec from Wolfson device driver names
Devices that aren't MFDs don't need to distinguish this.

SoC Codec WM8728

- ASoC: Convert WM8728 to table based control init
- ASoC: Add device tree binding for WM8728
- ASoC: Remove some more redundant -codecs from driver names
- ASoC: Use data based init for wm8728 DAPM

SoC Codec WM8731

- ASoC: Use table based init for wm8731_snd_controls
- ASoC: Ensure WM8731 register cache is synced when resuming from disabled
- ASoC: wm8731: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Add device tree binding for WM8731
Tested with the famous "hey, look! this compiles" test plan.
Acked by: Grant Likely <grant.likely@secretlab.ca>
- ASoC: Manage WM8731 ACTIVE bit as a supply widget
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
- ASoC: wm8731: fix wm8731_check_osc() connected condition
The crystal oscillator is only enabled if the WM8731_SYSCLK_XTAL master clock
is specified. Fix the connected() struct snd_soc_dapm_route function to take
this into account. Oscillator is not enabled on machine that need it otherwise.
Machine drivers have to make sure that they use the proper SYSCLK value.
- ASoC: Convert WM8731 to table based DAPM setup

SoC Codec WM8737

- ASoC: Add device tree binding for WM8737

SoC Codec WM8741

- ASoC: Convert WM8741 to table based DAPM and control init
- ASoC: Convert WM8741 to devm_kzalloc()
- ASoC: wm8741: Use snd_soc_cache_sync to sync reg_cache with the hardware
- ASoC: wm8741: Fix setting interface format for DSP modes
According to the datasheet:
Format Control (05h)
BITS[3:2]
FMT[1:0] Audio data format selection
00 = right justified mode
01 = left justified mode
10 = I2S mode
11 = DSP mode
BIT[4] LRP Polarity selec for LRCLK/DSP mode select
0 = normal LRCLK poalrity/DSP mode A
1 = inverted LRCLK poarity/DSP mode B
For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003.
For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013.
- ASoC: Fix setting update bits for WM8741_DACRMSB_ATTENUATION
After checking the code and datasheet, I think what we want in the second
snd_soc_update_bits call is to update WM8741_DACRMSB_ATTENUATION register
instead of WM8741_DACRLSB_ATTENUATION.
- ASoC: Add device tree binding for WM8741
- ASoC: Add SPI support for WM8741
- ASoC: Refactor WM8741 regulator handling into CODEC generic code
No meaningful runtime impact but is more in line with other CODECs and
will support further work.
- ASoC: Remove some more redundant -codecs from driver names

SoC Codec WM8750

- ASoC: Convert WM8750 to table based DAPM and control init
- ASoC: Convert WM8750 to devm_kzalloc()
- ASoC: wm8750: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: wm8750: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Remove redundant -codec from WM8750
- ASoC: Add device tree binding for WM8750
- ASoC: Fix compile warning in wm8750.c
sound/soc/codecs/wm8750.c:784:2: warning: missing braces around initializer
sound/soc/codecs/wm8750.c:784:2: warning: (near initialization for ‘wm8750_spi_ids[2].name’)
It's because struct spi_device_id.name is a char array, not a pointer,
while the driver initializes explicitly with 0.
- ASoC: Terminate WM8750 SPI device ID table
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: Fix typo in wm8750 spi_ids
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: Fix SPI driver binding for WM8987
As we had no id_table only the driver name would be matched against
meaning that WM8987 devices wouldn't be bound.

SoC Codec WM8753

- ASoC: Convert WM8753 to table based DAPM and control init
- ASoC: wm8753: Skip noop reconfiguration of DAI mode
This patch makes it possible to set DAI mode to its currently applied
value even if codec is active. This is necessary to allow
aplay -t raw -r 44100 -f S16_LE -c 2 < /dev/urandom &
alsactl store -f backup.state
alsactl restore -f backup.state
to work without returning errors. This patch is based on a patch sent
by Klaus Kurzmann <mok@fluxnetz.de>.
- ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC
Current code set update bits for WM8753_LDAC and WM8753_RDAC twice,
but missed setting update bits for WM8753_LADC and WM8753_RADC.
I think it is a copy-paste bug in commit 776065
"ASoC: codecs: wm8753: Fix register cache incoherency".
- ASoC: Add device tree binding for WM8753
- ASoC: Remove unneeded -codec from WM8753 driver name
- ASoC: codecs: wm8753: Fix DAI mode switching
The wm8753 codec supports switching between different DAI modes.
The current drivers tries to implement this by changing the DAI driver at
runtime. But to properly work this would require support from the ASoC core.
So this patch takes a different approch on how the DAI mode switching is
implemented.
The only difference, from a driver point of view, between the different DAI modes
is how to program the DAI format to the hardware. So what this patch is, it
stores the current format for each DAI in the drivers private struct and when
the DAI mode is changed the format gets simply reprogrammed according to the
new DAI mode.
Futhermore this patch restricts the changing of the DAI format to when the
codec is inactive.

SoC Codec WM8770

- ASoC: Convert wm8770 to devm_kzalloc()
- ASoC: Make WM8770 SPI usage unconditional
The device only supports SPI.
- ASoC: Add device tree binding for WM8770

SoC Codec WM8776

- ASoC: Convert WM8776 to devm_kzalloc()
- ASoC: Remove unused variable in wm8776 driver
- ASoC: Convert wm8776 to table based control and DAPM init
- ASoC: wm8776: add missing break in sample size switch
Broken in commit d1dc698a54259cb454284456483b45f67c865cf8
- ASoC: wm8776: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: support all possible sample rates in the WM8776 driver
The WM8776 supports a continuous range of sample rates rather than
discrete values and supports a wider range of sample rates on the
playback path than is currently supported. Update the constraints on
the DAIs to reflect this.
- ASoC: support sample sizes properly in the WM8776 codec driver
Use snd_pcm_format_width() to determine the sample size, instead of
checking specify sample formats and assuming that those are the only
valid format.
This change adds support for big-endian architectures (which use the _BE
formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE).
[Fixed single letter variable name legibility problem -- broonie]
- ASoC: Add device tree binding for WM8776
- ASoC: Remove redundant -codec from WM8776 driver name

SoC Codec WM8782

- ASoC: wm8782: Add __devexit_p at necessary place
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
We have __devexit annotation for wm8782_remove(), thus add __devexit_p at
necessary place.
- ASoC: add WM8782 ADC Codec Driver
[zonque@gmail.com: transform to new ASoC structure]

SoC Codec WM8804

- ASoC: Convert WM8804 to table based control init
- ASoC: Add device tree binding for WM8804
- ASoC: WM8804 does not support sample rates below 32kHz
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>

SoC Codec WM8900

- ASoC: Convert WM8900 to table based DAPM and control init
- ASoC: Remove unused -codec from Wolfson device driver names
Devices that aren't MFDs don't need to distinguish this.
- ASoC: Convert wm8900 MICBIAS to a supply widget
- ASoC: wm8900: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: wm8900: Fix the mask defines
Now we have done bitwise NOT against the mask bits for the defines of
WM8900_REG_CLOCKING1_BCLK_MASK,
WM8900_REG_CLOCKING1_OPCLK_MASK and WM8900_LRC_MASK.
But we don't have the bitwise NOT against the mask bits for the defines of
WM8900_REG_CLOCKING2_DAC_CLKDIV,
WM8900_REG_CLOCKING2_ADC_CLKDIV and WM8900_REG_DACCTRL_AIF_LRCLKRATE.
It is error prone to mix the inconsistent meaning for different mask defines.
So lets make the defines for each mask to be corresponding to the bits
defines in datasheet. Don't add extra "bitwise NOT" to the defines.
- ASoC: wm8900: Fix wrong mask for setting DAC_CLKDIV/ADC_CLKDIV/LRCLK_MODE
After checking the datasheet, I think what we want to do here is to
clear the WM8900_REG_CLOCKING2_DAC_CLKDIV/WM8900_REG_CLOCKING2_ADC_CLKDIV/
WM8900_REG_DACCTRL_AIF_LRCLKRATE bits and then OR with div value.
- ASoC: wm8900: fix a memory leak if wm8900_set_fll fails

SoC Codec WM8903

- ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.
- ASoC: WM8903: Add of_match_table
This allows the device to be matched against the device tree using the
compatible flag directly, as is standard, rather than falling back to
matching .id_table against the non-vendor portion of the first compatible
property value.
- ASoC: Don't fail if we can't read the IRQ type in WM8903
If we fail to read the IRQ type from the interrupt controller don't
fail, just assume a value and solider on - we may fail later when we try
to request the IRQ but it's possible we'll succeed.
- ASoC: WM8903: Add device tree binding
Document the device tree binding for the WM8903 codec, and modify the
driver to extract platform data from the device tree, if present.
Based on work by John Bonesio, but significantly reworked since then.
- ASoC: WM8903: Get default irq_active_low from IRQ controller
If the WM8903 is hooked up to an interrupt, set the irq_active_low flag
in the default platform data based on the IRQ's IRQ_TYPE. Map IRQ_TYPE_NONE
(a lack of explicit configuration/restriction) to irq_active_low = false;
the previous default.
This code is mainly added to support device tree interrupt bindings,
although will work perfectly well in a non device tree system too.
Any interrupt controller that supports only a single IRQ_TYPE could
set each IRQ's type based on that restriction. This applies equally
with and without device tree. To cater for interrupt controllers
that don't do this, for which irqd_get_trigger_type() will return
IRQ_TYPE_NONE, the platform data irq_active_low field may be used
in systems that don't use device tree.
With device tree, every IRQ must have some IRQ_TYPE set.
Controllers that support DT and multiple IRQ_TYPEs must define the
interrupts property (as used in interrupt source nodes) such that it
defines the IRQ_TYPE to use. When the core DT setup code initializes
wm8903->irq, the interrupts property will be parsed, and as a side-
effect, set the IRQ's IRQ_TYPE for the WM8903 probe() function to read.
Controllers that support DT and a single IRQ_TYPE could arrange to
set the IRQ_TYPE somehow during their initialization, or hard-code
it during the processing of the child interrupts property.
- ASoC: WM8903: Remove conditionals checking pdata != NULL
The pdata pointer is now always valid. Remove any conditions that check
its validity.
This patch is mostly just removing an indentation level. One variable had
to be moved due to the removal of a scope, and one comment was split into
two. Viewing the patch with git show/diff -b will show that it's actually
very small.
Note that WM8903_MIC_BIAS_CONTROL_0 is now written unconditionally,
whereas it used to be written only if pdata was supplied. Since
defpdata.micdet_cfg = 0, this unconditional write simply echos the HW
defaults in the case where pdata is not supplied.
Based on work by John Bonesio, but significantly reworked since then.
- ASoC: WM8903: Fix platform data gpio_cfg confusion
wm8903_platform_data.gpio_cfg[] was intended to be interpreted as follows:
0: Don't touch this GPIO's configuration register
1..7fff: Write that value to the GPIO's configuration register
8000: Write zero to the GPIO's configuration register
other: Undefined (invalid)
The rationale is that platform data is usually global data, and a value of
zero means that the field wasn't explicitly set to anything (e.g. because
the field was new to the pdata type, and existing users weren't update to
initialize it) and hence the value zero should be ignored. 0x8000 is an
explicit way to get 0 in the register.
The code worked this way until commit 7cfe561 "ASoC: wm8903: Expose GPIOs
through gpiolib", where the behaviour was changed due to my lack of
awareness of the above rationale.
This patch reverts to the intended behaviour, and updates all in-tree users
to use the correct scheme. This also makes WM8903 consistent with other
devices that use a similar scheme.
WM8903_GPIO_NO_CONFIG is also renamed to WM8903_GPIO_CONFIG_ZERO so that
its name accurately reflects its purpose.
- ASoC: WM8903: Create default platform data structure
When no platform data is supplied, point pdata at a default platform
structure. This enables two future changes:
a) Defines the default platform data values in a single place.
b) There is always a valid pdata pointer, so some conditional code can
be simplified by a later patch.
Based on work by John Bonesio, but significantly reworked since then.
- ASoC: Move initial WM8903 identification and reset to I2C probe
Get control of the device earlier and avoid trying to do an ASoC probe
on a card that won't work.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Convert WM8903 to direct regmap API usage
Converting to an rbtree cache as regcache doesn't have a flat cache.
Since the top of the register map is fairly sparse this should be an
overall win.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Don't resync WM8903 register cache on reset
We only do this on initial power on so it's at best a waste of time as
the core will have already defaulted to the same values.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Use a normal cache sync for WM8903
The driver used to use a complicated method to sync the register cache
after having brought the bias level up to standby in resume due to the
use of the write sequencer to manage the initial power up. Now that we
don't use the write sequencer there is no need for this and we can just
use snd_soc_cache_sync() directly.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: WM8903 only supports I2C so don't ifdef it
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Use table based control init for WM8903
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Convert WM8903 to devm_kzalloc()
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: WM8903: Disallow all invalid gpio_cfg pdata values
The GPIO registers are 15 bits wide. Hence values, higher than 0x7fff are
not legal GPIO register values. Modify the pdata.gpio_cfg handling code
to reject all illegal values, not just WM8903_GPIO_NO_CONFIG (0x8000). This
will allow the later use of 0xffffffff as an invalid value in future device
tree bindings, meaning "don't touch this GPIO's configuration".
- ASoC: Convert WM8903 MICBIAS to a supply widget
Also rename it to MICBIAS to reflect the pin name and help any out of tree
users notice the change.
- ASoC: WM8903: Free IRQ on device removal
Without this, request_irq on subsequent device initialization fails, and
the codec cannot be used.
- ASoC: Fix wrong data type access in a few codec drivers
Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol")
changed the control private data type that is passed to snd_soc_cnew when
creating dapm mixer and mux controls. Commit did not update a few codec
drivers that are using their own put callbacks and thus are accessing a
wrong data type.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: WM8903: Fix Digital Capture Volume range
Increase the range of the Digital Capture Volume control to be 120 steps.
Each step is 0.75dB, and the range starts at -72dB, giving a max setting
of 18dB, which matches the latest datasheet, to the precision of the step
size.
- ASoC: WM8903: Implement DMIC support
In addition to the currently supported analog capture path, the WM8903
also supports digital mics.
The analog and digital capture paths are exclusive; a mux is present to
select the capture source.
Logically, the mux exists to select the decimator's input, from either
the ADC or DMIC block outputs. However, the ADC power domain also
includes the DMIC interface. Consequently, this change represents the
mux as existing immediately before the ADC, and selecting between the
Input PGA and DMIC block outputs.
An alternative might be to represent the mux in its correct location,
and associate the ADC power enable controls with both the real ADC, and
a fake ADC for the DMIC?
- ASoC: WM8903: HP and Line out PGA/mixer DAPM fixes
Update the headphone and line out mixers and PGAs use the same logical
set of register bits and sequencing as the speaker mixer/PGA.
This allows ALSA controls for mute and volume on headphone and line out
to operate correctly.
Per conversation on alsa-devel, earlier datasheets indicated that the
POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register
bits 0 and 4, and hence only one copy of those bits was programmed.
However, later datasheets corrected this.
[swarren: Applied same change to headphone widgets]
- ASoC: Convert WM8903 to table based DAPM setup
- ASoC: Warn if WM8903 platform data is used to enable microphone IRQ
The WM8903 interrupts are clear on read so if the WM8903 detection is
enabled from platform data when the IRQ is in use (rather than using a
direct signal from a GPIO) status may be lost during startup. Help users
spot this misconfiguration by adding a WARN_ON().
- ASoC: Use explicit sequence for WM8903 bias off
This makes no real difference compared to the write sequencer sequence
that was previously used but can run without a clock being provided.
Also remove the write sequencer support code as this was the last use
of it.
- ASoC: Don't use write sequencer to power up WM8903
The write sequencer sequencer sequence takes longer than is desirable
as it brings up a full playback path which is not required at this
point. Open coding the sequence cuts the startup time by two thirds.
- ASoC: Convert WM8903 bias management to use snd_soc_update_bits()
- ASoC: Actively manage WM8903 DC servo configuration
Explicitly cache the DC servo offsets for digital paths in the driver,
allowing them to be preserved over suspend and resume, and ensure that
we recalibrate analogue outputs paths when they are in use so that we
cover any changes in the input offset.
- ASoC: Fix WM8903 DAC mute default
The WM8903 register map does not mute the DAC by default at startup
so we need to explicitly do so.
- ASoC: Dynamically manage CLK_SYS in WM8903
- ASoC: Convert WM8903 to use PGA_S for output stage enables
This simplfies the code and slightly reduces the startup time.
- ASoC: Add support for AIF channel muxing on WM8903
- ASoC: Display WM8903 chip revision alphabetically
- ASoC: Remove redundant -codec from WM8903 driver name
It causes noisy -codecs to appear in things like .codec_name.
- ASoC: Correct definition of WM8903_VMID_RES_5K
- ASoC: WM8903: Fix mic detection enable logic
The mic detection HW should be enabled when either mic or short detection
is required, not when only both are required.
- ASoC: WM8903: Fix mic detection register definitions
* There is no hysteresis enable field in the current datasheet.
* Mic detection threshold field is only 2 bits wide.
- ASoC: Accept any logical value WM8903 GPIO set()
- ASoC: wm8903: Expose GPIOs through gpiolib
Also, update platform_data GPIO handling to have an explicit "don't
touch this pin" option.
Add #defines for the GPIO pin functions.

SoC Codec WM8904

- ASoC: Remove unused -codec from Wolfson device driver names
Devices that aren't MFDs don't need to distinguish this.
- ASoC: Convert wm8904 MICBIAS to a supply widget
- ASoC: WM8904: Set `invert' bit for Capture Switch
Set `invert' bit for Capture Switch. Otherwise analogue is muted when
Capture Switch is ON.
- ASoC: Remove unused "control_data" field of struct wm8904_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Add basic WM8918 support
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.

SoC Codec WM8915

- ASoC: Rename WM8915 to WM8996
For marketing reasons the part will be called WM8996. In order to avoid
user confusion rename the driver to reflect this.
- ASoC: Allow WM8915 BCLK calculation outside hw_params()
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.
- ASoC: Error out when FLL lock interrupt is not delivered on WM8915
When the FLL locks on the WM8915 an interrupt is generated. For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up. Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.
- ASoC: Suppress noop SYSCLK updates in WM8915
- ASoC: Use a lower detection rate when monitoring headphones on WM8915
We only need to increase the detection rate to maximum if we're monitoring
for button presses as the response times needed for user interaction there
are much lower.
- ASoC: Remove internally generated WM8915 supplies
DCVDD and MICVDD are intended to be (and almost always are) generated by
on-board LDOs which are transparently controlled by the driver so we
shouldn't really be requesting them from the regulator API. If the driver
is updated to support external supply of these then we will need to change
the way we handle this.
- ASoC: Support edge triggered IRQs for WM8915
Really this should be something the IRQ core can cope with for us but since
it doesn't currently do so (at least for threaded interrupts like this) do
so in the driver. This allows us to run with interrupt controllers that
only support edge triggered interrupts.
- ASoC: Add missing break in WM8915 FLL source selection
- ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK
- ASoC: Remove duplicate linux/delay.h inclusion.
It's enough to include linux/delay.h just once in
sound/soc/codecs/wm8915.c, so remove the duplicate.
- ASoC: Remove outdated FIXME from WM8915
Actually the current code is perfectly sensible given the hardware.
- ASoC: Use shared controls for input signal path in WM8915
Gives finer grained power management.
- ASoC: Support 24.576MHz MCLKs in WM8915
We can safely divide these down to within the supported SYSCLK range.
- ASoC: Move WM8915 FLL operations onto the CODEC
Since the WM8915 FLL is not tied to a particular audio interface move it
to a CODEC wide operation.
- ASoC: Add WM8915 CODEC driver
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.

SoC Codec WM8940

- ASoC: wm8940: Fix setting PLL Output clock division ratio
According to the datasheet:
The PLL Output clock division ratio is controlled by BIT[5:4] of
WM8940_GPIO register(08h).
Current code read/write the WM8940_ADDCNTRL(07h) register which is wrong.
- ASoC: wm8940: Fix mask for setting BCLKDIV
According to the datasheet:
BCLK is controlled by BIT[4:2] of WM8940_CLOCK(06h) register.
- ASoC: wm8940: Properly set codec->dapm.bias_level
Reported-by: Chris Paulson-Ellis <chris@edesix.com>
- ASoC: wm8940: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Remove unused "control_data" field of struct wm8940_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: wm8940: remove unnecessary if statements
removing unnecessary if(ret) checks
This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.

SoC Codec WM8958

- ASoC: Say how long short WM8958 DSP2 firmwares are
- ASoC: WM8958: correctly show firmware magic on mismatch
- ASoC: Don't restart an already running WM8958 DSP2
Don't want to upset the DSP.
- ASoC: Skip noop reconfiguration of WM8958 DSP2 algorithms
If we're setting the currently applied value for one of the DSP algorithm
configurations we can just skip all the handling as the control set is a
noop. This ensures we do not disrupt a running DSP.
- ASoC: fix wm8958-dsp2 printk format warnings
Fix printk format warnings in wm8958-dsp2.c:
sound/soc/codecs/wm8958-dsp2.c:103: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm8958-dsp2.c:111: warning: format '%d' expects type 'int', but argument 3 has type 'size_t'
sound/soc/codecs/wm8958-dsp2.c:144: warning: format '%d' expects type 'int', but argument 5 has type 'size_t'
- ASoC: Add WM8958 enhanced EQ support
DSP2 in the WM8958 can be used to support an upgraded EQ for use in
demanding applications.
- ASoC: Add WM8958 VSS support
With appropriate firmware the WM8958 can support Virtual Surround Sound or
VSS, widening the stereo audio image for improved user experience. Enable
support for this mode of operation when the appropriate firmware can be
loaded at runtime.
- ASoC: Refactor WM8958 DSP to support additional algorithms
In preparation for the addition of additional WM8958 algorithms
reorganise the code to make it easier to add such support later.
- ASoC: Support download of WM8958 MBC firmware
Allow userspace to supply an update to the ROM firmware. The firmware
request is non-blocking so userspace can load the firmware at its
leisure without delaying startup, the driver will begin using the
firmware the next time MBC is started after it has been supplied.
- ASoC: Handle startup sequencing of WM8958 DSP2 with deferred clocking
The DSP2 startup requires that the clock be enable so if we've deferred
clock startup we need to defer DSP2 startup
- ASoC: Factor WM8958 DSP2 handling into separate file
DSP2 on the WM8958 has a default ROM which provides a multi-band
compressor for enhanced performance on mobile devices but can also
support runtime download of alternative firmware. In preparation for
more exploiting this functionality refactor the code to split the
handling of DSP2 into a separate file.

SoC Codec WM8960

- ASoC: Convert WM8960 to devm_kzalloc()
- ASoC: wm8960: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Remove unused AUDIO_NAME define from WM8960
- ASoC: Remove I2C ifdefs from WM8960
The driver only supports I2C as the control interface.
- ASoC: Convert wm8960 MICBIAS to a supply widget
- ASoC: wm8960: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Remove unused "control_data" field of struct wm8960_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

SoC Codec WM8961

- ASoC: Use standard cache sync code in wm8961
We write the reset register with the default value so it should not be
mistakenly written.
- ASoC: Convert wm8961 to devm_kzalloc()
- ASoC: Remove I2C ifdefs from wm8961 driver
The driver only supports I2C so no need to conditionalise its use.
- ASoC: Fix comments for disabling amplifier and PGA
- ASoC: Convert wm8961 MICBIAS to a supply widget
- ASoC: wm8961: Convert codec->hw_read to snd_soc_read
codec->hw_read is broken now, let's covert to snd_soc_read.

SoC Codec WM8962

- ASoC: Fix return value of wm8962_gpio_direction_out()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.
- ASoC: Enable ASoC register map dump for some regmap CODECs
It's still useful to be able to poke around in the register map at
runtime.
- ASoC: Make WM8962 beep a signal generator
- ASoC: Convert WM8962 to devm_kzalloc()
- ASoC: Disable debounce on some WM8962 interrupts
Allow them to work when the device is unclocked.
- ASoC: Convert WM8962 to direct regmap usage
This initial conversion just moves the register init, regulator acquisition
and device verification out to the I2C probe(). Movement of other parts of
the driver like the GPIO and beep generation code will follow.
- ASoC: wm8962: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the arrays.
- ASoC: Manage thermal shutdown for WM8962
Disable the thermal shutdown circuits for headphone and speaker when the
relevant outputs are not enabled in order to save current in idle modes.
- ASoC: Disable MICBIAS and SYSCLK when stopping WM8962 accessory detection
They aren't needed any more. If machines need them for other purposes then
further changes will be required.
- ASoC: WM8962 accessory detection requires MICBIAS
Force MICBIAS on as well as SYSCLK as the WM8962 accessory detection can't
function without both. No point in making machine drivers manually enable
it.
- ASoC: Enable SYSCLK last when enabling WM8962 mic detection
Ensure everything is set up before we start detecting.
- ASoC: Start WM8962 FLL if SYSCLK is enabled
Since we have code to automatically manage the start and stop of the FLL
based on the SYSCLK widget if SYSCLK is already enabled and the FLL is
configured then we need to start it up.
- ASoC: Ensure we always delay for WM8962 FLL when starting from SYSCLK
- ASoC: Ensure the WM8962 oscillator and PLLs start up disabled
Since there is no current software control for these they would otherwise
be left enabled, consuming power.
- ASoC: Ensure WM8962 PLL registers are reset
The WM8962 has a separate software reset for the PLL registers. Ensure that
these are reset also on startup.
- ASoC: Remove direct register cache accesses from WM8962 driver
Also fix return values for speaker switch updates.
- ASoC: Fix a bug in WM8962 DSP_A and DSP_B settings
- ASoC: Convert WM8962 MICBIAS to a supply widget
A supply widget is generally clearer than a MICBIAS widget and a mic bias
is just a type of supply so use a supply widget for the MICBIAS. This also
avoids confusion with the routing when connected to multiple inputs.
- ASoC: Rename WM8962 DMIC widget to DMIC_ENA
Matches the register name and avoids confusion with board widgets.
- ASoC: Remove bitrotted wm8962_resume()
This functionality is now subsumed within the bias management, using the
standard cache management functionality, without assuming the cache type.
- ASoC: Provide more detail on WM8962 thermal shutdown status
- ASoC: Clear any outstanding WM8962 FLL lock completions before waiting
Ensure that we don't spuriously trigger early.
- ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962
This should never happen with level triggered IRQs.
- ASoC: Initial WM8962 DSP2 support
The WM8962 features a DSP providing a number of signal processing
features including HD Bass and Virtual Surround Sound (VSS). Enable
initial support for this, allowing users to enable and disable the
algorithms using the default coefficient sets. Further patches will
add support for runtime configuration of the DSP coefficients.
- ASoC: Add basic WM8962 capture low/high pass filter control
- ASoC: Move WM8962 CLKREG_OVD earlier
When the clocking registers are not overriden some of the registers are
not writable.
- ASoC: Acknowledge WM8962 interrupts before acting on them
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.
- ASoC: Defer all WM8962 clocking configuration until power up
Don't require an audio rate SYSCLK in hw_params() in order to better
support microphone detection use cases.
- ASoC: Implement base 5 band EQ control for WM8962
ReTune Mobile modes are not currently supported.
- ASoC: Report errors when we have a WM8962 IRQ and don't get FLL lock
We really should be getting the interrupt - if we don't get one it's very
likely that the configuration is incorrect and audio will fail. Also
increase the timeout substantially in this case for safety.
- ASoC: Factor out I2C usage in WM8962 driver
The chip can actually support SPI so we shouldn't assume we've got an I2C
device even though that's the most common configuration.
- ASoC: Fix WM8962 headphone volume update for use of advanced caches
- ASoC: Implement WM8962 ADC high pass filter configuration
- ASoC: Don't warn if the WM8962 SYSCLK FLL setting doesn't match reality
When bringing up audio low power modes boards may configure SYSCLK before
they actually start the FLL as we do much of the clocking setup prior to
the power up sequence.
- ASoC: Implement WM8962 DMIC support
DMIC support is automatically disabled when none of the GPIOs are set up
to bring out the DMICCLK and DMICDAT pins at startup.
Note that there's no support for controlling DMIC routing except the power
control so the board DAPM configuration will need to manage DMIC enable and
disable if analogue mics (eg, a headset) also exist.
- ASoC: Define constants for WM8962 GPIO functions
- ASoC: Move WM8962 FLL configuration to CODEC
There's only one DAI anyway.
- ASoC: Support FLL lock interrupt on WM8962
- ASoC: Accept any logical value for WM8962 GPIO set()

SoC Codec WM8971

- ASoC: Convert wm8971 MICBIAS to a supply widget
- ASoC: wm8971: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Avoid writing to WM8971_RESET in wm8971_resume
Writing to WM8971_RESET resets all registers to the default state.
Thus we should avoid writing to WM8971_RESET on resume.
- ASoC: wm8971: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write

SoC Codec WM8974

- ASoC: Convert wm8974 MICBIAS to a supply widget
- ASoC: wm8974: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: Remove references to linux@wolfsonmicro.com

SoC Codec WM8978

- ASoC: Remove unused "control_data" field of struct wm8978_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Add TLV information for WM8978 DAC limiter
- ASoC: Fix broken bitfield definitions in WM8978

SoC Codec WM8983

- ASoC: Convert wm8983 MICBIAS to a supply widget
- ASoC: Writing register default value for the reset register
The WM8983 can be reset by performing a write of any value to
the software reset register.
To avoid writing to the software reset register while resume,
we should write the same value in wm8983_reg_defs to software
reset register in wm8983_probe().
The write to the reset register is suppressed by the cache
restore code when it skips writes of default registers.
- ASoC: WM8983: Initial driver
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.

SoC Codec WM8985

- ASoC: Convert wm8985 MICBIAS to a supply widget

SoC Codec WM8988

- ASoC: Convert wm8988 MICBIAS to a supply widget
- ASoC: wm8988: Convert to snd_soc_cache_sync
Convert to snd_soc_cache_sync for sync reg_cache with the hardware.
- ASoC: wm8988: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: wm8988: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write

SoC Codec WM8990

- ASoC: Convert wm8990 MICBIAS to a supply widget
- ASoC: wm8990: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
This patch also includes a comment fix in wm8990_set_dai_pll(),
if freq_in and freq_out are 0, what we do is to clear WM8990_PLL_ENA bit.
Thus the comment should be "Turn off PLL".
- ASoC: wm8990: Fix wrong bit setting for WM8990_POWER_MANAGEMENT_2
If (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | (1 << WM8990_AINRMUX_PWR_BIT)))
is false, we should clear WM8990_AINR_ENA bits instead of WM8990_AINL_ENA.
- ASoC: wm8990: Remove incorrect comments
- ASoC: wm8990: Convert to snd_soc_cache_sync for sync reg_cache with the hardware

SoC Codec WM8991

- ASoC: Convert wm8991 MICBIAS to a supply widget
- ASoC: wm8991: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: wm8991: Fix wrong bit setting for WM8991_POWER_MANAGEMENT_2
If (fakepower & ((1 << WM8991_INMIXR_PWR_BIT)|(1 << WM8991_AINRMUX_PWR_BIT))))
is false, we should clear WM8991_AINR_ENA bits instead of WM8991_AINL_ENA.
- ASoC: Remove references to linux@wolfsonmicro.com
- ALSA: Remove unneeded version.h includes from sound/
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.

SoC Codec WM8993/4

- ASoC: Wait for WM8993 FLL to stabilise
Ensure the FLL is locked before we return from set_fll().
- ASoC: Fix partial cherry pick in wm8993
- ASoC: Use standard register cache sync in wm8993
- ASoC: Convert wm8993 to devm_kzalloc()
- ASoC: Convert WM8994 MICBIASes to supply widgets
There are some in tree systems using the driver but none use the MICBIAS
widgets.
- ASoC: wm_hubs: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the array.
- ASoC: wm8993: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the array.
- ASoC: Disable thermal shutdown when not using speakers in wm_hubs
The thermal shutdown support in wm_hubs devices is tied to the speaker
drivers (which are the only high power subsystems within the device).
Ensure minimal current usage when the thermal shutdown support is not
required by disabling the circuit when the speaker drivers are powered
down.
- ASoC: Replace remaining use of *_volsw_2r with *_volsw
The snd_soc_*_volsw_2r functionality has been merged to
*volsw callbacks.
Few places still used the get, or put variant of volsw_2r,
replace those with the corresponding *_volsw.
- ASoC: Add VMID widget for wm_hubs devices
Currently this does not actually do anything, it is being introduced in
order to facilitate additional power optimisations for current generation
devices.
- ASoC: Support separate left and right channel dcs_codes values
Some devices can have performance optimized by setting different offsets
for left and right channels.
- ASoC: Implement new DC servo readback mode for late WM8994 revisions
Later WM8994 devices implement a new DC servo readback mode with the
register used to access the offset moved to register 0x59. Implement
support for this and enable it on the appropriate devices.
- ASoC: Disable wm_hubs periodic DC servo update
This does not function correctly in all circumstances so disable the
periodic updates unconditionally for stable; a future patch will reenable
where appropriate.
- ASoC: Handle spurious wm_hubs DC servo done interrupts
Don't assume the first fire indicates that we're done.
- ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().
- ASoC: Use late enable handling for direct voice, speaker and headphone
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.
- ASoC: Correct left/right swap in wm_hubs DC offset correction
It was consistently wrong for everything except WM8993 so should be no
functional change.
- ASoC: Allow suppression of series updates on wm_hubs devices
Some devices do not support manual updates of the DC servo.
- ASoC: Trigger wm_hubs series update startup off a separate flag
Allowing the two to be used independently.
- ASoC: Fix wm_hubs input PGA ZC bits
- ASoC: Fix wrong data type access in a few codec drivers
Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol")
changed the control private data type that is passed to snd_soc_cnew when
creating dapm mixer and mux controls. Commit did not update a few codec
drivers that are using their own put callbacks and thus are accessing a
wrong data type.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Add some missing volume update bit sets for wm_hubs devices
- ASoC: Ensure output PGA is enabled for line outputs in wm_hubs
Also fix a left/right typo while we're at it.
- ASoC: Fix output PGA enabling in wm_hubs CODECs
The output PGA was not being powered up in headphone and speaker paths,
removing the ability to offer volume control and mute with the output
PGA.
- ASoC: Log wm_hubs DC servo operation code when reporting a timeout
- ASoC: Hook wm_hubs micbiases up to CLK_SYS
The microphone detection functionality requires a clock to work. In any
non-detection case where the MICBIAS is enabled CLK_SYS will be needed
anyway so there is no negative impact on power consumption.

SoC Codec WM8994

- ASoC: Enable ASoC register map dump for some regmap CODECs
It's still useful to be able to poke around in the register map at
runtime.
- ASoC: Remove unused label from wm8994 probe()
- ASoC: Remove WM8994 register cache
Now that the mfd is using the register map cache there's no need for the
CODEC driver to do any register cache management or any funny dances to
interact with the other drivers using the device so just remove the cache
initialisation and volatility information.
- ASoC: Remove ASoC-specific WM8994 I/O code
Just go directly to the regmap API, saving code and making integration
that bit more direct.
- sound: mfd: Define some additional wm8994 registers
Add a bunch of definitions for wm8994 registers that are not currently
used by software.
- ASoC: Rely on core enabling the wm8994 with runtime PM
No need to do this in the driver now.
- ASoC: Add missing err label
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: Convert WM8994 to devm_kzalloc()
Still have a manual free in there for some realloc()ed memory as there's
no devm version of that.
- ASoC: Add platform data for WM8958/WM1811 microphone detection rates
Allow systems to override the default microphone detection rates using
platform data in case the settings are not suitable (eg, due to an
unusually noisy jack).
- ASoC: Ensure we reconfigure WM8958 microphone detection on rate changes
We don't need to rerun DAPM if the clock source is the same but we do
need to adjust the microphone detection rate in case we are moving from
an audio to a non-audio rate.
- ASoC: Implement support for WM1811A jack detection
The WM1811A features an advanced low power accessory detection subsystem
which allows the device to be maintained in a very low power state while
the system is idle without sacrificing any accessory detection features.
Implement software support for this, automatically managing the power
configuration of the device depending on the detected accessory.
- ASoC: Rename WM8994 detecting flag to mic_detecting
More specific and avoids confusion with a following change.
- ASoC: Allow more WM8958/WM1811 button levels with default handler
The WM8958 and WM1811 support detecting a range of buttons. Allow the
user to provide platform data enabling more of these levels without
having to write a custom detection handler.
- ASoC: Tune down active mode detection rate for WM8958 mic detection
Saves a little power.
- ASoC: Don't use control_data to get struct wm8994
This will support refactoring to make use of the regmap API more directly
in the core.
- ASoC: Provide debug log of accessory status on WM8958
- ASoC: Enhance default WM8958 microphone detection
Actively manage the detection rate for microphones with WM8958, providing
improved power consumption and maximising the benefit from the hardware
debounce.
- ASoC: Put WM8958 and WM1811 MICBIAS into bypass mode when no audio
When we don't have any active audio we can put the microphone biases into
bypass mode to save power at the expense of performance.
- ASoC: Ensure SYSCLK is enabled for WM8958 accessory detection
Ensure SYSCLK is enabled while running accessory detection on WM8958.
It is always required so there is no sense in requiring machine drivers
to individually do this.
- ASoC: Mark WM8994 ADC muxes as virtual
Since they don't actually have power bits but do have events associated
with them it's important that we bootstrap their state properly which
making them virtual does.
- ASoC: Supply dcs_codes for newer WM1811 revisions
Based on initial data.
- ASoC: Error out if we can't generate a LRCLK at all for WM8994
- ASoC: Ensure we get an impedence reported for WM8958 jack detect
Occasionally we may see an accessory reported before we have a stable
impedance for the accessory. If this happens then reread the status in
order to ensure that the handler can take the appropriate action for the
status change.
- ASoC: Don't use wm8994->control_data when requesting IRQs
The field is no longer initialised so this will crash if running on
wm8958.
Reported-by: Thomas Abraham <thomas.abraham@linaro.org>
- ASoC: Don't use wm8994->control_data in wm8994_readable_register()
The field is no longer initialised so this will crash if running on
wm8958.
Reported-by: Thomas Abraham <thomas.abraham@linaro.org>
- ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
- ASoC: Update WM1811 DCS codes for latest evaluation results
Evaluation of larger quantities of material has provided new DCS codes
values to be applied for WM1811.
- ASoC: Remove impossible case from wm8994_hw_params
We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params.
Thus no need to check wm8994-aif3 in wm8994_hw_params.
- ASoC: wm8994: Fix setting rate_reg for wm8994-aif2
For wm8994-aif2, the rate_reg should be WM8994_AIF2_RATE.
- ASoC: wm8994: Slightly optimize configure_clock
snd_soc_update_bits() will only write new register value
if the old value is different from the new value.
In additional, snd_soc_update_bits() returns 0 for no change.
No need to read WM8994_CLOCKING_1 register before calling snd_soc_update_bits().
- ASoC: Add WM1811 support
The WM1811 is mostly register compatible with the WM8994 and WM8958,
providing a high performance audio hub CODEC in a small form factor
suitable for ultra compact system designs.
- ASoC: Fix backport of WM8994 thermal warning
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
- ASoC: Implement WM8994 thermal warning and shutdown interrupt support
ALSA doesn't really have good mechanisms for dealing with these so we just
log them - the hardware already has automatic shutdown support.
- ASoC: Add WM8958 noise gate support
- ASoC: Disable pulls on WM8994 AIF2 when starting it
Pull control is availalbe for WM8994 AIF2, generally disabled as part of
the GPIO configuration in order to save power after system startup. As on
newer devices in the series there is no GPIO functionality on these pins
this will happen less naturally so have the driver disable the pulls as the
AIF is probed.
- ASoC: Disable WM8994 VMID for digital only paths
On WM8994 class devices only the analogue portions of the CODEC require
VMID so when running digital only paths we can leave VMID disabled.
On some earlier devices the FLL uses VMID so we don't use DAPM reference
counting alone, we maintain an internal reference count which is also
enabled and disabled by the FLL startup.
- ASoC: Add VMID widget for wm_hubs devices
Currently this does not actually do anything, it is being introduced in
order to facilitate additional power optimisations for current generation
devices.
- ASoC: Specify register defaults for WM8958 MICBIAS1 and MICBIAS2
- ASoC: Support separate left and right channel dcs_codes values
Some devices can have performance optimized by setting different offsets
for left and right channels.
- ASoC: Implement new DC servo readback mode for late WM8994 revisions
Later WM8994 devices implement a new DC servo readback mode with the
register used to access the offset moved to register 0x59. Implement
support for this and enable it on the appropriate devices.
- ASoC: Add missing break in WM8994 probe
This error would have no effect on current silicon revisions, the fall
through case has the same behaviour.
- ASoC: Correct WM8994 MICBIAS supply widget hookup
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
- ASoC: Reduce power consumption for idle DAIs in WM8994
If DAIs are idle but their clocks are in use for some reason (eg, as
SYSCLK or for accessory detect) then set the clock dividers to the maximum
to reduce slightly the power consumption of the unclocked circuits.
- ASoC: Handle failed WM8994 FLL lock waits
Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.
- ASoC: Fix shift in WM8958 accessory detection default implementation
- ASoC: Log WM8994 FIFO errors from the interrupt
We should spot them anyway on state changes but logging them gives us
better time information about when the misconfiguration happened.
- ASoC: Don't warn on low WM8994/58 AIFnCLKs
We can have valid but very low clocks in accessory detection modes.
- ASoC: Use WM8994 FLL lock interrupt
If we have interrupts then wait for the FLL lock interrupt rather than
using dead reckoning when waiting for the FLL to start.
- ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958
- ASoC: Use late enable handling for direct voice, speaker and headphone
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.
- ASoC: Conditionalize the enable of WM8994 ADC TDM mode
Future devices will not benefit from this.
- ASoC: Trigger wm_hubs series update startup off a separate flag
Allowing the two to be used independently.
- ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
This delay is very conservative.
- ASoC: Disable WM8994/58 microphone detection over suspend
It will be non-functional with the basises and clocks off anyway, if the
system needs microphone detection enabled over suspend then it should be
causing the CODEC to ignore suspend using the APIs for that to prevent
the biases being disabled.
- ASoC: Set left channel volume update bits for WM8994
Ensures that we apply volume updates that don't affect the right channel.
- ASoC: Add WM8958 enhanced EQ support
DSP2 in the WM8958 can be used to support an upgraded EQ for use in
demanding applications.
- ASoC: Add WM8958 VSS support
With appropriate firmware the WM8958 can support Virtual Surround Sound or
VSS, widening the stereo audio image for improved user experience. Enable
support for this mode of operation when the appropriate firmware can be
loaded at runtime.
- ASoC: Refactor WM8958 DSP to support additional algorithms
In preparation for the addition of additional WM8958 algorithms
reorganise the code to make it easier to add such support later.
- ASoC: Support download of WM8958 MBC firmware
Allow userspace to supply an update to the ROM firmware. The firmware
request is non-blocking so userspace can load the firmware at its
leisure without delaying startup, the driver will begin using the
firmware the next time MBC is started after it has been supplied.
- ASoC: Mark WM8958 DSP2 registers readable
- ASoC: Handle startup sequencing of WM8958 DSP2 with deferred clocking
The DSP2 startup requires that the clock be enable so if we've deferred
clock startup we need to defer DSP2 startup
- ASoC: Factor WM8958 DSP2 handling into separate file
DSP2 on the WM8958 has a default ROM which provides a multi-band
compressor for enhanced performance on mobile devices but can also
support runtime download of alternative firmware. In preparation for
more exploiting this functionality refactor the code to split the
handling of DSP2 into a separate file.
- ASoC: Treat WM8958 revision A as WM8994 revision D
The first WM8958 revision requires similar treatment.
- ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly
Since we began using the late clock disable functionality, ensure that
we don't disable the clock if any of the ADC or DAC paths are still
enabled. This happens when we have simultaneous playback and recording.
- ASoC: Fix section mismatch warnings in WM8994
Annoying as the __devinitdata is actually correct.
- ASoC: WM8994: Ensure MICBIAS is provided with a clock
The patch 'ASoC: WM8994: Improve Playback Robustness' did not handle
this case properly.
- ASoC: Ensure WM8958 gets all WM8994 late revision widgets
Without this fix the driver won't instantiate properly on relevant
devices.
- ASoC: Fix typo in late revision WM8994 DAC2R name
Without this fix the driver won't instantiate properly on relevant
devices.
- ASoC: WM8994: Ensure late enable events are processed for the ADCs
Ensure that the ADCs are provided with a clock as the previous patch
"ASoC: WM8994: Improve playback robustness" did not handle this case
properly.
- ASoC: WM8994: Improve playback robustness
On WM8994 revision D and earlier ensure proper playback robustness
as some rare use cases can trigger issues.
- ASoC: WM8994: Improve robustness in some use cases
Ensure that on disabling certain registers such as AIF1DAC1L,
AIF1DAC1R etc. the AIF1CLK and AIF2CLK remain enabled. Similarly
when enabling those registers, AIF1CLK and AIF2CLK will remain
disabled.
- ASoC: Simplify default WM8958 jack detection code
The default WM8958 jack detection handler implements a full set of buttons
and also support for video detection. Support for multi-button jacks is
fairly system specific and will usually require some tuning for headsets
so simplify the implementation to only report a simple short to ground
button, leaving multi-button headsets to be handled by system specific
code.
- ASoC: Support configuration of WM8958 microphone bias analogue parameters
The WM8958 has a different microphone bias architecture to WM8994 so needs
different configuration to WM8994. Support this in platform data.
- ASoC: Support WM8958 direct microphone detection IRQ
Allow direct routing of the WM8958 microphone detection signal to a GPIO
to be used, saving the need to demux the interrupt.
- ASoC: Mark WM8958 microphone bias registers as readable
- ASoC: Mark WM8958 microphone detection registers readable
So they show up in codec_reg.
- ASoC: Fix missing space in WM8994
- ASoC: Fix WM8958 default microphone detection argument ordering
- ASoC: Improve WM8994 digital power sequencing
On WM8994 revision D and earlier ensure optimal sequencing with
simultaneous usage of AIF1 and AIF2 by tying the signals together
so if paths through both are connected the streams are started
simultaneously.
- ASoC: Create an AIF1ADCDAT signal widget to match AIF2
Due to the different routing for AIF1 and AIF2 we weren't using a
single widget to represent the ADCDAT signal. For consistency add
one.

SoC Codec WM8995

- ASoC: Fix wm8995 regmap usage
- ASoC: Convert WM8995 to direct regmap usage
Large code size increase due to the addition of readability information
and the reformatting of the defaults table.
- ASoC: Convert wm8995 MICBIASes to supply widgets
- ASoC: wm8995: Slightly optimize configure_clock
snd_soc_update_bits() will only write new register value
if the old value is different from the new value.
In additional, snd_soc_update_bits() returns 0 for no change.
No need to read WM8995_CLOCKING_1 register before calling snd_soc_update_bits().
- ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove
- ASoC: wm8995: Remove unused i2c variable in wm8995_remove()
- ASoC: wm8995: Return -EINVAL if device ID mismatch

SoC Codec WM8996

- ASoC: Enable ASoC register map dump for some regmap CODECs
It's still useful to be able to poke around in the register map at
runtime.
- ASoC: Tune the accessory detection rates for WM8996
Use longer intervals when the microphone is not inserted to increase
robustness against leisurely insertion.
- ASoC: Convert wm8996 to use devm_kzalloc()
- ASoC: Convert WM8996 gpiolib to regmap
Actually pretty straightforward.
- ASoC: Move most WM8996 resource acquisition to I2C probe
Now that the WM8996 driver is using the regmap API for register I/O we no
longer need the ASoC card to be active in order to interact with the chip.
In order to be more idiomatic for Linux move most of the existing probe()
function out into the I2C probe() function prior to registration with ASoC.
The IRQ and GPIO init will be moved separately as these are slightly more
involved.
- ASoC: Convert WM8996 to direct regmap API usage
- ASoC: Fix WM8996 24.576MHz clock operation
Record the clock after the divider as that is what all SYSCLK users see.
Without this the other clock configuration in the device comes out at
half rate.
- ASoC: wm8996: Avoid a redundant i2c_get_clientdata call in wm8996_i2c_remove
- ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
- ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
- ASoC: Remove needless codec->dapm.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: Add DRC control for WM8996
- ASoC: Refcount WM8996 bandgap from FLL too
For digital only paths we need to make sure the bandgap is enabled prior
to starting the FLL which isn't tied into DAPM.
- ASoC: Fix unused variable warning in WM8996
- ASoC: Initial WM8996 headphone impedance measurement support
The WM8996 can measure the impedance of accessories connected to the
headphone output. Implement initial support for this, measuring the
left channel impedance when an accessory is detected and using this
to distinguish between a line load and a headphone load.
- ASoC: WM8996 only needs bandgap for analogue functionality
Rather than managing the bandgap in the bias level control use a supply
widget as we only actually need to enable it for analogue paths, not
fully digital ones.
- ASoC: Fix WM8996 DC servo operation without IRQ
We need to count the timeout down.
Reported-by: Axel Lin <axel.lin@gmail.com>
- ASoC: Correct channel numbers for WM8996 AIF2
The AIF1 channels are numbered from zero than one; do the same thing for
AIF2 too.
- ASoC: Disable WM8996 CPVDD supply when not in use
The WM8996 only requires CPVDD when the charge pump is active so control
it separately to the other supplies, only enabling it when the charge pump
is active. This will result in a small power saving on systems which are
able to provide independent software control of the supply.
- ASoC: Check that WM8996 FLL started even if we don't have the IRQ
We can directly read the FLL lock status on WM8996 so even if we don't
have an interrupt wired up we can still verify that the FLL started
successfully.
- ASoC: Add 3D stereo support for wm8996
My first patch to ASoC ever! If I did something wrong, blame Ian.
- ASoC: Correct element count for WM8996 sidetone HPF
I can count. Honest.
- ASoC: Clear completions from late WM8996 FLL lock IRQs
In case we have a pending completion, for example due to a problem with
the input clock which got corrected after we timed out.
- ASoC: Optimise WM8996 no interrupt path
This occurs frequently if we are in edge triggered mode as we must poll the
interrupt status register until we get no more interrupts so it's worth
the effort - it means we skip writing null acknowledgements to the chip.
- ASoC: Automatically manage WM8996 MICBIAS regulating mode
For non-audio uses like accessory detection we can use a lower quality,
unregulated microphone bias, saving a little power. As the hardware can
manually enable and disable the biases we can select regulating mode
automatically with supply widgets connected to the biases.
- ASoC: Fix configuration of WM8996 input enables
There's no need for separate widgets for the enables (as the map already
shows).
- ASoC: WM8996 record paths need AIFCLK
Make AIFCLK supply the record paths otherwise record will not work unless
there is a simultaneous playback.
- ASoC: Acknowledge WM8996 interrupts before acting on them
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.
- ASoC: Rename WM8915 to WM8996
For marketing reasons the part will be called WM8996. In order to avoid
user confusion rename the driver to reflect this.

SoC Codec WM9081

- ASoC: Remove cache default for volatile wm9081 reset register
- ASoC: Convert wm9081 driver to use devm_kzalloc()
- ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
- ASoC: Convert WM9081 to direct regmap API usage
- ASoC: wm9081: Don't write WM9081_BIAS_ENA bit to WM9081_VMID_CONTROL register
WM9081_BIAS_ENA is the bit[1] of WM9081_BIAS_CONTROL_1 register (05h).
Current code incorrectly write it to WM9081_VMID_CONTROL(04h) register.
- ASoC: wm9081: Fix reading wrong register for setting VMID 2*240k
VMID Divider Enable and Select is controlled by BIT[2:1] of WM9081_VMID_CONTROL
register (04h).
Current code reads wrong register (WM9081_BIAS_CONTROL_1) for setting
VMID 2*240k.
- ASoC: Only enable thermal shutdown when required on WM9081
The WM9081 thermal shutdown is only effective when the speaker output is
enabled so disable it when that is not in use for a small current saving.
- ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k
According to the datasheet:
BIT 2:1
VMID_SEL[1:0] VMID Divider Enable and Select
00 = VMID disabled
01 = 2x40k Omh divider
10 = 2x240k Omh divider
11 = 2x5k Omh divider
To set VMID 2*240k, we should OR reg with 0x04 instead of 0x40.
- ASoC: WM9081 interrupt status register is volatile
Not that we have interrupt handling in the driver at the minute.
- ASoC: Convert WM9081 to table based control init
At least for the core controls, the optionally selected controls are still
added programatically.
- ASoC: Remove unused "control_data" field of struct wm9081_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Allow source specification for CODEC level sysclk
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
- ASoC: Change WM9081 speaker output enable to _OUT_DRV
More for neatness than any actual performance improvement.
- ASoC: Simplify WM9081 speaker startup by using widgets for spaker output
Now we have a register write minimisation code in DAPM we don't need to
worry about the ordering of the enable and disable of the PGA and the
output stage.
- ASoC: Convert WM9081 SYSCLK configuration to be device wide
Also respace the CODEC ops a bit for legibility.
- ASoC: Use data based init for WM9081 DAPM
- ASoC: Add platform data for WM9081 IRQ pin configuration
The WM9081 IRQ output can be either active high or active low and can
support either CMOS or open drain modes.
- ASoC: Fix WM9081 platform data initialisation
It went AWOL in the multi-component conversion.
- ASoC: Remove -codec suffix from WM9081 driver

SoC Codec WM9090

- ASoC: Use standard snd_soc_cache_sync() for WM9090
- ASoC: Convert WM9090 to devm_kzalloc()
- ASoC: wm9090: fix DB_RANGE size
Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent
reading more data than actually is in the arrays.
- ASoC: Remove unused variable 'wm9090' in wm9090_probe
Eliminate below build warning:
CC sound/soc/codecs/wm9090.o
sound/soc/codecs/wm9090.c: In function 'wm9090_probe':
sound/soc/codecs/wm9090.c:550: warning: unused variable 'wm9090'
- ASoC: Remove unused "control_data" field of struct wm9090_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: Add device ID for WM9093 to WM9090 driver
The WM9093 is an enhanced version of the WM9093. Add the device ID to
the driver, further patches will add support for the additional features
in the WM9093.
- ASoC: Remove unused mutex from WM9090 driver

SoC Codec WM9705

- ASoC: Use data based init for wm9705 DAPM

SoC Codec WM9712

- ASoC: Use data based init for wm9712 DAPM

SoC Codec WM9713

- ASoC: Use data based init for wm9713 DAPM

SoC Codec ads1174/8

- ASoC: Delete ads117x.h
This is not required after multi-component patch.

SoC Codec alc5621/2/3

- ASoC: Convert alc5623 to devm_kzalloc()
- ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls
The module desciption says this is ASoC alc5621/2/3 driver.
Make the naming consistent with the reset of the code.
- ASoC: alc5623: Convert codec->hw_read to snd_soc_read
codec->hw_read is broken now, let's covert to snd_soc_read.
- ASoC: Remove unused "control_data" field of struct alc5623_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
- ASoC: alc5623: Remove unused mutex

SoC Codec alc5632

- ASoC: Rename ALC5632 MICBIAS to common name convention.
- ASoC: alc5632: Remove volatile registers from regmap defaults
There is no need to provide defaults for the volatile
registers and doing so might cause confusion.
- ASoC: alc5632: Update of i2c_probe function to use regmap API only
- ASoC: alc5632: Added support of two undocumented registers
There are two undocumented registers in use in alc5632_i2c_probe
function. It must be added to support future rewrite of this
function to use regmap API completely.
- ASoC: alc5632: Remove unrelevant registers and name the relevant
- ASoC: alc5632: rename volume/switch contols for master and speaker volumes.
- ASoC: Convert ALC5632 codec to use regmap API
- ASoC: alc5632: Fix compile without CONFIG_PM
- ASoC: Remove unused control_data and mutex fields from struct alc5632_priv
Acked-off-by: Leon Romanovsky <leon@leon.nu>
- ASoC: Remove unnecessary backslash from alc5632 codec
- ASoC: Remove unused defines in alc5632 codec
- ASoC: Add new Realtek ALC5632 CODEC driver
This driver implements basic functionality, using I²C for the control
channel.

SoC Codec tlv320aic32x4

- ASoC: Convert tlv320aic32x4 to devm_kzalloc()
- ASoC: tlv320aic32x4: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
- ASoC: tlv320aic32x4 fix initialization of micpga routing
Checking the pdata-flags used 'or', so the check is always true. Use 'and' to
correctly mask the flags.
- ASoC: Fix outdated API usage in tlv320aic32x4
- ASoC: Add TI tlv320aic32x4 codec support.
This patch adds support for tlv320aic3205 and tlv320aic3254 codecs.
It doesn't include miniDSP support for aic3254.

SoC DaVinci

- ASoC: davinci-i2s.c: use devm_ functions
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
In this case, the original code did not contain a call to iounmap, nor does
one appear anywhere else in the file. I have assumed that it is safe to
use devm_ioremap for the allocation in any case.
- ASoC: davinci-mcasp.c: use devm_ functions
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
In this case, the original code did not contain a call to iounmap, nor does
one appear anywhere else in the file. I have assumed that it is safe to
use devm_ioremap for the allocation in any case.
- ASoC: davinci-vcif.c: use devm_ functions
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
- ASoC: davinci: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert davinci directory to module_platform_driver
Factor out some boilerplate code.
- Drop default from "DM365 codec select" choice
SND_DM365_EXTERNAL_CODEC does not exist, so it's a useless default.
- ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
OMAP drivers are omitted as I know Peter already has patches for them.
- ASoC: Davinci: Fix FS polarity for I2S format
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions
- ASoC: davinci-pcm: trivial: replace link with actual chan/link
The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.
Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.
- ASoC: davinci-mcasp: add support for unsigned PCM formats
Although the McASP supports sign-extending samples in RX or TX [1]; the
davinci-mcasp driver does not touch the {R,X}PBIT or {R,X}PAD field of the
{R,X}FMT registers meaning that the McASP will serialize the bytes it is given
regardless of their signedness. So supporting unsigned formats is as simple
as adding them to the metadata of the davinci-mcasp driver.
Update the FMTBITs reported in the snd_soc_dai_driver and also update the case
statements in davinci-mcasp's hw_params() function so that the McASP can be
connected to CODECs that use unsigned values.
[1] http://www.ti.com/lit/ug/sprufm1/sprufm1.pdf
- ASoC: davinci: add missing break statement
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
- ASoC: davinci: fix codec start and stop functions
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf
- ASoC: davinci-pcm: comments for the conversion to BATCH mode
In the previous commit 'ASoC: davinci-pcm: convert to BATCH mode', the phase
offset of 2 was mentioned in the commit message but not well commented in the
source.
Add descriptive comments of the phase offset with and without ping-pong
buffers enabled.
- ASoC: davinci-pcm: convert to BATCH mode
The davinci-pcm driver's snd_pcm_ops pointer function currently calls into
the edma controller driver to read the current positions of the edma channels
to determine pos to return to the ALSA framework. In particular,
davinci_pcm_pointer() calls edma_get_position() and the latter has a comment
indicating that "Its channel should not be active when this is called" whereas
the channel is surely active when snd_pcm_ops.pointer is called.
The operation of davinci-pcm in capture and playback appears to follow close
the other pcm drivers who export SNDRV_PCM_INFO_BATCH except that davinci-pcm
does not report it's positions from pointer() using the last transferred
chunk. Instead it peeks directly into the edma controller to determine the
current position as discussed above.
Convert the davinci-pcm driver to BATCH mode: count the periods elapsed in the
prtd->period member and use its value to report the 'pos' to the alsa
framework in the davinci_pcm_pointer function.
There is a phase offset of 2 periods between the position used by dma setup
and the position reported in the pointer function. Either +2 in the dma
setup or -2 in the pointer function (with wrapping, both) accounts for this
offset -- I opted for the latter since it makes the first-time setup clearer.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- ASoC: davinci-pcm: extract period elapsed functions
Extract functions that modify the prtd->period member in preparation for
conversion to BATCH mode playback.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- ASoC: davinci-pcm: fix audible glitch on 2nd ping-pong playback
The release of the dma channels was being performed in prepare and there was a
edma_resume call for the asp-channel only being executed on START, RESUME and
PAUSE_RELEASE.
The mcasp on da850evm with ping-pong buffers enabled was exhibiting an audible
glitch on every playback after the first. It was determined through trial and
error that the following two changes fix this problem:
1) Move the edma_start calls from prepare to trigger and 2) reverse the order
of starting the asp and ram channels.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- ASoC: davinci-pcm: increase the maximum channels
Based on the registration of davinci-mcasp.1 in the davinci-evm platform
setup for da830 and dm6467, davinci-pcm can handle more than the currently
reported maximum channels of 2.
Increase the maximum channels to 384 to match the maximum reported by
davinci-mcasp.1.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- ASoC: davinci-pcm: expand the .formats
Based on the data_type test in ping_pong_dma_setup, davinci-pcm is capable of
handling data of width up to and including 32bits.
"
if ((data_type == 0) || (data_type > 4)) {
printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
return -EINVAL;
}
"
Update the .format member of the snd_pcm_hardware instances it registers to
reflect this capability.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- ASoC: davinci-pcm: trivial: make ping-pong params setup symmetrical
The setup of the pong channel uses EDMA_CHAN_SLOT instead of & 0x3f as the
setup of the ping channel does.
Make the setup of ping and pong symmetric. There is no functional change
introduced by this patch.
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
- mfd: Use mfd cell platform_data for davinci cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
- ASoC: davinci-mcasp: enable ping-pong SRAM buffers
The davinci-i2s driver copies the platform data for playback and capture
sram sizes which is in turn used by davinci-pcm to allocate ping-pong
buffers.
Copy also the platform data in davinci-mcasp probe.
- davinci-mcasp: fix _CBM_CFS pin directions
The current davinci_mcasp_set_dai_fmt() sets bits ACLKX and ACLKR in the PDIR
register for the codec clock-master/frame-slave mode; however, this results in
the ACLKX and ACLKR pins being outputs according to SPRUFM1 [1] which
conflicts with "codec is clock master."
Similarly to the previous patch in this series, "fix _CBM_CFS hw_params" --
For codec clock-master/frame-slave mode (_CMB_CFS), clear bits ACLKX and ACLKR
in the PDIR register to set the pins as inputs and hence allow externally
sourced bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
- davinci-mcasp: fix _CBM_CFS hw_params
The current davinci_mcasp_set_dai_fmt() sets bits ACLKXE and ACLKRE (CLKXM
and CLKRM as they are reffered to in SPRUFM1 [1]) for codec clock-slave/
frame-slave mode (_CBS_CFS) which selects internally generated bit-clock and
frame-sync signals; however, it does the same thing again for codec
clock-master/frame-slave mode (_CBM_CFS) in the very next case statement which
is incorrectly selecting internally generated bit-clocks in this mode.
For codec clock-master/frame-slave mode (_CBM_CFS), clear bits ACLKXE and
ACLKRE to select externally-generated bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
- davinci-mcasp: use bitfield definitions for PDIR
The current driver creates value for set/clr of PDIR using (x<<26) instead
of the #defines that are convieniently made available.
Update the driver to use the bitfield definitions of PDIR. There is no
functional change introduced by this patch.
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
- ASoC: davinci-mcasp: correct tdm_slots limit
The current check for the number of tdm-slots specified by platform data is
always true (x >= 2 || x <= 32); therefore the else branch that warns of an
incorrect number of slots can never be taken.
Check that the number of tdm slots specified by platform data is between 2
and 32, inclusive.
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
- asoc: davinci_voicecodec: use mfd_data instead of driver_data
Use mfd_data for passing information from mfd drivers to soc
clients. The mfd_cell's driver_data field is being phased out.
Clients that were using driver_data now access .mfd_data
via mfd_get_data().
- ASoC: Davinci: Replace usage of IO_ADDRESS with ioremap()
This patch modifies the Davinci i2s and mcasp drivers to make use of
ioremap() instead of IO_ADDRESS()
- ASoC: Davinci: Call clk_disable() and clk_put() in case of error
In case of any error in probe() function, clk_disable() and clk_put()
should be called if clk_enable() and clk_get() went through.
- ASoC: Davinci: Use resource_size() helper function
This patch modifies the Davinci i2s and mcasp drivers
to make use of the resource_size() helper function for readability.
- asoc: davinci: da830/omap-l137: correct cpu_dai_name
McASP1 is used on the DA830/OMAP-L137 platform for the codec.
This is different from the DA850/OMAP-L138 platform which uses McASP0.
This is fixed by adding a new snd_soc_dai_link struct.

SoC Dynamic Audio Power Management

- ASoC: dapm - Fix check for codec context in dapm_power_widgets().
Fixes a NULL pointer dereference in dapm_power_widgets() if the dapm context
has no codec.
- ASoC: Dynamically allocate the rtd device for a non-empty release()
The device model needs a release() function so it can free devices when
they become dereferenced. Do that for rtds.
- ASoC: Take a pm_runtime reference on DAPM devices that are enabled
As for PCMs take a runtime power management reference to devices that are
in a non-off bias, avoiding the need to do this in individual drivers.
- ASoC: Add signal generator widget type
A signal generator behaves as an input would but is not considered for
any of the special behaviour associated with external input pins. This
is especially useful when automatically working out not connected widgets.
- ASoC: Log automatic pin disconnection per CODEC rather than per card
This makes the output a bit less confusing on multi-CODEC systems as the
same pin may appear in multiple CODECs.
- ASoC: Implement fully_routed card property
A card is fully routed if the DAPM route table describes all connections on
the board.
When a card is fully routed, some operations can be automated by the ASoC
core. The first, and currently only, such operation is described below, and
implemented by this patch.
Codecs often have a large number of external pins, and not all of these pins
will be connected on all board designs. Some machine drivers therefore call
snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core
never to activate them.
However, when a card is fully routed, the information needed to derive the
set of unused pins is present in card->dapm_routes. In this case, have
the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused
codec pin.
This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c.
- ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget
We really should be doing this in the core, not in a driver...
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
- ASoC: Cache connected input and output recursions
The number of connected input and output endpoints for a given widgets
can't change during a DAPM run so there is no need to redo the recursion
through branches of the tree we've already visited. Doing this on one of
my test systems gives an improvement of:
Power Path Neighbour
Before: 63 607 731
After: 63 141 181
which scales up well as more widgets are involved in paths.
- ASoC: Assign power_check when we allocate DAPM widgets
This ensures none of the rest of the code ever encounters a widget which
does not have a power check function.
- ASoC: Suppress early calls to snd_soc_dapm_sync()
Ensure we only have one sync during the initial startup of the card by
making snd_soc_dapm_sync() a noop on non-instantiated cards. This avoids
any bounces due to things like jacks reporting their initial state on
partially initialised cards. The callers that don't also get called at
runtime should just be removed.
- ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency
- ASoC: Stop checking for supplied widgets after we find the first
We don't really care how many widgets a supply is supplying, we just care
if the number is non-zero. This didn't actually produce any improvement
in the test cases I've been using but seems obviously sensible enough that
I'm pushing it out anyway.
We could do a similar thing for other widgets but this may be unhelpful
for further refactorings Liam was working on aiming to allow us to
identify connected audio paths.
- ASoC: Don't mark the outputs of supplies as dirty on state changes
The whole point of supply widgets is that they aren't inputs to their
sinks so a state change in a supply should never affect the state of the
widget being supplied and we don't need to mark them as dirty.
Power Path Neighbour
Before: 69 727 905
After: 63 607 731
This is particularly useful where supplies affect large portions of the
chip (eg, a bandgap supplying the analogue sections).
- ASoC: Only run power_check() on a widget once per run
Some widgets will get power_check() run on them more than once during a
DAPM run, most commonly due to supply widgets checking to see if their
consumers are powered up. It's wasteful to do this so cache the result
of power_check() during a run. For one system I tested this on I got an
improvement of:
Power Path Neighbour
Before: 106 970 1186
After: 69 727 905
from this.
- ASoC: Add verbose debugging showing why widgets get marked dirty
Help diagnose why we're checking widgets by providing some logging when
we first dirty them. This should possibly be a trace point if it's useful
but can be absurdly verbose if enabled, we can always change it later if
desired.
- ASoC: Reduce the number of neigbours we mark dirty when updating power
If two widgets are not currently connected then there is no need to
propagate a power state change between them as we mark the affected
widgets when we change a connection. Similarly if a neighbour widget is
already in the state being set for the current widget then there is no
need to recheck.
On one system I tested this gave:
Power Path Neighbour
Before: 114 1066 1327
After: 106 970 1186
which is an improvement, although relatively small.
- ASoC: Do DAPM power checks only for widgets changed since last run
In order to reduce the number of DAPM power checks we run keep a list of
widgets which have been changed since the last DAPM run and iterate over
that rather than the full widget list. Whenever we change the power state
for a widget we add all the source and sink widgets it has to the dirty
list, ensuring that all widgets in the path are checked.
This covers more widgets than we need to as some of the neighbour widgets
won't be connected but it's simpler as a first step. On one system I tried
this gave:
Power Path Neighbour
Before: 207 1939 2461
After: 114 1066 1327
which seems useful.
- ASoC: Mark headphone, mic, speaker and line widgets as always connected
We're not actually doing any dynamic power management based on connection
and output drivers (which are pretty much the same thing) are marked as
unconditionally connected already.
- ASoC: Factor out widget power check operation
We've got the same code in two different places, let's have it in a single
place instead.
- ASoC: Ensure all DAPM widgets have a power check callback
Makes the code simpler.
- ASoC: Move bias level decision into main dapm_power_widgets()
Future patches will try to reduce the number of widgets we check on each
DAPM run but we're still going to need to look and see if the devices is
on at all so we can manage the overall device bias. Move these checks out
into the main dapm_power_widgets() function so we don't have to think about
them for now.
Once we're doing more incremental updates it'll probably be worth using
refcounts for each bias level to avoid having to do the sweep over all
widgets but that's not going to be where the big performance wins are.
- ASoC: Factor write of widget power out into a separate function
Split the decision about what the new power should be out from the
implementation of that decision.
- ASoC: Also count neighbour checks for supplies
Missed when the stat was originally added.
- ASoC: Don't force bias on ground referenced devices
Currently we force all devices in the system to be at the same bias level.
This is due to concerns about power or pop/click impacts from either
ramping VMID or mismatching VMID on the analogue I/O lines between
connected devices but does mean we power devices up more often than we
really need to.
If a device flags idle_bias_off this will usually mean that it's either
all digital or ground referenced (in which case the idle and powered bias
levels are identical) so this concern does not apply and we can save some
power by leaving it off when not needed itself.
- ASoC: Add another DAPM stat for neighbour checks
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.
- ASoC: Factor out per-widget DAPM power checks
The indentation is getting a little deep. Should be straight code motion,
no functional changes.
- ASoC: Trace and collect statistics for DAPM graph walking
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
- ASoC: Display the error code when we fail to add a DAPM control
Useful for diagnostics.
- ASoC: soc-dapm: Fix parameter comment for snd_soc_dapm_free
We have dapm_context instead of codec parameter.
- ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch
Currently it is only possible to route one source per switch into a mixer.
This patch modifies the code, so that it is possible to route multiple sources
into a mixer via the same switch. One use-case for this is routing a stereo
channel pair into a mono-mixer via the same switch.
- ASoC: dapm - change stream event dbg to vdgb
Stream event debug can be noisy on larger audio devices so improve the
debug SNR by changing it to the verbose level.
- ASoC: dapm - Add DAPM stream completion event.
In preparation for Dynamic PCM (AKA DSP) support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs
after completion of stream events.
- ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Provide convenience methods to retrieve the soc_card or snd_card from a
DAPM context.
- ASoC: dapm - add DAPM macro for external enum widgets
Add a convenience macro for external enumerated widgets.
- ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
Makes life a little easier if you want to add subsequences to an existing
driver as you can use -1 to put things at the start of sequences.
- ASoC: core - Add platform widget IO
Allow platform driver widgets to perform any IO required for DAPM.
- ASoC: Fix DAPM sequence run for per-widget I/O methods
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.
Fix originally suggested by Liam on IM.
- ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
This time with soc_widget_update_bits reflecting recent soc_update_bits changes.
Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP
devices have mixer components that can alter power usage and hence require full
DAPM support.
This provides a generic widget IO operation wrapper in preparation for
future patches that implement platform driver DAPM.
- ASoC: Fix mismerge with release branch
- ASoC: Add weak routes for sidetone style paths
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.
Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.
- ASoC: Only provide a default bias level update for CODEC contexts
This allows the card driver to use the bias level variable more easily in
multi component systems.
- ASoC: Simplify logic in snd_soc_dapm_set_bias_level()
No functional changes but much less indentation.
- ASoC: Remove trace for DAPM bias level logging
It's redundant now thanks to the use of the generic trace infrastructure.
- ASoC: Indentation fix for null loop operation
More with the legibility.
- ASoC: Don't bring the CODEC up to full power for supplies and biases
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so
If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.
- ASoC: Specify target bias state directly as a bias state
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
- ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a
DAPM context not matching the widgets context. This can lead to a wrong
prefix_len calculation, which will result in undefined behaviour. To avoid
this always use the DAPM context from the widget itself.
- ASoC: simple style fix
replace the tab with spaces,
make it align with other paragraphs
- ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
Commit af46800 ("ASoC: Implement mux control sharing") introduced
function dapm_is_shared_kcontrol.
When this function returns true, the naming of DAPM controls is derived
from the kcontrol_new. Otherwise, the name comes from the widget (and
possibly a widget's naming prefix).
A bug in the implementation of dapm_is_shared_kcontrol made it return 1
in all cases. Hence, that commit caused a change in control naming for
all controls instead of just shared controls.
Specifically, a control is always considered shared because it is always
compared against itself. Solve this by never comparing against the widget
containing the control being created.
Equally, controls should never be shared between DAPM contexts; when the
same codec is instantiated multiple times, the same kcontrol_new will be
used. However, the control should no be shared between the multiple
instances.
I tested that with the Tegra WM8903 driver:
* Shared is now mostly 0 as expected, and sometimes 1.
* The expected controls are still generated after this change.
However, I don't have any systems that have a widget/control naming
prefix, so I can't test that aspect.
Thanks for Jarkko Nikula for pointing out how to fix this.
Reported-by: Liam Girdwood <lrg@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
- ASoC: Fix power down for widgetless per-card DAPM context case
Commit 52ba67b ("ASoC: Force all DAPM contexts into the same bias state")
powers up all the DAPM contexts in a card if any DAPM context becomes
active. Unfortunately power down newer happens if per-card DAPM context
doesn't have any widgets.
Reason for this is that power state of per-card DAPM context without
widgets is never cleared and thus all the DAPM contexts remain permanently
active. Test for widgetless calling DAPM context in dapm_power_widgets()
doesn't work for per-card DAPM context since power change is never
originating from widgetless per-card DAPM context.
Fix this by pre-clearing power state flag of non-codec DAPM context at the
beginning of power sequence.
- ASoC: Implement mux control sharing
Control sharing is enabled when two widgets include pointers to the
same kcontrol_new in their definition. Specifically:
static const struct snd_kcontrol_new adcinput_mux =
SOC_DAPM_ENUM("ADC Input", adcinput_enum);
static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
};
This is useful when a single register bit or field affects multiple
muxes at once. The common case is to have separate control bits or
fields for each mux (channel). An alternative way of looking at this
is that the mux is a stereo (or even n-channel) mux, rather than
independant mono muxes.
Without this change, a separate kcontrol will be created for each
DAPM_MUX. This has the following disadvantages:
* Confuses the user/programmer with redundant controls that don't
map to separate hardware.
* When one of the controls is changed, ASoC fails to update the DAPM
logic for paths solely affected by the other controls impacted by
the same register bits. This causes some paths not to be correctly
powered up or down. Prior to this change, to work around this, the
user or programmer had to manually toggle all duplicate controls away
from the intended setting, and then back to it.
Control sharing implies that the control is named based on the
kcontrol_new itself, not any of the widgets that are affected by it.
Control sharing is implemented by: When creating kcontrols, if a
kcontrol does not yet exist for a particular kcontrol_new, then a new
kcontrol is created with a list of widgets containing just a single
entry. This is the normal case. However, if a kcontrol does already
exists for the given kcontrol_new, the current widget is simply added
to that kcontrol's list of affected widgets.
- ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol
A future change will allow multiple widgets to be affected by the same
control. For example, a single register bit that controls separate muxes
in both the L and R audio paths.
This change updates the code that handles relevant controls to be able
to iterate over a list of affected widgets. Note that only the put
functions need significant modification to implement the iteration; the
get functions do not need to iterate, nor unify the results, since all
affected widgets reference the same kcontrol.
When creating the list of widgets, always create a 1-sized list, since
the control sharing is not implemented in this change.
- ASoC: Add w->kcontrols, and populate it
Future changes will need reference to the kcontrol created for a given
kcontrol_new. Store the created kcontrol values now.
- ASoC: s/w->kcontrols/w->kcontrol_news/g
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.
- ASoC: Remove DAPM debugfs entries before freeing widgets
Remove the DAPM debugfs entries before freeing the context's widgets, otherwise a
use after free situation might occur.
- ASoC: Move DAPM widget debugfs entry creation to snd_soc_dapm_new_widgets
Currently debugfs entries for a DAPM widgets are only added in
snd_soc_dapm_debugfs_init. If a widget is added later (for example in the
dai_link's probe callback) it will not show up in debugfs.
This patch moves the creation of the widget debugfs entry to
snd_soc_dapm_new_widgets where it will be added after the widget has been
properly instantiated.
As a side-effect this will also reduce the number of times the DAPM widget list
is iterated during a card's instantiation.
Since it is possible that snd_soc_dapm_new_widgets is invoked form the codecs or
cards probe callbacks, the creation of the debugfs dapm directory has to be
moved before these are called.
- ASoC: Move DAPM debugfs directory creation to snd_soc_dapm_debugfs_init
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.
- ASoC: Add dapm_find_widget helper
This patch adds a helper function for searching DAPM widgets by name.
This allows to streamline functions which operate on widgets by name.
It also allows to get rid of copy'n'pasted code which was added to fallback to
widgets from other contexts if the widget was not found in the current context.
- ASoC: fix a simple coding style issue
- ASoC: snd_soc_dapm_get_pin_status: Match other contexts too
Not all widgets on a card are within the codec's DAPM context. Fix
snd_soc_dapm_get_pin_status to search all contexts when looking for a
widget.
This change is required when modifying tegra_wm8903 to use
snd_soc_card.widgets rather than calling snd_soc_dapm_new_controls; the
former adds the widgets to the card's DAPM context, whereas tegra_wm8903
uses the codec's DAPM context when calling snd_soc_dapm_new_controls.
By code inspection, I suspect this also applies to Samsung Speyside.
- ASoC: Allow DAPM pin operations to match any context
The DAPM pin operations currently require that the specific DAPM context
that the pin being operated in is contained in be specified. With multi
component and especially with the addition of a per-card DAPM context
this isn't ideal as it means that things like disabling unused pins on
CODECs require looking up the CODEC DAPM context.
Fix this by falling back to matching a widget in any context if there isn't
a match in the current context. The code isn't ideal currently but will do
the job.
- ASoC: Force all DAPM contexts into the same bias state
Currently we allow all DAPM contexts to determine their own bias level.
While this should in general work in most situations and will deliver the
lowest possible power it causes problems for our integration with the
card bias level as we're calling the card bias level functions for each
DAPM context even though they're card wide but don't say which CODEC
we're calling them for. Mitigate against this by forcing everything to
be in the same state.
- ASoC: Remove special casing for registerless widgets
Since we recently explicitly set the register for registerless widgets
to no register there is no longer any need to special case power updates
for them, we can allow them to be handled with the register compression
code as other widgets are.
As this is the only remaining user of dapm_generic_apply_power() and
dapm_update_bits() also remove those functions.
Noticed-by: Lu Guanqun <guanqun.lu@intel.com>
- ASoC: Add bias level data to DAPM context debugfs
This is also in the old sysfs diagnostics but it's nice to have everything
in one place.
- ASoC: Explicitly say registerless widgets have no register
This stops code that handles widgets generically from attempting to access
registers for these widgets.
- ASoC: Use the correct DAPM context when cleaning up final widget set
Now we've got multi-component we need to make sure that the DAPM context
(and hence register I/O context) we use to apply the pending updates at
the end of a DAPM sequence is the one we were processing rather than the
one that was used to initate the state change.
- ASoC: Fix double addition of prefixes due to widget prefixing
We're not only prefixing all controls, we're also prefixing the widget
names in the runtime data. This causes us to add the prefix twice - once
when using the widget name to generate the control name and once when
adding the control.
Really we shouldn't be prefixing the widget names at all, the matching
code should be handing this as we always know which DAPM context a
widget came from and always display the widget name in terms of a DAPM
context. However, we're quite close to the merge window and that's
relatively invasive.
Reported-by: Jarkko Nikula <jhnikula@gmail.com>
- ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
- ASoC: Check for a CODEC before dereferencing in DAPM
A CODEC pointer is optional (and is checked for in most contexts within
DAPM) - add checks to the few places where it was missed.
- ASoC: Get the card directly from the DAPM context
Rather than indirecting through the CODEC we can look the card up directly
from the card pointer in the DAPM context.
- ASoC: soc-dapm: Include quotes around contents in debugfs entries
Sometimes the name of the control switch of a dapm route contains
spaces which makes it impossible to distinguish it from the source widget.
Add quotes around the names of the widgets to makes these parsable.
- ASoC: Run bias level changes for all DAPM contexts in parallel
As bias level changes can be quite time consuming and the bias changes
for multiple devices aren't strongly tied to each other (if anything it
can be advantageous to bring different devices up together) we can improve
the state transition time for multi-component systems by running the bias
level changes for all the devices in parallel. This is very simple to
achieve using the kernel async functionality so use that to schedule the
work.
This should have no practical effect for the overwhelming majority of
systems which have a single DAPM context - we'll bounce into another
thread to do the bias level change but otherwise everything will happen
in exactly the same order as it did before.
- ASoC: Remove card from snd_soc_dapm_set_bias_level()
We can get the card from the DAPM context so don't bother passing it as
an argument.
- ASoC: Remove export of snd_soc_dapm_stream_event()
The only thing that should ever be calling this is soc-core and that is
built as part of the same module so doesn't need the export.
- ASoC: Ensure supplies are maintained for force enabled widgets
If a widget has been force enabled then not only do we need to keep the
widget itself enabled, we also need to keep any supplies the widget
requires enabled. The user could force all the individual widgets on but
this requires too much knowledge of device internals.
Tested-by: Stephen Warren <swarren@nvidia.com>
- ASoC: Sync initial widget state with hardware
ASoC generally uses the register defaults for everything, but in some
cases the hardware will default to enabling some of the DAPM widgets
(clocks for example). Ensure that DAPM knows about the actual widget
state at initialisation by reading the enable bits after instantiating
the widgets so they don't get left enabled needlessly.
- ASoC: Add subsequence information to seq_notify callbacks
Allows drivers to distinguish which subsequence is being notified when
they get called back.
- ASoC: Remove controls from sequenced PGA arguments
We have zero users for PGA controls and the core support for them was
removed a while ago so no point in cut'n'pasting them into new macros,
even if it's too much hassle to update the existing ones.
- ASoC: Provide per widget type callback when executing DAPM sequences
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
- ASoC: Add support for sequencing within
With larger devices there may be many widgets of the same type in series
in an audio path. Allow drivers to specify an additional level of ordering
within each widget type by adding a subsequence number to widgets and then
splitting operations on widgets so that widgets of the same type but
different sequence numbers are processed separately. A typical example
would be a supply widget which requires that another widget be enabled
to provide power or clocking.
SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided
allowing this to be used with PGAs and supplies as these are the most
commonly affected widgets.
- ASoC: Explicitly say if we're powering up or down
Rather than passing the sequence to use for DAPM widgets around by reference
explicitly say if we're powering up or down until the point where we need
the sequence itself. This should make no practical difference in itself but
supports future refactoring.
- ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()
snd_soc_dapm_put_volsw() has variables for both the unshifted and
shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in
the middle of DAPM sequences) got confused between the two of these.
Since there's no need to keep a copy of the unshifted mask fix this and
simplify the code by using only one mask variable.
[Completely rewrote the changelog to describe the issue -- broonie.]

SoC EP93XX

- sound-soc: move to dma_transfer_direction
fixup usage of dma direction by introducing dma_transfer_direction,
this patch moves asoc drivers to use new enum
- ASoC: ep93xx: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Use dai_fmt in snappercl15 machine driver
Reviewed-by: Mika Westerberg <mika.westerberg@iki.fi>
- ASoC: Use dai_fmt in edb93xx machine driver
- ASoC: Convert ep93xx directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Staticise ep93xx_ac97_dai
- ASoC: edb93xx: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
- ASoC: simone: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to
use snd_soc_register_card() instead of creating a "soc-audio" platform device.
In addition we use platform_device_register_simple() to create a platform
device for the codec. This function will handle putting and deleting the
device automatically which simplifies the error handling in the machine
driver.
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
- ASoC: ep93xx-pcm: add MODULE_ALIAS
To get the PCM module loaded automatically by udev et al. we need to add a
proper MODULE_ALIAS.
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
- ASoC: snappercl15: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
- sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree
Introduce a new label that includes kfree and jump to that one.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
- Change Ryan Mallon's email address across the kernel
I no longer work at Bluewater Systems. Update my email address accordingly. I
have deleted my email address from C files rather than change it. This
was suggested by several people, since the commit from my new email
address will cause scripts/get_maintainer.pl to function properly. I
have not added the .mailmap entry as suggested by Joe because I think
it is no longer necessary if I touch all the files which had my name
in them.
Reviewed-by: Jesper Juhl <jj@chaosbits.net>
- ASoC: ep93xx: convert to use the DMA engine API
Now that we have the EP93xx DMA engine driver in place, we convert the ASoC
drivers (I2S, AC97 and PCM) to take advantage of this new API. There are no
functional changes.
- ASoC: Enable 192kHz sample rate for EP93xx.
Enable 192kHz sample rate for EP93xx.
- ASoC: Improve EP93xx I2S clocks management.
Improve EP93xx I2S clocks management.
Some freqs values are set not exact as they requested for MCLK and
original code was not able to find divisors for SCLK and LRCLK.
This code just picks up nearest value from 3 possible variants.
This patch makes 44100 and 192000 rates working and fixes
capture function (by selecting SCLK/LRCLK=64 where possible).
All other rates should work as before.
- ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx
machine driver.
- ASoC: Remove warnings in ep93xx-i2s.c
Remove warnings in ep93xx-i2s.c
- ASoC: ep93xx-ac97: remove extra empty line
- ASoC: EDB93xx machine sound driver with CS4271
Added support for EDB93xx sound with CS4271 CODEC.
Features:
- Playback, Capture
- Sample rates from 8kHz to 96kHz tested

SoC FSI SH7724

- ASoC: fsi-ak4642: modify specification method of FSI / ak464x
Current fsi-ak4642 was using id_entry name in order to specify
FSI port and ak464x codec.
But it was no sense, no flexibility.
Platform can specify FSI/ak464x pair by this patch.
- ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format
Current format selection of FSI-codecs depended on platform information for FSI,
and chip default settings for codecs. It is not understandable/formal method.
This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt.
But FSI can use I2S/PCM and SPDIF format today.
It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF.
So, this patch change FSI platform information to have DAI/SPDIF mode.
If platform selects DAI mode (default),
FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt,
and if it is SPDIF mode, FSI become SPDIF format.
- ASoC: sh: fsi: Add snd_soc_dai_set_fmt support
This patch add snd_soc_dai_ops :: set_fmt to FSI driver and
select master/slave clock mode by snd_soc_dai_set_fmt on
fsi-xxx.c instead of platform infomation code.
This patch remove fsi_is_master function which is no longer needed.

SoC Freescale

- ASoC: fsl: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: fsl/powerpc: don't rely on the cell-index property
Instead of using the 'cell-index' property in the I2C adapter node to
determine the adapter number, just query the i2c_adapter object directly.
Previously, the I2C nodes always appeared in cell-index order, so the
dynamic numbering coincided with the cell-index property. With commit
ab827d97 ("powerpc/85xx: Rework P1022DS device tree"), the I2C nodes are
unintentionally reversed in the device tree, and so the machine driver
guesses the wrong I2C adapter number.
- ASoC: p1022ds: add support for fsl,P1022 and fsl,P1022DS model names
Commit ab827d97 ("powerpc/85xx: Rework P1022DS device tree") renamed the
the /model property of the P1022DS device tree from "fsl,P1022" to
"fsl,P1022DS". To support both old and new device trees, the ASoC
machine driver for the P1022DS needs to query the /model property and
update the platform driver object dynamically.
- ASoC: Convert fsl directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: mpc8610: tell the CS4270 codec that it's the master
Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is
detected") requires the machine driver to tell the CS4270 codec driver
whether the CS4270 should be configured for master or slave operation.
- ASoC: fsl_ssi: properly initialize the sysfs attribute object
Commit 6992f533 ("sysfs: Use one lockdep class per sysfs attribute")
requires 'struct attribute' objects to be initialized with sysfs_attr_init().
- ASoC: fsl: Fix error handling if platform_device_add fails
Call platform_device_put() instead of platform_device_unregister() if
platform_device_add() fails.
- ASoC: improve asynchronous mode support in the fsl_ssi driver
The Freescale SSI audio controller supports "synchronous" and "asynchronous"
modes. In synchronous mode, playback and capture use the same input clock,
so sample rates must be the same during simultaneous playback and capture.
Unfortunately, the code which supports asynchronous mode is just broken in
various ways. In particular, it was constraining sample sizes as well as
the sample rate.
The fix also allows us to simplify the code by eliminating the 'asynchronous',
'playback', and 'capture' variables that were used to keep track of playback
and capture streams.
Unfortunately, it turns out that simulataneous playback and record does not
actually work on the only platform that supports asynchronous mode: the
Freescale P1022DS reference board. If a second stream is started, the SSI
grinds to halt for both streams. This is true even if the P1022 is configured
for synchronous mode, so it's likely a hardware problem that needs to be
worked around.
- ASoC: Remove redundant -codec from WM8776 driver name
- ASoC: MPC5200: replace of_device with platform_device
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma)
accordingly. This fixes a build break.
- sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put
The first change is to add an of_node_put, since codec_np has previously
been allocated. The rest of the patch reorganizes the error handling code
so the only code executed is that which is needed.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
- sound/soc/fsl/p1022_ds.c: add missing of_node_put
dma_channel_np has been accessed at this point, so decrease its reference
count before leaving the function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
- sound/soc/fsl/fsl_dma.c: add missing of_node_put
of_parse_phandle increments the reference count of np, so this should be
decremented before trying the next possibility.
Since we don't actually use np, we can decrement the reference count
immediately.
Reported-by: Julia Lawall <julia@diku.dk>
- ASoC: fsl: fix build warning in fsl_dma
The previous patch to fsl_dma.c ("fix initialization of DMA buffers")
left behind an unused local variable that causes a build warning.
- ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when
the device is opened, which means that the /proc/interrupts entry for
the SSI exists only during playback or capture. This also meant that
the user won't know that the IRQ number is wrong until he tries to use
the device. Instead, we should claim the IRQ when the device is probed.
- ASoC: p1022ds: fix incorrect referencing of device tree properties
Device tree integer properties are encoded in big-endian format, but some of
the Freescale ASoC drivers were assuming that the host is in big-endian format
as well. Although this is true, it's better to use endian-safe accessors.
Also add a check for a failed ioremap() call in the SSI driver.
- ASoC: fsl: fix initialization of DMA buffers
The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs
for playback and capture, so DMA buffers should be allocated only for the
initialized streams. Instead of checking for the number of active channels,
which apparently is not reliable, check to see if the actual stream object
exists.
Also provide a better name for the DMA interrupt.
- dt/sound: Eliminate users of of_platform_{,un}register_driver
Get rid of users of of_platform_driver in drivers/sound. The
of_platform_{,un}register_driver functions are going away, so the
users need to be converted to using the platform_bus_type directly.
- ASoC: Replace pdev with card in machine driver probe and remove
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.

SoC Ingenic JZ4740

- ASoC: jz4740: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert jz4740 directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Staticise jz4740_pcm_new()
It is not used outside this driver so no need to make the symbol global.
- ASoC: JZ4740: Fix i2s shutdown
The i2s shutdown callback has the check whether it should be disabled reversed.
Currently it is disabled if another stream is still active, but kept enabled if
the last stream is closed. This patch fixes it.
- ASoC: JZ4740: qi_lb60: Use the SND_SOC_DAPM_EVENT_OFF for the speakers status
Use SND_SOC_DAPM_EVENT_OFF for determining whether the speaker should be turned
on or off instead of open coding it.
- ASoC: JZ4740: qi_lb60: Use gpio_request_array to request and setup gpios
This patch changes the qi_lb60 setup code to use gpio_request_array instead of
manually calling gpio_request and gpio_direction_output for each gpio.
Doing so makes the code a bit more compact.
- ASoC: JZ4740: Convert qi_lb60 codec to table based DAPM setup
Use the newly introduced dapm_widgets, dpam_routes and fields of the
snd_soc_card struct to setup DAPM.

SoC Intel Medfield MID platform

- ASoC: mid-x86: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Fix compile error in sound/soc/mid-x86/sst_platform.c
The variable ret_val is used but not declared. This causes
the following compile error:
sound/soc/mid-x86/sst_platform.c: In function ‘sst_platform_open’:
sound/soc/mid-x86/sst_platform.c:274:2: error: ‘ret_val’ undeclared (first use in this function)
sound/soc/mid-x86/sst_platform.c:274:2: note: each undeclared identifier is reported only once for each function it appears in
make[1]: *** [sound/soc/mid-x86/sst_platform.o] Error 1
Fix this.
- ASoC: Staticise mfld_msic_dailink
Acked by: Vinod Koul <vinod.koul@linux.intel.com>
- ASoC: Staticise sst_pcm_new and sst_soc_platform_drv
Acked-by Vinod Koul <vinod.koul@linux.intel.com>
- ASoC: sst_platform: fix the dsp driver interface
lower level drivers typically register with upper layers.
So fix by exporting symbols from sst_platform driver for dsp driver to
register to sst platform driver
Now this driver doesnt depend on sst driver, so remove the dependency
and the header files
- ASoC: Convert mid-x86 directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Staticise sst_platform_dai
It is not used outside this driver so no need to make the symbol global.
- ASoC: Remove unused fields in struct mfld_mc_private
Both *socdev and *codec of struct mfld_mc_private are not being used
in this driver, remove it.
- ASoC: sst_platform: fix memory leak
snd_pcm_hw_constraint_integer() could return -1, in this case, sst platform is
not opened successfully. However the corresponding close callback isn't able
to be called later on to release these two allocated memories, thus resulting
in memory leak.
This patch moves the check for hardware contraints earlier, thus resolving this
issue.
- ASoC: sst_platform: using builtin function
Use the builtin snd_soc_set_runtime_hwparams() instead of assigning it by
myself.
- ASoC: sst_platform: trivial coding style fix
- ASoC: sst_platform: add hw_free callback to fix resource leak
- ASoC: sst_platform: Fix lock acquring
Fix the possible dead lock shown below:
spin_lock
sst_get_stream_status
sst_period_elapsed
intel_sst_interrupt
handle_IRQ_event
handle_fasteoi_irq
do_IRQ
common_interrupt
spin_lock
sst_set_stream_status
sst_platform_pcm_trigger
- ASoC: sst_platform: unregister sst card when being closed
- ASoC: sst_platform: free the resources on fail path
- ASoC: sst_platform: initialize module_name properly
module_name will be checked in register_sst_card.
It will fail to register sst card if it's not initialized.
- ASoC: sst_platform: fix the pulseaudio error
Pulseaudio doesnt work with current driver and it was root caused to absense of
hw_params function and malloc_pages in it.
This patch adds this and allows pa to work fine with these drivers
- ASoC: mfld_machine: make use of soc_register_card API
This patch removes the old method of soc-audio device creation in mfld machine
and makes use of new soc_register_card API to register the card
- ASoC: mid-x86: Use the soc-jack apis for jack type detection
This patch modifies the mfld_machine to use the new jack apis for adding the
voltage zones for jack type detection. It also modifed TI sn95031 codec driver
to use these new apis
- ASoC: mfld_machine: Add support for jack detection
This patch adds support for registering jack interupt
and registering jack with core
- ASoC: mid-x86: Fix dependency on intel_sst driver
Enabling medfield asoc driver causes compliation error when intel_sst
is not selected
ERROR: "register_sst_card" [sound/soc/mid-x86/snd-soc-sst-platform.ko]
undefined!
This patch puts proper dependency to elimate build error
Reported-by: Andrew Morton <akpm@linux-foundation.org>
- ASoC: mid-x86: Add support for capture in machine driver
This configures the capture unused pins
- ASoC: sst_platform: add support for capture stream on headset dai

SoC Layer

- Rediff the usbusx2y.c and soc-core.c patches to match recent kernel
- Don't include obsolete trace/events/asoc.h hack
- Fix a typo in soc-core.patch in the prvioust commit
- Fix build-errors of soc-core due to new trace points
Ugly hacks again.
- Add build stub for soc/soc-io.c
- Fix a build of soc-pcm.c with older kernels without mutex_lock_nested()
- Fix wrongly refreshed soc-core.patch
- Add missing soc/soc-pcm.c build stub
- Regenerated soc-core.patch to fix fuzz
- Fix a compile error of soc/soc-jack.c with older kernels
Just define a dummy irq_set_irq_wake().
- Regenerated soc/soc-core.patch
- Regenerate ac97_codec.patch and soc-core.patch to resolve fuzz
- Fixed typos in soc-core.patch
to fix build errors.
- Fix yet another build error in ASoC for older kernels
- Fix build with the recent ASoC updates
- Regenerated soc-core.patch
- Always use the dummy trace/events/asoc.h
- ASoC: core - Free platform DAPM context at platform removal.
Fix platform removal by freeing the platform DAPM resources and remove
it from the DAPM list.
- ASoC: Dynamically allocate the rtd device for a non-empty release()
The device model needs a release() function so it can free devices when
they become dereferenced. Do that for rtds.
- ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
The original code does not cover the case that two DAIs(CPU) have different
ASoC core PCM operations(like mmap, pointer...). Currently we have only one
global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different
pointer functions, second DAI's pointer function is set for both first DAI
and second DAI in case of original code.
This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So
each DAIs can have different ASoC core PCM operations. This is needed to
support multiple DAIs.
- ASoC: Declare soc_new_pcm() properly
Ensure that everything is seeing the same declaration by moving it to
a header file rather than putting the declaration in soc-core.c
- ASoC: Allow DAI links to be specified using device tree nodes
DAI link endpoints and platform (DMA) devices are currently specified
by name. When instantiating sound cards from device tree, it may be more
convenient to refer to these devices by phandle in the device tree, and
for code to describe DAI links using the "struct device_node *"
("of_node") those phandles map to.
This change adds new fields to snd_soc_dai_link which can "name" devices
using of_node, enhances soc_bind_dai_link() to allow binding based on
of_node, and enhances snd_soc_register_card() to ensure that illegal
combinations of name and of_node are not used.
- ASoC: Remove ifdefs for GPIO_SYSFS
It is part of the GPIO API so should be stubbed appropriately.
- ASoC: Add utility to parse DAPM routes from device tree
Implement snd_soc_of_parse_audio_routing(), a utility function that can
parses a simple DAPM route table from device tree.The machine driver
specifies the DT property to use, since this is binding-specific.
- ASoC: Add utility to set a card's name from device tree
Implement snd_soc_of_parse_card_name(), a utility function that sets a
card's name from device tree. The machine driver specifies the DT
property to use, since this is binding-specific.
- ASoC: Remove rbtree register cache
All users now use regmap directly so delete the ASoC version of the code.
- ASoC: Fix an obvious copy paste error in an error message
The message was obviously copied from soc_init_codec_debugfs()
- ASoC: Refactor some conditions and loop in soc_bind_dai_link()
Transform some loops from:
for_each(x) {
if (f(x)) {
work_on(x);
}
}
to new structure:
for_each(x) {
if (!f(x))
continue;
work_on(x);
}
This will allow future modification of f(x) with less impact to the code.
- ASoC: Hold runtime PM references to components of active DAIs
Every device that implements runtime power management for DAIs is doing
it in pretty much the same way: in the startup callback they take a
runtime PM reference and then in the shutdown callback they release that
reference, keeping the device active while the DAI is active. Given the
frequency with which this is done and the obviousness of the need to keep
the device active in this period factor the code out into the core, taking
references on the device for each CPU DAI, CODEC DAI and DMA device in the
core.
As runtime PM is reference counted this shouldn't interfere with any
other reference holding by the drivers, and since (in common with the
existing implementations) we don't check for errors on enabling it
shouldn't matter if the device actually has runtime PM enabled or not.
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
- ASoC: Sort WM9090 in with the CODEC drivers
The driver itself has been a regular CODEC driver for a while now.
- ASoC: Convert WM2000 into a standard CODEC driver
We've been able to handle external amps for a while now.
- ASoC: Drop unused state parameter from CODEC suspend callback
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
- ASoC: Move SigmaDSP firmware loader to ASoC
It has been pointed out previously, that the firmware subsystem is not the right
place for the SigmaDSP firmware loader. Furthermore the SigmaDSP is currently
only used in audio products and we are aiming for better integration into the
ASoC framework in the future, with support for ALSA controls for firmware
parameters and support dynamic power management as well. So the natural choice
for the SigmaDSP firmware loader is the ASoC subsystem.
- ASoC: Fix CODEC enumeration for auto_nc_codec_pins
We need to enumerate all the CODECs that are part of the card we're
instantiating, not all the CODECs that are in the system as the system
may have multiple cards.
- ASoC: Implement fully_routed card property
A card is fully routed if the DAPM route table describes all connections on
the board.
When a card is fully routed, some operations can be automated by the ASoC
core. The first, and currently only, such operation is described below, and
implemented by this patch.
Codecs often have a large number of external pins, and not all of these pins
will be connected on all board designs. Some machine drivers therefore call
snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core
never to activate them.
However, when a card is fully routed, the information needed to derive the
set of unused pins is present in card->dapm_routes. In this case, have
the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused
codec pin.
This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c.
- ASoC: Add support for CS42L73 codec
This patch adds support for the Cirrus Logic CS42L73 low power stereo
codec.
- ASoC: Remove LZO cache type
There are no current users and new drivers ought to be using the regmap
API and its cache implementation directly so just delete the ASoC copy.
- ASoC: Add new Realtek ALC5632 CODEC driver
This driver implements basic functionality, using I²C for the control
channel.
- ASoC: Remove extra space in runtime struct definition
My usual technique for finding definitions is to search for "name {"
which breaks with the extra space.
- ASoC: Fix build dependency for SND_SOC_JZ4740_CODEC
Currently SND_SOC_JZ4740_CODEC depends on SOC_JZ4740 but SOC_JZ4740 is not
defined in any Kconfig. Thus the codec driver will not be built when select
"Build all ASoC CODEC drivers".
(Unless it is selected by SND_JZ4740_SOC_QI_LB60).
Remove the dependency with SOC_JZ4740, then this code driver can be built when
select "Build all ASoC CODEC drivers".
- ASoC: Provide a more complete DMA driver stub
Allow userspace applications to do more parameter setting by providing a
more complete stub DMA driver specifying a wildcard set of formats and
channels and essentially random values for the DMA parameters. This is
required for useful runtime operation of the dummy DMA driver until we
are able to figure out how to power up links and do hw_params() from DAPM.
Sending to stable as without this the dummy driver is not terribly
useful.
Reported-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Tested-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
- ASoC: skip resume of soc-audio devices without codecs
There are cases where there is no working codec on the soc-audio devices,
and snd_soc_suspend() will skip such device when suspending. Yet its
counterpart snd_soc_resume() does not check this, causing complaints
about spinlock lockup:
[ 176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8
[ 176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24)
[ 176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158)
[ 176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68)
[ 176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c)
[ 176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0)
[ 176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c)
[ 176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0)
[ 176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0)
[ 176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8)
- ASoC: Remove needless unlikely()
There's no point in adding unlikely() annotations outside of hot paths
and on systems using these features the annotation will always be wrong
(as opposed to being something that only comes up once in a while) so
the annotation may even be harmful.
- ASoC: Allow machines to ignore pmdown_time per-link
With this flag, each dai_link in machine driver can choose
to ignore pmdown_time during DAPM shut down sequence.
If the ignore_pmdown_time is set, the DAPM for corresponding DAI
will be executed immediately.
- ASoC: Fix prefixing of DAPM controls
We don't want to clear the prefix while we're creating the DAPM controls
for the device as the prefix is applied during control creation.
- ASoC: core: Add flag to ignore pmdown_time at pcm_close
With this flag codec drivers can indicate that it is desired
to ignore the pmdown_time for DAPM shutdown sequence when
playback stream is stopped.
The DAPM sequence will be executed without delay in this case.
- ASoC: Instantiate card widgets immediately
This ensures they are available prior to the card late_probe().
- ASoC: Fix typo in Kconfig symbol for tlv320aic32x4
It is currently named "TVL" instead of "TLV".
- ASoC: Squash error codes from regmap down to -1 on read
The ASoC code always uses -1 as the error code due to reporting errors in
band with the value. Ensure we don't confuse anything by making sure we
don't pass actual error codes back into the rest of the code on read.
- ASoC: Ensure all DAPM widgets are instantiated with the card
Specifically for the widgets added by machine driver late probe functions.
- ASoC: Suppress early calls to snd_soc_dapm_sync()
Ensure we only have one sync during the initial startup of the card by
making snd_soc_dapm_sync() a noop on non-instantiated cards. This avoids
any bounces due to things like jacks reporting their initial state on
partially initialised cards. The callers that don't also get called at
runtime should just be removed.
- ASoC: fix codec breakage caused by the volsw/volsw_2r merger
By accident few places still uses the _2r calls from
the core.
This is a quick fix, the drivers using the old callbacks
going to be changed.
- ASoC: Ensure DAPM widgets are set up before we sync jacks
We synchronise jack state on startup - when we do that make sure that we
have set up all the DAPM widgets first in case we end up touching any of
the partially set up widgets when syncing the jack pins.
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
- ASoC: core: Remove snd_soc_put_volsw_2r definition
We do not have users for snd_soc_put_volsw_2r anymore.
It can be removed.
- ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions
Handle the put_volsw/put_volsw_2r in one function.
To avoid build breakage in twl6040 keep the
snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw.
- ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions
Handle the get_volsw/get_volsw_2r in one function.
- ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions
Handle the info_volsw/info_volsw_2r in one function.
- ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation
SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the
bits are within one register.
Assign .rreg to be the same as .reg for these types.
With this change we can tell if the mixer in question:
is mono:
mc->reg == mc->rreg && mc->shift == mc->rshift
is stereo, within single register:
mc->reg == mc->rreg && mc->shift != mc->rshift
is stereo, in two registers:
mc->reg != mc->rreg
The patch provide a small inline function to query, if the mixer
is stereo, or mono.
- ASoC: Do DAPM power checks only for widgets changed since last run
In order to reduce the number of DAPM power checks we run keep a list of
widgets which have been changed since the last DAPM run and iterate over
that rather than the full widget list. Whenever we change the power state
for a widget we add all the source and sink widgets it has to the dirty
list, ensuring that all widgets in the path are checked.
This covers more widgets than we need to as some of the neighbour widgets
won't be connected but it's simpler as a first step. On one system I tried
this gave:
Power Path Neighbour
Before: 207 1939 2461
After: 114 1066 1327
which seems useful.
- ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro
With the new macro we can remove duplicated code
for the SOC_DOUBLE_R type of controls.
- ASoC: core: Introduce SOC_DOUBLE_VALUE macro
With the new macro we can remove duplicated code
for the SOC_DOUBLE type of controls.
We can also remap the SOC_SINGLE_VALUE macro to
SOC_DOUBLE_VALUE
- ASoC: Instantiate DAPM widgets before we do the DAI link init
The DAI init function may want to do something that needs the widgets to
be instantiated.
- ASoC: Allow DAI formats to be specified in the dai_link
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.
Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.
- ASoC: Add Kconfig and Makefile entries for rt5631 codec
- ASoC: soc-core: symmetry checking for each DAIs separately
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.
We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: Ensure we generate a driver name
Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver
field) broke generation of a driver name for all ASoC cards relying on the
automatic generation of one. Fix this by using the old default with spaces
replaced by underscores.
- ASoC: Trace and collect statistics for DAPM graph walking
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
- ASoC: Add WM5100 driver
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.
- ASoC: Remove unused step size from debugfs CODEC write function
We don't use the step size so there's no need to work it out.
- ASoC: Fix reporting of partial jack updates
We need to report the entire jack state to the core jack code, not just
the bits that were being updated by the caller, otherwise the status
reported by other detection methods will be omitted from the state seen
by userspace.
- ASoC: Allow source specification for CODEC level sysclk
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
- ASoC: Allow register defaults to be larger than unsigned short
Devices that need this exist; obviously the newer regmap defaults
mechanism will deal with this more happily.
- ASoC: soc_codec_reg_show use snd_soc_codec_readable_register
Use snd_soc_codec_readable_register instead of open-coding it.
- ASoC: Fix register cache sync register_writable WARN_ONs
Currently the condition for these WARN_ONs is reversed and they are placed
before the actual check whether we are going to write to that register. So if
the codec implements the register_writable callback we'll get a warning for each
writable register when syncing the register cache.
While we are at it change the check to use snd_soc_codec_writable_register
instead of open-coding it.
- ASoC: snd_soc_codec_{readable,writable}_register change default to true
Change the default return value of snd_soc_codec_{readable,writable}_register to
true when no codec specific callback for this function is given. Otherwise all
registers of that codec will neither be readable nor writable, which is most
certainly not what we want.
- ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
GFP_ATOMIC is not needed here, use GFP_KERNEL instead.
- ASoC: Allow idle_bias_off to be specified in CODEC drivers
If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
- ASoC: Fix check for symmetric rate enforcement
The ASoC core tries to not enforce symmetric rates when
two streams open simultaneously. It does so by checking
rtd->rate being zero. This works exactly once after booting
because it is not set to zero again when the streams close.
Fix this by setting rtd->rate when no active stream is left.
[This leads to lots of warnings about not enforcing the symmetry in some
situations as there's a race in the userspace API where we know we've
got two applications but don't know what rates they want to set.
-- broonie ]
- ASoC: Add ADAU1373 codec support
This patch adds support for the Analog Devices ADAU1373 audio codec.
- ASoC: soc-pcm: Remove unused global mutex
Since commit b8c0dab9bf3373010e857a8d3f1b594c60a348dd
"ASoC: core - PCM mutex per rtd",
the global pcm_mutex is not being used any more.
- ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_lzo_get_blksize
Since commit aea170a099793abcd0e6de46b947458073204241
"ASoC: soc-cache: Add reg_size as a member to snd_soc_codec",
the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize.
- ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_flat_cache_init
Since commit d779fce5d79525d66269c8f6e430e1515d697f3d
"ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache",
the codec_drv pointer variable is not used any more.
- ASoC: soc-jack: Fix checking return value of request_any_context_irq
request_any_context_irq() returns a negative value on failure.
On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED.
- ASoC: Support !CONFIG_REGMAP builds
Since we changed regmap to be selected and register per bus rather than
via the core only we can't rely on it being enabled by the ASoC core.
Support compiling it out.
Reported-by: Axel Lin <axel.lin@gmail.com>
- ASoC: soc-io: Fix CONFIG_REGMAP_I2C/SPI guards to support regmap modules
When CONFIG_REGMAP_I2C/SPI are m, CONFIG_REGMAP_I2C_MODULE is set in the
pre-processor instead of CONFIG_REGMAP_I2C. This removes SND_SOC_I2C as a
valid option for snd_soc_codec_set_cache_io()'s control parameter, and
causes any ASoC regmap-using codec built as a module to fail to initialize.
- ASoC: soc-io: Add CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI guards for regmap_init_i2c/regmap_init_spi
In the case of "make da8xx_omapl_defconfig;make", the SPI support is disabled.
Thus calling regmap_init_spi in soc-io.c has below build error.
ERROR: "regmap_init_spi" [sound/soc/snd-soc-core.ko] undefined!
make[1]: *** [__modpost] Error 1
make: *** [modules] Error 2
This patch fixes the build error by adding CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI
guards for regmap_init_i2c/regmap_init_spi.
- ASoC: Add regmap as a control type
Allow drivers to set up their own regmap API structures. This is mainly
useful with MFDs where the core driver will have set up regmap at the
minute, though it may make sense to push the existing regmap setup out
of the core into the drivers.
- ASoC: Use new register map API for ASoC generic physical I/O
Remove all the ASoC specific physical I/O code and replace it with calls
into the regmap API. The bulk write code can only be used safely if all
regmap calls are locked with the CODEC lock, we need to add bulk support
to the regmap API or replace the code with an open coded loop (though
currently it has no users...).
- ASoC: Trivial formatting fix in soc-core.c
Utterly trivial but it annoys me.
- ASoC: mxs: add asoc configuration files
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: dapm - Add DAPM stream completion event.
In preparation for Dynamic PCM (AKA DSP) support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs
after completion of stream events.
- ASoC: Rename WM8915 to WM8996
For marketing reasons the part will be called WM8996. In order to avoid
user confusion rename the driver to reflect this.
- ASoC: core: make comments fit the code
In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.
- ASoC: Mark cache as dirty when suspending
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.
The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.
- ASoC: twl6040: Convert into TWL6040 MFD child
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
- MFD: twl4030-codec -> twl4030-audio: Rename the driver
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
- ASoC: Don't use codec->control_data in bulk write
In order to facilitate merging with the register map I/O replace the use
of control_data for the bulk writes with direct lookup of the client data
from the device.
- ASoC: pcm - rename snd_codec_close() to snd_pcm_close().
Make sure we follow naming convention for all PCM ops.
- ASoC: WM8983: Initial driver
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.
- ASoC: core - Add platform IO tracing
Trace platform IO just like CODEC IO.
- ASoC: core - Add convenience register for platform kcontrol and DAPM
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
- ASoC: core - Add platform widget IO
Allow platform driver widgets to perform any IO required for DAPM.
- ASoC: core - Add API call to register platform kcontrols.
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
- ASoC: core - Add platform read and write.
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
- ASoC: core - Make platform probe more like codec probe.
In preparation for ASoC dynamic PCM support (AKA ASoC DSP)
Platform will also support DAPM so separate out the probe function
to simplify the code (just like the codec probe).
- ASoC: Add ADAV80x codec driver
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.
- ASoC: add WM8782 ADC Codec Driver
[zonque@gmail.com: transform to new ASoC structure]
- ASoC: add STA32X codec driver
[zonque@gmail.com: transform to new ASoC structure]
- ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others. There's a pending patch to make the directory build as standard
but it's not merged yet.
- ASoC: Add ADAU1701 codec driver
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.
- ASoC: Move register I/O code into a separate file
For clarity and to help ongoing refactoring in this area create a new file
to contain the physical I/O functions, separating them out from the cache
operations.
- ASoC: Factor out redundant read() functions
We've got a whole bunch of functions which just call straight through to
do_hw_read(). Simplify this situation by removing them and using hw_read()
directly.
- ASoC: core - PCM mutex per rtd
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
- ASoC: core - Separate out PCM operations into new file.
In preparation for Dynamic PCM support (AKA DSP support).
There will be future patches that add support to allow PCMs to be dynamically
routed to multiple DAIs at startup and also during stream runtime. This patch
moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will
in simplify the ASoC core features into distinct files.
- ASoC: Suppress restore of default register values for rbtree cache sync
Currently the rbtree code will write out the entire register map when
doing a cache sync which is wasteful and will slow things down. Check
to see if the value we're about to write is the default and don't bother
restoring it if it is, either the value will have been retained or the
device will have been reset and holds the value already.
We should really store the defaults in the nodes but this resolves the
immediate issue.
- ASoC: core - Allow components to probe/remove in sequence.
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
- ASoC: core - Optimise and refactor pcm_new() to pass only rtd
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
- ASoC: Add context parameter to card DAPM callbacks
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
- ASoC: Specify target bias state directly as a bias state
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
- ASoC: Enforce the mask in snd_soc_update_bits()
Avoids issues if someone does a read followed by restore and doesn't mask
out only the bits being updated.
- ASoC: Don't set invalid name string to snd_card->driver field
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters. The commit
2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.
A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
- ASoC: Check for NULL register bank in snd_soc_get_cache_val()
- ASoC: Use explicit endianness conversion in snd_soc_16_8_write()
- ASoC: Use cpu_to_be16() in 8x16 write
- ASoC: Convert 7x9 write to use cpu_to_be16()
Run the data through cpu_to_be16() so it's at least clear what we're up to.
- ASoC: core: Don't schedule deferred_resume_work twice
For cards that have two or more DAIs, snd_soc_resume's loop over all
DAIs ends up calling schedule_work(deferred_resume_work) once per DAI.
Since this is the same work item each time, the 2nd and subsequent
calls return 0 (work item already queued), and trigger the dev_err
message below stating that a work item may have been lost.
Solve this by adjusting the loop to simply calculate whether to run the
resume work immediately or defer it, and then call schedule work (or not)
one time based on that.
Note: This has not been tested in mainline, but only in chromeos-2.6.38;
mainline doesn't support suspend/resume on Tegra, nor does the mainline
Tegra ASoC driver contain multiple DAIs. It has been compile-checked in
mainline.
- ASoC: Convert 16x16 write to use cpu_to_be16()
Make it clear what we're doing.
- ASoC: soc-cache: Cache a pointer to the last accessed rbnode
Whenever we are doing a read or a write through the rbtree code, we'll
cache a pointer to the rbnode. To avoid looking up the register
everytime we do a read or a write, we first check if it can be found in
the cached register block, otherwise we traverse the rbtree and finally
cache the rbnode for future use.
- ASoC: soc-cache: Block based rbtree compression
This patch prepares the ground for the actual rbtree optimization patch
which will save a pointer to the last accessed rbnode that was used
in either the read() or write() functions.
Each rbnode manages a variable length block of registers. There can be no
two nodes with overlapping blocks. Each block has a base register and a
currently top register, all the other registers, if any, lie in between these
two and in ascending order.
The reasoning behind the construction of this rbtree is simple. In the
snd_soc_rbtree_cache_init() function, we iterate over the register defaults
provided by the driver. For each register value that is non-zero we
insert it in the rbtree. In order to determine in which rbnode we need
to add the register, we first look if there is another register already
added that is adjacent to the one we are about to add. If that is the case
we append it in that rbnode block, otherwise we create a new rbnode
with a single register in its block and add it to the tree.
In the next patch, where a cached rbnode is used by both the write() and the
read() functions, we also check if the register we are about to add is in the
cached rbnode (the least recently accessed one) and if so we append it in that
rbnode block.
- ASoC: core - remove superfluous new line.
- ASoC: core - fix module reference counting for CPU DAIs
Currently CODEC and platform drivers have their module reference count
incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference
count incremented in soc_bind_dai_link().
CPU DAIs should have their reference count incremented in soc_probe_dai_link()
just like the CODEC and platform drivers.
- ASoC: core: Don't set "(null)" as a driver name
Commit 22de71b ("ASoC: core - allow ASoC more flexible machine name")
writes "(null)" to driver name string in struct snd_card if card->driver_name
is NULL. This causes segmentation faults with some user space ALSA utilities
like aplay and arecord.
Fix this by using the card->name if no driver name is specified.
- ASoC: Asahi Kasei AK4641 codec driver
A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800
among others.
- ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning
Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following
warning:
In file included from arch/x86/include/asm/uaccess.h:573,
from include/linux/poll.h:14,
from include/sound/pcm.h:29,
from include/sound/ac97_codec.h:31,
from sound/soc/soc-core.c:34:
In function 'copy_from_user',
inlined from 'codec_reg_write_file' at
sound/soc/soc-core.c:252:
arch/x86/include/asm/uaccess_64.h:65:
warning: call to 'copy_from_user_overflow' declared with
attribute warning: copy_from_user() buffer size is not provably
correct
presumably due to buf_size being signed causing GCC to fail to
see that buf_size can't become negative.
- ASoC: Reintroduce do_spi_write()
There is an unfortunate difference in return values between spi_write()
and i2c_master_send() so we need an adaptor function to translate.
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
- ASoC: core - allow ASoC more flexible machine name
Allow ASoC machine drivers to register a driver name
and a longname. This allows user space to determine
the flavour of machine driver.
- ASoC: Use spi_write() for SPI writes
do_spi_write() is just an open coded copy of do_spi_write() so we can
delete it and just call spi_write() directly. Indeed, as a result of
recent refactoring all the SPI write functions are just very long
wrappers around spi_write() which don't add anything except for some
pointless copies so we can just use spi_write() as the hw_write
operation directly. It should be as type safe to do this as it is to do
the same thing with I2C and it saves us a bunch of code.
- ASoC: Remove byte swap in 4x12 SPI write
snd_soc_4_12_spi_write() contains a byte swap. Since this code was written
for an Analog CODEC on a Blackfin reference board it appears that this is
done because while Blackfin is little endian the CODEC is big endian (as
are most CODECs).
Push this up into the generic 4x12 write function and use cpu_to_be16() to
do the byte swap so things are more regular and things work on both CPU
endiannesses.
- ASoC: Don't squash 16x8 registers down to 8 bits
Currently we'll force all registers to fit in 8 bits before passing
down to the I/O function. Looks like a cut'n'paste bug.
- ASoC: soc-cache: Allow codec->cache_bypass to be used with snd_soc_hw_bulk_write_raw()
If we specifically want to write a block of data to the hw bypassing the
cache, then allow this to happen inside snd_soc_hw_bulk_write_raw().
- ASoC: Create codec DAPM widgets before calling the codecs probe function
This allows to create DAPM routes depending on those widgets in the
codecs probe function. This is helpful when supporting similar codecs
with minor differences in the DAPM routing with the same driver.
Something similar has already been done for cards in commit
a841ebb9 (ASoC: Create card DAPM widgets early so they can be used in
callbacks).
- ASoC: Don't crash on PM operations
The move over to exposing snd_soc_register_card() let the initialisation
of the driver data we use to find the card in PM operations go AWOL. Fix
this by setting the driver data when we register the card.
- ASoC: Move DAPM widget debugfs entry creation to snd_soc_dapm_new_widgets
Currently debugfs entries for a DAPM widgets are only added in
snd_soc_dapm_debugfs_init. If a widget is added later (for example in the
dai_link's probe callback) it will not show up in debugfs.
This patch moves the creation of the widget debugfs entry to
snd_soc_dapm_new_widgets where it will be added after the widget has been
properly instantiated.
As a side-effect this will also reduce the number of times the DAPM widget list
is iterated during a card's instantiation.
Since it is possible that snd_soc_dapm_new_widgets is invoked form the codecs or
cards probe callbacks, the creation of the debugfs dapm directory has to be
moved before these are called.
- ASoC: Move DAPM debugfs directory creation to snd_soc_dapm_debugfs_init
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.
- ASoC: Free the card's DAPM context
Free the card's DAPM context when the card is removed.
- ASoC: Declare const properly for enum texts
The enum texts are supposed to be const char * const []. Without the
second const, it gets compile warnings like
sound/soc/codecs/max98095.c:607:2: warning: initialization discards qualifiers from pointer target type
- ASoC: Work around allmodconfig failure
- ASoC: Add more natural support for no-DMA DAIs
Since we can now support multiple platforms allow machines to not specify
a platform in a DAI link. Since the rest of the code requires that we have
a struct device for all objects we do this by substituting in a dummy
device that we register automatically.
- ASoC: Allow platform drivers to have no ops structure
- ASoC: fix two ident style problems
- ASoC: remove unused comment
`type` parameter is not longer used in `snd_soc_codec_set_cache_io`,
so remove this line.
- ASoC: Make struct snd_soc_card's dapm_widgets and dapm_routes const
Those should not be modified (and are not) by the core code, so make them const.
This also makes them consistent with the same members of snd_soc_codec.
- ASoC: Create card DAPM widgets early so they can be used in callbacks
This helps with things like setting up the initial state.
- ASoC: Add initial WM1250-EV1 Springbank audio I/O module driver
The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas
reference platform provides a simple audio I/O with an independant clock
domain, intended to simulate cellular modem and bluetooth subsystems
within the platform.
The card supports some limited GPIO based control but this is currently not
implemented.
- ASoC: Add WM8915 CODEC driver
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.
- ASoC: Add soc_remove_dai_links
card->num_rtd should be 0 after soc_romve_dai_link
- ASoC: SSM2602: add SPI support
The ssm2602 codec has a SPI interface as well as I2C, so add the simple
bit of glue to make it usable.
- ASoC: Add data based control initialisation for CODECs and cards
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
- ASoC: fix config error path
initialize ret to invalid value so that when we reach the config error path in
soc_pcm_open, it will return the correct error code. without this patch, though
config error path is executed, soc_pcm_open will return 0 in
snd_pcm_open_substream and then cause double release of substream.
- ASoC: check channel mismatch between cpu_dai and codec_dai
Suppose we have:
cpu_dai
channels_min = 1
channels_max = 1
codec_dai
channels_min = 2
channels_max = 2
This is a mismatch that should not happen, however according to the current
code, the result of runtime->hw will be:
channels_min = 2
channels_max = 1
We better spot it early. This patch checks this mismatch.
- ASoC: Add max98095 CODEC driver
This patch adds the MAX98095 CODEC driver.
- ASoC: Tegra: Suspend/resume support
ASoC machine drivers that are their own platform_driver (as opposed to
those using the soc-audio platform_driver) need to explicitly set up
power-management operation callbacks.
To avoid cut/paste, snd_soc_pm_ops also needs to be exported.
- ASoC: Fix comment width in soc-cache.c
Lines should be less than 80 columns.
- ASoC: Remove excessively verbose logging on I2C write
We don't need to log every I2C transfer, and certainly not at error level.
- ASoC: Fix to avoid compile error
This patch fixes to avoid compile error when ASoC codec doesn't use I2C
nor SPI on snd_soc_hw_bulk_write_raw().
- ASoC: format_register_str: Don't clip register values
wordsize is used as the textual width of a register address.
regsize is used as the textual width of a register value.
The assignments to these values were swapped. In the case of WM8903, which
has 8-bit register addresses and 16-bit register values, this caused the
register values to be clipped to 2 digits instead of the full 4.
- ASoC: Name jack GPIOs based on jack not codec
snd_soc_jack_gpio has a name field. Use that name when registering the IRQ,
since this is far more informative than the codec driver name. This shows
up in /proc/interrupts.
- ASoC: soc-cache: Warn on syncing any non-writable registers
When syncing the cache, if the driver has given us a writable_register()
callback, use it to check if we are syncing a non-writable register
and if so warn the user.
- ASoC: soc-cache: Fix indentation issues
- sound: Fixup the last user of the old irq functions
I had seen that before, but now that I removed set_irq_wake it broke.
- ASoC: fix sorting order of codecs in kconfig
- ASoC: ad73311: drop I2C requirement
The AD73311 codec does not use I2C, so don't require it in Kconfig.
- ASoC: Add snd_soc_codec_{readable,writable}_register()
Provide the top level ASoC core functions for indicating whether
a given register is readable or writable.
- ASoC: Add default snd_soc_default_writable_register() callback
By using struct snd_soc_reg_access for the read/write/vol attributes
of the registers, we provide callbacks that automatically determine whether
a given register is readable/writable or volatile.
- ASoC: soc-cache: Return -ENOSYS instead of -EINVAL
These functions fail with -EINVAL if the corresponding callbacks
are not implemented. Change them to return -ENOSYS as it is more
appropriate for unimplemented callbacks.
- ASoC: soc-cache: Factor-out the SPI write code
The handling of all snd_soc_x_y_spi_write() functions is similar.
Create a separate function and update all callers to use it.
- ASoC: soc-cache: Factor-out the hw_read() specific code
The handling of all snd_soc_x_y_read() functions is similar.
Factor it out into a separate function and update all callers.
- ASoC: soc-cache: Factor-out the hw_write() specific code
The handling of all snd_soc_x_y_write() functions is similar.
Factor it out into a separate function and update all functions
to use it.
- ASoC: Add control_type in snd_soc_codec
This is mainly used by the soc-cache code to easily determine the
currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we
can distinguish it if it is not initialized or set.
- ASoC: soc-cache: Introduce raw bulk write support
As it has become more common to have to write firmware or similar
large chunks of data to the hardware, add a function to perform
raw bulk writes that bypass the cache. This only handles volatile
registers as we should avoid getting out of sync with the actual
cache.
- ASoC: soc-cache: Factor-out the I2C read code
The handling of all snd_soc_x_y_read_i2c() functions is similar.
Make a generic I2C read function and update all callers to use it.
- [media] ASoC: WL1273 FM radio: Access I2C IO functions through pointers
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.
Also update the year in the copyright statement.
- ASoC: Remove bogus check for register validity in debugfs write
Since not all registers need to be cached and the cache is entirely
optional anyway we shouldn't be checking that a register is in the
cached range. If the register is invalid then the actual I/O code
can determine that and report an error.
Similarly, the step size can and should be enforced by the lower level
code if it's important.
- ASoC: Add MAX9850 codec driver
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM
- ASoC: Add LM4857 to SND_SOC_ALL_CODECS
- ASoC: Add driver for the dfbmcs320 bluetooth module
This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used
on the neo1973 boards.
The patch also modifies the neo1937_wm8753 sound board driver to use the new
driver instead of registering the bluetooth DAI manually.
Previously there was a name mismatch between the bluetooth DAI and the bluetooth
DAI link and the sound card was not instantiated, with this patch the issue is
no longer present and sound support works again.
- ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
- ASoC: Warn rather than set a silly constraint when we can't do symmetry
Symmetric rate configuration can fail if the second stream starting tries
to apply the symmetric constraint before the first stream has got far
enough to pick a rate. Rather than try to enforce a nonsensical rate of
0Hz log a warning and allow the application to carry on. Things might go
wrong later on but the user will know about it and there's unlikely to be
lasting damage.
- ASoC: Refactor symmetric_rates check to reduce indentation
- ASoC: Provide CODEC clocking operations and API calls
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
- ASoC: Add DAPM widget and path data to CODEC driver structure
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
- ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module
This patch moves the code for the lm4857 AMP from the neo1973_wm8753 sound
board driver to its own module.
The lm4857 is a generic AMP IC and not specific to the neo1973.
- ASoC: Add missing debugfs conditionals
- ASoC: Add TI tlv320aic32x4 codec support.
This patch adds support for tlv320aic3205 and tlv320aic3254 codecs.
It doesn't include miniDSP support for aic3254.
- ASoC: Add a late_probe() callback to cards
This is run after the DAPM widgets and routes are added, allowing setup
of things like jacks using the routes. The main card probe() is run before
anything else so can't be used for this purpose.
- ASoC: Allow card DAPM widgets and routes to be set up at registration
These will be added after all devices are registered and allow most DAI
init functions in machine drivers to be replaced by simple data.
Regular controls are not supported as the registration function still
works in terms of CODECs.
- ASoC: Add a per-card DAPM context
This means that rather than adding the board specific DAPM widgets to a
random CODEC DAPM context they can be added to the card itself which is
a bit cleaner. Previously there only was one DAPM context and it was
tied to the single supported CODEC.
- ASoC: Add Freescale SGTL5000 codec support
Add Freescale SGTL5000 codec support.
Supported features:
- line-in and mic input
- headphone and line-out output
- line-in bypass ADC and DAC to headphone
- 16, 20, 24, 32 bit audio
- 8 ~ 96k sample rates
- ASoC: Pass the jack to jack notifiers
We're currently not passing anything and this will make the card and so on
more discoverable.
- ASoC: Add kerneldoc for jack_status_check callback
- ASoC: Allow GPIO jack detection to be configured as a wake source
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
- ASoC: Allow use sleeping gpio in soc-jack
It is safe to use sleeping gpio in snd_soc_jack_gpio_detect as it is not
called from interrupt context. This avoids WARN_ON from __gpio_get_value
if sleeping gpio is registered for jack.
- ASoC: add support for multiple jack types
This patch adds soc-jack support for adding voltage zones and for
detecting jack type
- ASoC: soc-cache: dereferencing before checking
The patch c358e640a66 "ASoC: soc-cache: Add trace event for
snd_soc_cache_sync()" introduced a dereference of "codec->cache_ops"
before we had checked it for NULL.
I pulled the check forward, and then pulled everything in an indent
level.
- ASoC: soc-core: Support debugfs entries larger than PAGE_SIZE bytes
For some codecs with large register maps, it was not possible to dump
all registers via the codec_reg file but only up to PAGE_SIZE bytes.
This patch fixes this problem.
- ASoC: Update PM ifdefs for exported suspend/resume
- ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size()
- ASoC: soc-core: Ensure codec_reg has fixed length fields
Make the format of the codec_reg file more easily parsable. Remove
the header field which gives the codec name. These changes are important
when it comes to extend the debugfs codec_reg file to dump more than
PAGE_SIZE bytes to make it easier to calculate offsets within the
file.
We still need to handle the case when the snd_soc_read() call fails
and <no data: %d> is outputted.
- ASoC: Move card list initialization to snd_soc_register_card
All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was
only called for cards backed by a "soc-audio" platform device, via
soc_probe(). However, it's also needed for cards backed by other platform
devices, and registered directly via snd_soc_register_card().
- ASoC: Add card driver data
Provide driver data for cards within the card structure. To simplify the
implementation of the PM operations we don't use the struct device driver
data as this is used by the core to retrieve the card in callbacks from
the device model and PM core.
- ASoC: soc-core: Increment codec and platform driver refcounts before probing
Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers"
moved codec and platform driver refcount increments from soc_bind_dai_link
to more appropriate places.
Adjust a little them so that refcounts are incremented before executing the
driver probe functions.
- ASoC: Add subsequence information to seq_notify callbacks
Allows drivers to distinguish which subsequence is being notified when
they get called back.
- ASoC: Make cache status available via debugfs
Could just as well live in sysfs but sysfs doesn't have the simple
value export helpers debugfs does.
- ASoC: Export card PM callbacks for use in direct registered cards
Allow hookup of cards registered directly with the core to the PM
operations by exporting the device power management operations to
modules, also exporting the default PM operations since it is
expected that most cards will end up using exactly the same setup.
Note that the callbacks require that the driver data for the card be
the snd_soc_card.
- ASoC: Replace pdev with card in machine driver probe and remove
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
- ASoC: Use card rather than soc-audio device to card PM functions
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.
- ASoC: Fix type for snd_soc_volatile_register()
We generally refer to registers as unsigned ints (including in the
underlying CODEC driver operation).
- ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
This patch makes it easy to see when the syncing process begins and
ends. You can also enable the snd_soc_reg_write tracepoint to see
which registers are being synced.
- ASoC: CS4271 codec support
Added support for CS4271 codec to ASoC.
- ASoC: soc-cache: Apply the cache_bypass option
Incorporate the use of the cache_bypass functionality in the
syncing functions. The snd_soc_flat_cache_sync() need not be
hooked as there is no performance benefit from using the
cache_bypass option.
- ASoC: soc-cache: Introduce the cache_bypass option
This is primarily needed to avoid writing back to the cache
whenever we are syncing the cache with the hardware. This gives a
performance benefit especially for large register maps.
- ASoC: Provide per widget type callback when executing DAPM sequences
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
- ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()
The .card member of the snd_soc_pcm_runtime structure pointed to by the
snd_soc_dai_link.init() argument used to be initialized before the
function being called. This has changed, probably unintentionally,
after recent refactorings. Since the function implementations are free
to make use of this pointer, move its assignment back before the
function is called to avoid NULL pointer dereferences.
Created and tested on Amstrad Delta againts linux-2.6.38-rc2
- ASoC: Fix module refcount for auxiliary devices
Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers"
moved codec driver refcount increments from soc_bind_dai_link into
soc_probe_codec.
However, the commit didn't remove try_module_get from soc_probe_aux_dev so
the auxiliary device reference counts are incremented twice as the
soc_probe_codec is called from soc_probe_aux_dev too.

SoC MXS

- Add missing build-stub for soc/mxs/*
- ASoC: mxs-saif: convert to clk_prepare/clk_unprepare
The patch converts mxs-saif driver to clk_prepare/clk_unprepare by
using helper functions clk_prepare_enable/clk_disable_unprepare.
- ASoC: mxs-saif: remove function in platform_data
Add master_mode and master_id in platfrom_data since it's board
specific and board knows it.
Then we can remove the function pointer in platfrom_data to make
the driver more devicetree friendly.
- ASoC: mxs: Add appropriate MODULE_ALIAS()
- ASoC: mxs: Add missing MODULE_LICENSE("GPL")
The sound driver refuses to load as module, because of the missing
MODULE_LICENSE("GPL").
The file header indicates that the driver is indeed published under
the GPL.
- ASoC: Convert mxs directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: mxs: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
- sound: Revert "ASoC: mxs: correct 'direction' of device_prep_dma_cyclic"
This reverts commit dbec3b30a601791717bc5bb827e210c3b5d6e067 as it
should never have been applied to the ASoC tree at all, let alone 3.2.
- ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
The commit 49920bc (dmaengine: add new enum dma_transfer_direction)
changes the type of parameter 'direction' of device_prep_dma_cyclic
from dma_data_direction to dma_transfer_direction.
- ASoC: mxs: Add appropriate MODULE_ALIAS()
- ASoC: mxs: Add missing MODULE_LICENSE("GPL")
The sound driver refuses to load as module, because of the missing
MODULE_LICENSE("GPL").
The file header indicates that the driver is indeed published under
the GPL.
- ASoC: keep pointer to resource so it can be freed
Add a new variable for storing resources accessed subsequent to the one
accessed using request_mem_region, so the one accessed using
request_mem_region can be released if needed.
The resource variable names are also changed to be more descriptive.
This code is also missing some calls to iounmap.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r@
expression E, E1;
identifier f;
statement S1,S2,S3;
@@
if (E == NULL)
{
... when != if (E == NULL || ...) S1 else S2
when != E = E1
*E->f
... when any
return ...;
}
else S3
// </smpl>
- ASoC: mxs-saif: add record function
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.
The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.
2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
- ASoC: mxs-sgtl5000: add record function
- sound/soc/mxs/mxs-saif.c: add missing kfree
Move the test on pdev->id before the kzalloc to avoid requiring kfree when
the test fails. This fix was suggested by Wolfram Sang.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: mxs-saif: clear clk gate first before register setting
Saif needs clear clk gate first before writing registers or the write
will not success.
The original xx_get_mclk function clear clk gate after mclk setting
that may cause the former mclk setting unwork, then the real output
mclk maybe inaccurate.
Placing the clear before setting mclk to avoid such an issue.
We also have to clear clk gate in startup instead of in prepare function.
- ASoC: mxs: add mxs-sgtl5000 machine driver
The driver only supports playback firstly.
For recording, as we have to use two saif instances to implement full
duplex (playback & recording) due to hardware limitation, we need to
figure out a good design to fit in ASoC.
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: mxs: add mxs-pcm driver
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: mxs: add mxs-saif driver
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
- ASoC: mxs: add asoc configuration files
Tested-by: Wolfram Sang <w.sang@pengutronix.de>

SoC Marvell Kirkwood

- ARM: Orion: Get address map from plat-orion instead of via platform_data
Use an getter function in plat-orion/addr-map.c to get the address map
structure, rather than pass it to drivers in the platform_data
structures. When the drivers are built for none orion platforms, a
dummy function is provided instead which returns NULL.
Tested-by: Michael Walle <michael@walle.cc>
- ASoC: Convert kirkwood-t5325 to table based DAPM init
- ASoC: Use dai_fmt in kirkwood-t5325 machine driver
- ASoC: Use dai_fmt in kirkwood-openrd machine driver
- ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c
Since commit db33f4de "ARM: Orion: Remove address map info from all platform data structures",
the dram is removed from struct kirkwood_asoc_platform_data.
This patch fixes below build error:
CC sound/soc/kirkwood/kirkwood-i2s.o
sound/soc/kirkwood/kirkwood-i2s.c: In function 'kirkwood_i2s_dev_probe':
sound/soc/kirkwood/kirkwood-i2s.c:444: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram'
sound/soc/kirkwood/kirkwood-i2s.c:450: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram'
make[3]: *** [sound/soc/kirkwood/kirkwood-i2s.o] Error 1
make[2]: *** [sound/soc/kirkwood] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
- ASoC: kirkwood: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert kirkwood directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C
SND_KIRKWOOD_SOC_T5325 selects SND_SOC_ALC5623, but SND_SOC_ALC5623 needs
CONFIG_I2C. So we need to make SND_KIRKWOOD_SOC_T5325 depend on I2C,
otherwise I got below build error if CONFIG_I2C is not selected.
CC sound/soc/codecs/alc5623.o
sound/soc/codecs/alc5623.c: In function 'alc5623_i2c_probe':
sound/soc/codecs/alc5623.c:1002: error: implicit declaration of function 'i2c_smbus_read_word_data'
sound/soc/codecs/alc5623.c:1009: error: implicit declaration of function 'i2c_smbus_read_byte_data'
sound/soc/codecs/alc5623.c: In function 'alc5623_modinit':
sound/soc/codecs/alc5623.c:1096: error: implicit declaration of function 'i2c_add_driver'
sound/soc/codecs/alc5623.c: In function 'alc5623_modexit':
sound/soc/codecs/alc5623.c:1108: error: implicit declaration of function 'i2c_del_driver'
make[3]: *** [sound/soc/codecs/alc5623.o] Error 1
make[2]: *** [sound/soc/codecs] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Also fix the same issue for SND_KIRKWOOD_SOC_OPENRD.
- ASoC: kirkwood-i2s: Add __devexit_p at necessary place
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
We have __devexit annotation for kirkwood_i2s_dev_remove(), thus add __devexit_p
at necessary place.
- ASoC: Fix trivial build regression in Kirkwood I2S
A fix merged in 3.1-rc2 introduced a small regression, this should get it
to build again.
- sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
Adjust the goto to jump to the error handling code that includes kfree.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier x;
expression E1!=0,E2,E3,E4;
statement S;
iterator I;
@@
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>

SoC NVIDIA Tegra

- ASoC: tegra: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Tegra+WM8903 machine: Add device tree binding
This driver is parameterized in two ways:
a) Platform data, which supplies the set of GPIOs used by the driver.
These GPIOs can now be parsed out of device tree.
b) Machine-specific DAPM route arrays embedded into the ASoC machine
driver itself. Historically, the driver picks the appropriate array
to use using machine_is_*(). The driver now requires this array to
be parsed from device tree when instantiated through device tree,
using the core ASoC support for this parsing.
Based on work by John Bonesio, but significantly reworked since then.
- ASoC: Tegra machine ASoC driver for boards using ALC5332 codec
At this stage only Toshiba AC100/Dynabook supported.
- ASoC: Tegra: Move DAS configuration into DAS driver
Move DAS routing setup into the DAS driver itself. This removes the need
to duplicate this in each machine driver, of which we'll soon have three.
An added advantage is that the machine drivers no longer call the Tegra20-
specific DAS functions by name, so the machine driver no longer needs to
be split up into Tegra20 and Tegra30 versions.
If individual machine drivers need a different routing setup to this
default, they can still call the DAS functions to set that up.
Long-term, DAS will be a codec driver, and user-space will be able to
control its routing, possibly within constraints that the machine driver
sets up. Configuring the DAS routing from the DAS driver is a very slight
move in that direction.
- ASoC: Tegra I2S: Add device tree binding
- ASoC: Convert tegra_spdif to use module_platform_driver()
Use the module_platform_driver() macro which makes
the code smaller and a bit simpler.
- ASoC: Convert WM8903 MICBIAS to a supply widget
Also rename it to MICBIAS to reflect the pin name and help any out of tree
users notice the change.
- ASoC: TrimSlice machine: Set the new fully_routed flag
Set card.fully_routed to request the ASoC core calculated unused codec
pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded
calls.
- ASoC: Tegra+WM903 machine: Set the new fully_routed flag
Set card.fully_routed to request the ASoC core calculated unused codec
pins, and call snd_soc_dapm_nc_pin() for them. Remove the open-coded
calls.
- ASoC: Tegra I2S: Remove dependency on pdev->id
When devices are instantiated from device-tree, pdev->id is set to -1.
Rework the driver so it doesn't depend on the ID.
Tegra I2S instantiated from board files are configured with pdev
name "tegra-i2s" and ID 0 or 1. The driver core then names the
device "tegra-i2s.0" or "tegra-i2s.1". This is not changing.
When a device is instantiated from device-tree, it will have
pdev->name="" and pdev->id=-1. For this reason, the pdev->id value is
not something we can rely on.
This patch doesn't actually change any names though:
When a device is instantiated from device-tree, the overall device name
will be "${unit_address}.${node_name}". This causes issues such as
clk_get() failures due to lack of a device-name match. To solve that,
AUXDATA was invented, to force a specific device name, thus allowing
dev_name() to return the same as the non-device-tree case. Tegra
currently uses AUXDATA for the I2S controllers. Eventually, AUXDATA will
go away, most likely replaced by phandle-based references within the
device tree.
- ASoC: Tegra TrimSlice machine: Use devm_ APIs and module_platform_driver
module_platform_driver saves some boiler-plate code.
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.
- ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver
module_platform_driver saves some boiler-plate code.
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.
- ASoC: Tegra DAS: Add device tree binding
- ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver
module_platform_drive saves some boiler-plate code.
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.
- ASoC: Tegra: Move DAS configuration into machine drivers
This removes potentially machine-specific routing knowledge from the
I2S driverinto the machine drivers, which is better equipped to know
what the appropriate routing configuration is.
- ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver
module_platform_drive saves some boiler-plate code.
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.
- ASoC: Tegra PCM: Use module_platform_driver
This saves some boiler-plate code.
- ASoC: Tegra: sparse cleanup
Fixes the following sparse warnings:
sound/soc/tegra/tegra_das.c:215:8: warning: Using plain integer as NULL pointer
sound/soc/tegra/tegra_das.c:237:8: warning: Using plain integer as NULL pointer
sound/soc/tegra/tegra_pcm.c:370:32: warning: symbol 'tegra_pcm_platform' was not declared. Should it be static?
- ASoC: Remove unused srate variable in tegra_spdif_hw_params
- ASoC: tegra: Staticise tegra_i2s_dai and tegra_spdif_dai
- ASoC: Tegra: wm8903 machine driver: Drop Ventana support
Board file support for Ventana is not yet mainlined, and probably won't
ever be given the move to Device-Tree. Consequently, the Ventana entry
is being removed from arch/arm/tools/mach-types in the next merge window,
since it was registered over a year ago.
This will also remove function machine_is_ventana(), which is used by
the ASoC Tegra WM8903 machine driver. This will cause compilation
failures. Drop Ventana support to resolve this.
Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to
configure itself from Device-Tree, and hence we'll be able to re-instate
Ventana support just by creating a .dts file for the board.
Also note that Aebl support is in a similar boat. However, that board
isn't scheduled for deprecation for at least another 5 months, and
perhaps we will have completely removed non-Device-Tree support from
tegra_wm8903.c by then and/or adjusted mach-types policy.
- ASoC: Tegra: wm8903 machine driver: Allow re-insertion of module
Two issues were preventing module snd-soc-tegra-wm8903.ko from being
removed and re-inserted:
a) The speaker-enable GPIO is hosted by the WM8903 chip. This GPIO must
be freed before snd_soc_unregister_card() is called, because that
triggers wm8903.c:wm8903_remove(), which calls gpiochip_remove(), which
then fails if any of the GPIOs are in use. To solve this, free all GPIOs
first, so the code doesn't care where they come from.
b) We need to call snd_soc_jack_free_gpios() to match the call to
snd_soc_jack_add_gpios() during initialization. Without this, the
call to snd_soc_jack_add_gpios() fails during any subsequent modprobe
and initialization, since the GPIO and IRQ are already registered. In
turn, this causes the headphone state not to be monitored, so the
headphone is assumed not to be plugged in, and the audio path to it is
never enabled.
- ASoC: Tegra: tegra_pcm_deallocate_dma_buffer: Don't OOPS
Not all PCM devices have all sub-streams. Specifically, the SPDIF driver
only supports playback and hence has no capture substream. Check whether
a substream exists before dereferencing it, when de-allocating DMA
buffers in tegra_pcm_deallocate_dma_buffer.
- ASoC: Tegra: Implement SPDIF CPU DAI
This is a minimal driver for the Tegra SPDIF controller.
In hardware, the SPDIF output signal is always routed to any active HDMI
display controllers, and may also be routed to external pins on Tegra
using the pinmux.
- ASoC: Tegra: I2S: s/clk_get_sys/clk_get/
The clock needed by the I2S driver is associated with the I2S device name
in the standard fashion. Hence, use clk_get(dev) instead of clk_get_sys(clk_name).
- ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.
Luckily, the dropped register writes historically had no effect:
TEGRA_I2S_TIMING: The value we wrote was the reset default.
TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.
However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.
- soc/tegra/Kconfig - add missing depends on MACH_HAS_SND_SOC_TEGRA_WM8903
Merge synchronization fix.
- ASoC: Remove default settings from Tegra Kconfig
There needs to be a strong reason for overriding the Kconfig default.
- ASoC: Tegra: Enable Kaen HP_MUTE at boot
We want the default state of the HP_MUTE signal to be asserted, so that
the headphones are muted before the first audio playback. Without this,
the headphones are left unmuted until shortly after the first audio
playback completes.
- ASoC: tegra: TrimSlice machine support
- ASoC: Tegra: wm8903: s/code/data/ for control/widget/maps
Replace calls to a variety of registration functions by updating
struct snd_soc_card snd_soc_tegra_wm8903 to directly point at the
various control/widget/map tables instead. The ASoC core now
performs any required registration based on these data fields.
(Applying Mark's TrimSlice review comments to the existing driver)
- ASoC: Tegra: Retrieve card from DAPM context not codec
Card widgets are created in the card's DAPM context, not any codec's DAPM
context. Hence, w->codec==NULL. Instead, find the card from the widget
through the DAPM context of the widget, not the codec of the widget.
- ASoC: Tegra: Don't return mclk_changed from utils_set_rate
Only the clock programming code needs to know whether the clocks changed,
and that is encapsulated within tegra_asoc_utils_set_rate(). The machine
driver's call to snd_soc_dai_set_sysclk(codec_dai, ...) is safe
irrespective of whether the clocks changed.
(Applying Mark's TrimSlice review comments to the existing driver)
- ASoC: Tegra: wm8903: Remove redundant drvdata clears
When the driver is not initialized/registered, nothing should be touching
these fields anyway, so there's no point clearing them out.
(Applying Mark's TrimSlice review comments to the existing driver)
- ASoC: Tegra: wm8903 probe: Don't call machine_is_*()
This machine driver is a platform driver, and hence will only be
instantiated on the correct machines. Hence, there is no need to
check the current machine during probe.
(Applying Mark's TrimSlice review comments to the existing driver)
- ASoC: Tegra: Support more boards
* Ventana is identical to Harmony.
* Seaboard, Kaen, and Aebl are all pretty similar, mainly with slightly
different sets of GPIOs, and slightly different WM8903 pin connectivity.
- ASoC: Tegra: Don't store snd_soc_jack_gpio in an array
Storing the struct in an array makes the assignments to the GPIO member a
little non-obvious, and is pointless when there's only a single GPIO.
(I thought I fixed this during the review cycle when first submitting this
driver, but I guess I overlooked that)
- ASoC: Tegra: Rename Kconfig SND_TEGRA_SOC_* to SND_SOC_TEGRA_*
The previous commit renames SND_TEGRA_SOC_HARMONY to SND_TEGRA_SOC_WM8903.
While we're breaking people's .config files, rename all Tegra/SOC-related
Kconfig variables to be more consistent with at least the core codec
variables. Note that there exist machines that name their variables both
ways.
- ASoC: Tegra: Rename harmony.c to tegra_wm8903.c
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the file in advance to reflect this.
Fix the content of tegra_wm8903.c to match the rename; replace references
to Harmony board with something more generic.
* s/struct tegra_harmony/struct tegra_wm8903/
* s/harmony/machine/ # variable name
* Similar rename for some functions
* Similar comment fix
* Similar MODULE_DESCRIPTION fix
- ASoC: Tegra: Fix compile when debugfs not enabled
The prototype of the inline dummy version of tegra_i2s_debug_add
was not consistent with the real version.
Reported-by: Rhyland-Klein <rklein@nvidia.com>
- ARM: Tegra: select MACH_HAS_SND_SOC_TEGRA_WM8903
CONFIG_SND_SOC_TEGRA_WM8903 is useful for many Tegra boards. To avoid the
ASoC tegra/Kconfig enumerating them all, instead have the Tegra machine
Kconfig select MACH_HAS_SND_SOC_TEGRA_WM8903 where appropriate, and have
SND_SOC_TEGRA_WM8903 depend on this.
[Redid ASoC diff so it applies. -- broonie]
- ASoC: Tegra: Suspend/resume support
ASoC machine drivers that are their own platform_driver (as opposed to
those using the soc-audio platform_driver) need to explicitly set up
power-management operation callbacks.
To avoid cut/paste, snd_soc_pm_ops also needs to be exported.
- ASoC: Tegra: Fix error handling in DMA channel alloc
tegra_dma_allocate_channel() returns NULL on errors, not an error pointer.
- ASoC: Tegra: Move utilities to separate module
The utilities will be required by every machine driver. Including the
utility object directly into every machine driver causes a build failure
if the modules are actually built into the kernel, since each will define
the symbols exported by the utility file. Solve this by moving the
utility object into a separate module.
- ASoC: Tegra: Add MODULE_ALIAS
With the appropriate MODULE_ALIAS in place, the audio modules will be
automatically loaded; there is no longer a need for manual modprobes.
- ASoC: Tegra: Harmony: Explicitly set mic enables
Harmony has both an external mic (a regular mic jack) and an internal mic
(a 0.1" two-pin header on the board).
The external mic is connected to the WM8903's IN1L pin, and is supported
by the current driver.
The internal mic is connected to the WM8903's IN1R pin, and is not supported
by the current driver.
It appears that no Harmony systems were shipped with any internal mic
connected; users were expected to provide their own. This makes the
internal mic connection less interesting.
The WM8903's Mic Bias signal is used for both of these mics. For each mic,
a GPIO drives a transistor which gates whether the mic bias signal is
actively connected to that mic, or isolated from it.
The dual use of the mic bias for both mics makes a general-purpose complete
implementation of mic detection using the mic bias complex. So, for
simplicity, the internal mic is currently ignored by the driver.
This patch configures the relevant GPIOs to enable the mic bias connection
to the external mic, and disable the mic bias connection to the internal
mic. Note that in practice, this is the default state if these GPIOs aren't
configured.
- ASoC: Harmony: Call snd_soc_dapm_nc_pin
- ASoC: Tegra: Harmony: Implement mic detection
* Add jack definition for mic jack
* Request wm8903 to enable mic detection
* Force mic bias on, since it's required for mic detection
- ASoC: Remove redundant -codec from WM8903 driver name
It causes noisy -codecs to appear in things like .codec_name.
- ASoC: Tegra: Harmony: Add switch control for speaker
- ASoC: Tegra: Harmony: Add headphone jack detection
- ASoC: Tegra: Harmony: Remove redundant !!
gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO
driver has been fixed to work this way, so remove the redundant !!
previously required when it didn't accept values >1.
- ASoC: Tegra: I2S: Use dev_err not pr_err
- ASoC: Tegra: utils: Don't use global variables
Instead, have the machine driver provide storage for the utility data
somehow.
For Harmony in particular, store this within struct tegra_harmony, itself
referenced by snd_soc_card's drvdata.
- ASoC: Tegra: Harmony: Use dev_err not pr_err
- ASoC: Tegra: Harmony: Fix indentation issue.
Indent with TABs not spaces.
- ASoC: Tegra: Harmony: Support the internal speaker
Add DAPM widget definitions for the internal speaker paths. Currently, this
path is always enabled while playback is active.
Add code to control the speaker amplifier GPIO.
The GPIO is requested during _init, since that's the first time it is
guaranteed that the WM8903 module is loaded, probed, and hence has exported
its GPIO chip.
- ASoC: Tegra: Harmony: Don't use soc-audio platform device
Previously, snd-soc-tegra-harmony internally instantiated a platform device
object whenever the module was loaded. Instead, switch to a more typical model
where arch/arm/mach-tegra defines a platform device, and snd-soc-tegra-harmony
acts as a driver for such a platform device.
Define a new struct tegra_harmony to store driver data in the future.

SoC Nuvoton NUC900

- upstream sync merge fix
- ASoC: mxs: Add appropriate MODULE_ALIAS()
- ASoC: nuc900: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert nuc900 directory to module_platform_driver
Factor out some boilerplate code.
- ARM: 7175/1: add subname parameter to mfp_set_groupg callers
commit 798681bf "ARM: 7158/1: add new MFP implement for NUC900"
adds subname parameter for mfp_set_groupg.
Thus add subname parameter to the callers.
- ASoC: Staticise nuc900_dma_getposition()
It is not used outside this driver so no need to make the symbol global.
- ASoC: nuc900-pcm: remove unused variable 'dai'
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_new':
sound/soc/nuc900/nuc900-pcm.c:321: warning: unused variable 'dai'
- ASoC: add missing clk_put to nuc900-ac97
This goto is after the call to clk_get, so it should go to the label that
includes a call to clk_put.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
expression e1,e2;
statement S;
@@
e1 = clk_get@p1(...);
... when != e1 = e2
when != clk_put(e1)
when any
if (...) { ... when != clk_put(e1)
when != if (...) { ... clk_put(e1) ... }
* return@p3 ...;
} else S
// </smpl>

SoC PXA2xx 88PM860x Tavor EVB3

- ASoC: Use dai_fmt in tavorevb3 machine driver
- ASoC: Convert tavorevb3 to table based DAPM init
Also remove a unsued ret variable to silence the build warning.

SoC PXA2xx Aeronix Zipit Z2

- ASoC: Convert z2 to table based DAPM init
- ASoC: Use dai_fmt in z2 machine driver
- ASoC: Remove redundant -codec from WM8750
- ASoC: PXA: Z2: Fix codec pin name
MONO was renamed to MONO1.
- ASoC: PXA: z2: Mute internal speaker when headphones are connected

SoC PXA2xx Corgi

- ASoC: pxa: Convert corgi to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: Convert corgi to table based DAPM and control init
- ASoC: Use dai_fmt in corgi machine driver
- ASoC: pxa: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Fix CODEC device name for Corgi
Got typoed in the multi-component changes.

SoC PXA2xx E740

- ASoC: pxa: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: pxa: Convert e740_wm9705 to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: Convert e740_wm9705 to use gpio_request_one()
- ASoC: correct pxa AC97 DAI names
Correct names for pxa AC97 DAI are pxa2xx-ac97 and pxa2xx-ac97-aux. Fix
that for all PXA platforms.

SoC PXA2xx E750

- ASoC: pxa: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: pxa: Convert e750_wm9705 to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: Convert e750_wm9705 to use gpio_request_one()
- ASoC: correct pxa AC97 DAI names
Correct names for pxa AC97 DAI are pxa2xx-ac97 and pxa2xx-ac97-aux. Fix
that for all PXA platforms.

SoC PXA2xx E800/WM9712

- ASoC: pxa: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: pxa: Convert e800_wm9712 to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: Convert e800_wm9712 to use gpio_request_one()
- ASoC: Fix error handling in e800_init to free gpios
- ASoC: correct pxa AC97 DAI names
Correct names for pxa AC97 DAI are pxa2xx-ac97 and pxa2xx-ac97-aux. Fix
that for all PXA platforms.

SoC PXA2xx MIOA701

- ASoC: Convert pxa directory to module_platform_driver
Factor out some boilerplate code.

SoC PXA2xx Poodle

- ASoC: Use dai_fmt in poodle machine driver
- ASoC: Convert poodle to table based DAPM and control init
- ASoC: pxa: Convert poodle to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.

SoC PXA2xx Spitz

- ASoC: Convert spitz to table based DAPM and control init
- ASoC: Use dai_fmt in spitz machine driver
- ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
- ASoC: Remove redundant -codec from WM8750
- ASoC: Properly handle spitz MIC GPIO
This patch firstly restructurizes the code a bit by getting rid of continuous
checking for machine type in spitz_mic_bias().
Then the patch properly requests the MIC GPIO in the spitz_init() and frees it
in spitz_exit().

SoC PXA2xx Tosa

- ASoC: pxa: Convert tosa to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
- ASoC: Replace pdev with card in machine driver probe and remove
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.

SoC PXA2xx Zylonite

- ASoC: use a valid device for dev_err() in Zylonite
A recent conversion has introduced references to &pdev->dev, which does
not actually exist in all the contexts it's used in.
Replace this with card->dev where necessary, in order to let
the driver build again.
- ASoC: zylonite: set .codec_dai_name in initializer
Fix the initialization of .codec_dai_name in zylonite_dai initializer,
do not mix it with the initialization of .codec_name which is set
already a few lines above.
- ASoC: Replace pdev with card in machine driver probe and remove
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
- ASoC: Use card rather than soc-audio device to card PM functions
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.

SoC PXA2xx saarb

- ASoC: Convert saarb to table based DAPM init
Also remove a unused ret variable to silence the build warning.
- ASoC: Use dai_fmt in saarb machine driver

SoC S6000

- ASoC: s6000: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Convert s6000 directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: s6000-pcm: remove unused variable 'dai'

SoC SH7760 AC97

- ASoC: sh: Add .owner to struct snd_soc_card
Add missing .owner of struct snd_soc_card. This prevents the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: Use core pm_runtime callbacks for fsi
Now that the core holds a pm_runtime reference to the device while the
link is active there is no need for the driver to do so.
- ASoC: Use core pm_runtime callbacks for siu_dai
Now that the core holds a pm_runtime reference to the device while the
link is active there is no need for the driver to do so.
- ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform
Both sh4_hac_dai and sh7760_soc_platform are changed to static
by multi-component patch and they are not used in sh7760-ac97.c now.
- ASoC: Convert sh directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: fsi-ak4642: modify specification method of FSI / ak464x
Current fsi-ak4642 was using id_entry name in order to specify
FSI port and ak464x codec.
But it was no sense, no flexibility.
Platform can specify FSI/ak464x pair by this patch.
- ASoC: fsi: add valid data position control support
FSI2 can control valid data position, like
package in front/back or stream mode (16bit x 2).
But current fsi driver is assuming it was in-back.
- ASoC: fsi: fixup compile warning
This patch fixup below warning
${linux}/sound/soc/sh/fsi.c:442:3:\
warning: passing argument 1 of '__fsi_reg_read' makes pointer\
from integer without a cast
${linux}/sound/soc/sh/fsi.c:517:3: \
warning: passing argument 1 of '__fsi_reg_write' makes pointer\
from integer without a cast
${linux}/sound/soc/sh/fsi.c:663:3: \
warning: passing argument 1 of '__fsi_reg_mask_set' makes pointer\
from integer without a cast
- ASoC: sh: use correct __iomem annotations
This removes a few unnecessary type casts and avoids
sparse warnings.
- ASoC: Staticise sh4_ssi_dai
- ASoC: sh: fsi-hdmi: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
- ASoC: sh: fsi-da7210: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
- ASoC: sh: fsi-ak4642: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
- ASoC: sh: fsi: add fsi_hw_startup/shutdown
This patch is preparation of cleanup suspend/resume patch.
- ASoC: sh: fsi: cleanup suspend/resume
Current FSI driver was using saved_xxx variable for suspend/resume.
OTOH, the start and stop of power/clock are controlled by
fsi_hw_startup/fsi_hw_shutdown in current FSI driver.
The all necessary registers value are set by fsi_hw_startup.
So, if fsi_hw_shutdown is called when "suspend" is generated,
and fsi_hw_startup is called at "resume",
the saved_xxx are not needed.
- ASoC: sh: fsi: remove fsi_module_init/kill
FSIA/B ports is enabled by default when power-on,
and current FSI is supporting RuntimePM.
In addition, current fsi_module_init/kill doesn't care
simultaneous playback/recorde.
This mean FSI port control is not needed.
This patch remove fsi_module_init/kill
- ASoC: sh: fsi: make sure fsi_stream_push/pop access by spin lock
fsi_stream_push/pop might be called in same time.
This patch protect it.
- ASoC: sh: fsi: remove pm_runtime from fsi_dai_set_fmt.
pm_runtime_get/put_sync were used to access FSI register in fsi_dai_set_fmt
which is called when ALSA probe.
But this register value will disappear after pm_runtime_put_sync
if platform is supporting RuntimePM.
To solve this issue, this patch adds new variable for format,
and remove pm_runtime_get/put_sync from fsi_dai_set_fmt.
- ASoC: sh: fsi: tidyup unclear variable naming
Some variables on this driver were a unclear naming,
and were different unit (byte, frame, sample).
And some functions had wrong name
(ex. it returned "sample width" but name was "fsi_get_frame_width").
This patch tidy-up this issue, and the minimum unit become "sample".
Special thanks to Takashi YOSHII.
- ASoC: sh: fsi: irq control moves to fsi_port_start/stop
Using fsi_irq_enable/disable in fsi_port_start/stop is very natural.
This patch is preparation of cleanup suspend/resume patch.
- ASoC: sh: fsi: add fsi_set_master_clk
Current FSI driver is using set_rate call back function which is for
master mode.
By this patch, it is used from fsi_set_master_clk.
This patch is preparation of cleanup suspend/resume patch.
- ASoC: sh: fsi: tidyup parameter of fsi_stream_push
It is possible to create buff_len and period_len
from substream->runtime.
This patch is preparation of tidyup unclear variable naming patch.
- ASoC: sh: fsi: Add module/port clock control function
The FIFO of each port were always working though it was not used
in current FSI driver.
This patch add module/port clock control function for fixing it.
This patch is also caring suspend/resume.
Reviewed-by: Simon Horman <simon@horms.net>
- ASoC: sh: fsi: add dev_pm_ops :: suspend/resume
Current FSI driver sets important settings when probing.
And it are not set again as long as driver is not bind again.
This mean FSI driver will lost it from register
if suspend/resume are happen.
This patch save important settings for suspend/resume.
Reviewed-by: Simon Horman <simon@horms.net>
- ASoC: sh: fsi: add fsi_is_clk_master function
If FSI port is clock master, it use set_rate function
which is callback from platform,
and it is not necessary to call it if FSI port is clock slave.
Current FSI driver called this callback if platform provide it.
This patch modify it.
Reviewed-by: Simon Horman <simon@horms.net>
- ALSA: add a module alias to the FSI driver
This patch enables FSI driver autoloading on sh-mobile systems.
Reviewed-by: Simon Horman <horms@verge.net.au>
- ASoC: fsi: driver safely remove for against irq
free_irq and pm_runtime_disable should be called before
snd_soc_unregister_xxx
- ASoC: fsi: modify vague PM control on probe
- ASoC: fsi: take care in failing case of dai register
- ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format
Current format selection of FSI-codecs depended on platform information for FSI,
and chip default settings for codecs. It is not understandable/formal method.
This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt.
But FSI can use I2S/PCM and SPDIF format today.
It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF.
So, this patch change FSI platform information to have DAI/SPDIF mode.
If platform selects DAI mode (default),
FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt,
and if it is SPDIF mode, FSI become SPDIF format.
- ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info
Current FSI driver assumed master->info is not NULL.
This patch allow NULL in master->info
- ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv
- ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection
This patch add platform_device_id which can control
PortA/PortB for FSI2-HDMI from platform data.
- ASoC: sh: fsi: Add snd_soc_dai_set_fmt support
This patch add snd_soc_dai_ops :: set_fmt to FSI driver and
select master/slave clock mode by snd_soc_dai_set_fmt on
fsi-xxx.c instead of platform infomation code.
This patch remove fsi_is_master function which is no longer needed.
- ASoC: sh: fsi: Add fsi_get_priv_frm_dai function
- ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support
This patch support LEFT_J / I2S only for now

SoC Texas Instruments OMAP

- ASoC: cx20442: add bias control over a platform provided regulator
Now that a regulator device for controlling the codec chip reset state
over a platform agnostic regulator API is available on the only board
using this driver so far, extend the driver with a bias control function
which will request virtual power to the codec chip from that virtual
regulator, and will supersede the present implementation existing at the
sound card level.
Thanks to the regulator sharing mechanism, both the old (the sound card)
and the new (the codec) implementations should coexist smoothly until
the sound card file is updated. For this to work as expected, update the
sound card .set_bias_level callback to not touch codec->dapm.bias_level.
While extending the cx20442 structure, drop unused control_type member.
Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1:
ams-delta: set up a regulator over the modem reset GPIO pin".
- ASoC: omap: Add .owner to struct snd_soc_card
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
- ASoC: omap-mcbsp: Enable FIFO usage on OMAP4
Allow McBSP FIFO configuration from ASoC dai driver
on OMAP4 platform.
- ASoC: Staticise rx51_aux_dev
- ASoC: Use core pm_runtime callbacks for omap-mcpdm
Now that the core holds a pm_runtime reference to the device while the
link is active there is no need for the driver to do so.
- ASoC: Use core pm_runtime callbacks for omap-dmic
Now that the core holds a pm_runtime reference to the device while the
link is active there is no need for the driver to do so.
- ASoC: sdp4430: Add support for digital microphones
OMAP4 SDP/Blaze boards have digital microphones.
- ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC
Add support for OMAP4 Digital Microphone interface.
- ASoC: Convert omap directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: sdp3430: Let core to deal with the DAPM widgets
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
- ASoC: osk5912: Let core to deal with the DAPM widgets
Pass the DAPM widgets/routes via the snd_soc_card struct
to core. With this change we do not need the init function
since the remaining snd_soc_dapm_enable_pin calls are
not needed.
- ASoC: n810: Let the core to register DAPM widgets/routes and controls
Pass the DAPM widgets/routes and static controls via the
snd_soc_card struct to core. In this way the machine driver
does not need to handle the DAPM widgets/routes.
- ASoC: am3517evm: Let core to deal with the DAPM widgets
Pass the DAPM widgets/routes via the snd_soc_card struct
to core. With this change we do not need the init function
since the remaining snd_soc_dapm_enable_pin calls are
not needed.
- ASoC: sdp4430: No need to call dapm_pin_enable at init time
Widgets are connected by default.
- ASoC: sdp4430: Let core to deal with the DAPM widgets
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
- ASoC: zoom2: No need to call dapm_pin_enable at init time
Widgets are connected by default.
- ASoC: zoom2: Let core to deal with the DAPM widgets
Pass the DAPM widgets/routes via the snd_soc_card struct
to core.
- ASoC: OMAP machines: Remove soc_dapm_sync() call from init
No need to call soc_dapm_sync at init time.
- ASoC: fix checkpatch.pl error in omap-mcbsp
- ASoC: omap-pcm: Fix the no period wakeup implementation
After omap_request_dma the BLOCK_IRQ is enabled as default
configuration for the channel.
If we are requested for no period wakeup, we need to disable
the BLOCK_IRQ in order to not receive any interrupts.
- ASoC: omap: Use single hw_params callback in sdp3430 and zoom2
There is no need to use two hw_params callbacks in sdp3430 and zoom2 as
thet are now identical. Use instead the same snd_soc_ops structure and
hw_params callback for both DAI links.
- ASoC: omap: Convert bunch of machine drivers to use init time DAI format
- ASoC: omap-mcbsp: Prepare for init time DAI format setting
Before commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the
dai_link") expectation for omap-mcbsp was that snd_soc_dai_set_fmt is to be
called first in machine hw_params callback before other CPU DAI functions.
Thus it was enough that only omap_mcbsp_dai_set_dai_fmt cleared the
mcbsp->regs structure. [Note that this was pure convention, it's always
been OK to set things on init -- broonie]
Now this doesn't hold anymore since machine drivers can set the DAI format
only once on init time and thus mcbsp->regs may get out of sync when other
CPU DAI functions are modifying them dynamically with different values
between the calls. Therefore clear the accessed mcbsp->regs bits and
bitfields in other functions too.
- ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions.
- ASoC: sdp4430: Fix string for FM input name
The name contains invalid valid character (/), which
causes problems when trying to create the debugfs
directory structure:
ASoC: Failed to create Aux/FM Stereo In debugfs file
- ASoC: sdp4430: Configure McPDM offset cancellation
Based on the values from twl6040 codec (HSOTRIM L/R) we can configure
the McPDM offset cancellation.
- ASoC: omap-mcpdm: API to configure offset cancellation
The offset cancellation values can be different from board to board, even
on the same HW platform.
Provide a way for the machine drivers to configure the McPDM offset
cancellation.
- ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’
Fixes the following compile time warning:
omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’
- ASoC: omap-mcpdm: Correct the supported number of channels
OMAP4 McPDM supports 5 downlink (playback), and
3 uplink (capture) channels.
- ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
Attempt to change McBSP CLKS source while another stream is active is not
safe after commit d135865 ("OMAP: McBSP: implement functional clock
switching via clock framework") in 2.6.37.
CLKS parent clock switching using clock framework have to idle the McBSP
before switching and then activate it again. This short break can cause a
DMA transaction error to already running stream which halts and recovers
only by closing and restarting the stream.
This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce
l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a
severe timeout error and does BUG_ON().
Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk
if the McBSP is already active. This test should have been here just from
the beginning anyway.
- ASoC: twl6040/sdp4430: Change legacy DAI name
Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to
be more intuitive.
- ASoC: omap-mcpdm: Replace legacy driver
Reasons for the replacement:
The current driver for McPDM was developed to support the legacy mode only.
In preparation for the ABE support the current driver stack need the be
replaced.
The new driver is much simpler, easier to extend, and it also fixes some of the
issues with the old stack.
Main changes:
- single file for omap-mcpdm (mcpdm.c/h removed)
- Define names for registers, bits cleaned up, prefixed
- Full-duplex audio operation (arecord | aplay) has been fixed
- Less code
McPDM need to be turned off after all streams has been stopped.
This might cause pop noise on the output, if the codec's DAC is
still powered at this time.
- ASoC: OMAP4: McPDM: Convert to hwmod/omap_device
In order to probe, and operate correctly, the OMAP McPDM driver needs to
be converted to use hwmod.
The device name has been changed to probe the driver.
Replace the clk_* with pm_runtime_* calls to manage the clocks correctly.
Missing request_mem_region/release_mem_region added.
- ASoC: omap-mcpdm: Fix threshold and dma configuration
DMA packet_size must be configured based on the McPDM FIFO threshold
value, number of channels.
Due to the FIFO operation the DMA muse be configured differently for
playback, and capture.
At the same time fix the McPDM threshold values used for playback, and
capture to avoid broken code.
- ASoC: omap: Fix build errors in ams-delta
Fix "error: too few arguments to function 'ams_delta_set_bias_level'"
build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC:
Add context parameter to card DAPM callbacks") by adding dapm context
to ams_delta_set_bias_level calls.
- ASoC: omap: Update e-mail address of Jarkko Nikula
My gmail account got disabled and I'm not going to reopen it.
- ASoC: twl6040: Move PLL selection to codec driver
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.
- ASoC: twl6040: Convert into TWL6040 MFD child
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
- MFD: twl4030-codec -> twl4030-audio: Rename the driver
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
- treewide: Fix recieve/receive typos
Just spelling fixes.
- ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards
Add machine driver for HDMI audio on OMAP4 boards. This driver is
in charge of putting together the HDMI audio codec and the CPU DAI
and register the HDMI sound card with ALSA.
- ASoC: OMAP: Add CPU DAI driver for HDMI
Addition of the HDMI CPU DAI driver for OMAP4. This driver is in
charge of configuring DMA settings for HDMI. Also, it finds
the HDMI video device and determines if audio playback can proceed.
- ASoC: OMAP: Update Makefile and Kconfig for HDMI audio
Update Makefile and Kconfig to build HDMI audio support for
OMAP4 SDP and Panda boards.
- omap: Remove support for omap2evm
The board support has never been merged for it as noticed
by Russell King <linux@arm.linux.org.uk>. So let's remove the
related dead code.
- ASoC: omap-mcbsp: Remove restrictive checks for cpu type
Current checks for cpu type were too restrictive leading
to failures for other silicons in same family.
The problem was found while testing audio playback on
AM37x and AM35x processors. But should exist on OMAP36xx
as well.
- ASoC: omap-pcm: Period wakeup disabling on OMAP2+
Allow disabling ALSA period wakeup interrupts.
This can only be done on OMAP2+ (2/3/4), since there
we can chain the DMA.
- ASoC: RX51: Update e-mail address
- ASoC: omap-pcm: Update e-mail address
- ASoC: omap-mcbsp: Update e-mail address
- ASoC: omap: rx51: Enable McBSP2 sidetone
McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.
Sidetone feature can be tested with following commands:
(set up codec input and output paths)
# Enable and configure sidetone
amixer -D hw:0 set 'McBSP2 Sidetone' on
amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
# Do not loop audio via CPU
arecord -f dat >/dev/null |aplay /dev/zero
- ASoC: McBSP: get hw params from McBSP driver
Removed the use of macros to obtain base address and DMA channel number.
Instead use the McBSP driver API's that passes base address and DMA
channel number to the client driver.
- ASoC: omap: rx51: Add FM transmitter support
Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out
signals of TLV320AIC34 than TPA6130 headphone amplifier.
This patch adds route to transmitter and "FM Transmitter" control to keep
route active when needed.
- ASoC: omap: rx51: Report headset insertion instead of video out cable
It is more usefull to report headset instead of video out cable in response
to jack insertion as this is more usual use-case and because now the headset
feature is supported. Automatic accessory detection is not possible at the
moment so most sensible static accessory type have to be used.
- ASoC: omap: rx51: Add headset support
This patch adds support for headset microphone in Nokia RX-51/N900. The mic
signal from audio jack is routed to codec A LINE1L via two switches and the
mic bias is coming from codec B part.
First switch is the tv-out switch that is already supported and the second
switch selects between voltage detection circuit and codecs. As there is
no use for voltage detection at the moment the second switch is connected
statically to codecs in rx51_soc_init.
Headset can be active when control "Jack Function" is set to "Headset".
- ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpio
Just slight cleanup to be sync with upcoming change.
- omap: Start using CONFIG_SOC_OMAP
We want to have just CONFIG_ARCH_OMAP2, 3 and 4. The rest
are nowadays just subcategories of these.
Search and replace the following:
ARCH_OMAP2420 SOC_OMAP2420
ARCH_OMAP2430 SOC_OMAP2430
ARCH_OMAP3430 SOC_OMAP3430
No functional changes.
- ASoC: AM3517: Update codec name after multi-component update
The i2c client device name (".2-001a" in this case, including
the separator period) for the AIC23 codec on the TI AM3517-EVM
was appended to the codec_name member of am3517evm_dai to
resolve the names mismatch happening in soc_bind_dai_link(),
due to which the card was not getting registered.
- ASoC: omap: rx51: Add earphone support
Earphone in Nokia RX-51/N900 is connected to left HP output of B part of the
TLV320AIC34 dual codec. In RX-51 the codec A is used as a traditional codec
and the codec B as an auxiliary device.
Audio from codec A goes via the codec B to earphone:
MONO_LOUT of A -> LINE2R of B (B interconnects) -> HPLOUT of B -> Earphone.
Take earphone into use by utilizing the recent ASoC auxiliary and
cross-device support.
- ASoC: omap: rx51: Add stereo output support to audio jack
Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier.
This patch adds support for it and stereo output can be active when
"Jack Function" == "TV-OUT" || "Headphone".
As the TPA6130 can output very high volume levels the output is limited
with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources.
- ASoC: Amstrad Delta: fix const related build error
The Amstrad Delta ASoC driver used to override the digital_mute()
callback, expected to be not provided by the on-board CX20442 CODEC
driver, with its own implementation. While this is still posssible when
substituting the whole empty snd_soc_dai_driver.ops member (the CX20442
case), replacing snd_soc_dai_ops.digital_mute only is no longer correct
after the snd_soc_dai_driver.ops member has been constified, and results
in build error.
Drop this actually not used code path in hope the CX20442 driver never
provides its own snd_soc_dai_ops structure.
Created and tested against linux-2.6.38-rc2

Soc Codec STA32X

- ASoC: sta32x: add workaround for ESD reset issue
sta32x resets and loses all configuration during ESD test.
Work around by polling the CONFA register once a second
and restore all coeffcients and registers when CONFA
changes unexpectedly.
- ASoC: sta32x: add platform data definition
Add a structure for platform specific configuration and use it,
thereby removing a few FIXMEs which marked hard-coded values.

Soc PXA2xx Imote 2

- ASoC: Use dai_fmt in imote2 machine driver
- ASoC: pxa: Convert imote2 to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.

Soc PXA2xx Magician

- ASoC: Remove unused rate variable in magician_playback_hw_params

Soc PXA2xx Raumfeld

- ASoC: Add missing platform_device_put in raumfeld_audio_init error path
- ASoC: fix raumfeld platform
Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support)
broke support for Raumfeld platforms as it didn't take into account the
different hardware features on individual devices.
In particular, Raumfeld speakers have no S/PDIF output, so the members
of the snd_soc_card struct must be set dynamically.
- ASoC: PXA: formatting
- ASoC: Use card rather than soc-audio device to card PM functions
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.

Soc iPAQ hx4700

- ASoC: Use dai_fmt in hx4700 machine driver
- ASoC: Convert pxa directory to module_platform_driver
Factor out some boilerplate code.
- ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails
- ASoC: add iPAQ hx4700 machine driver
AK4641 connected via I2S and I2C, jack detection via GPIO.
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