- 10 2月, 2012 1 次提交
-
-
由 Masanari Iida 提交于
Correct spelling "memroy" to "memory" in sound/soc/codecs/twl4030.c Signed-off-by: NMasanari Iida <standby24x7@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 27 1月, 2012 1 次提交
-
-
由 Axel Lin 提交于
Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver for devices can unconditionally support idle_bias_off. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 20 1月, 2012 1 次提交
-
-
由 Peter Ujfalusi 提交于
Core can set the msbits constraint in behalf of the dai. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 02 12月, 2011 1 次提交
-
-
由 Lars-Peter Clausen 提交于
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. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 28 11月, 2011 1 次提交
-
-
由 Mark Brown 提交于
Factors out a bit of boilerplate. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 23 11月, 2011 1 次提交
-
-
由 Lars-Peter Clausen 提交于
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> Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 11 10月, 2011 1 次提交
-
-
由 Peter Ujfalusi 提交于
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 06 10月, 2011 1 次提交
-
-
由 Peter Ujfalusi 提交于
Use the macros for controls require custom get/put function. This is to make sure that the soc_mixer_control is used consistently among the drivers. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Cc: Arun KS <arunks@mistralsolutions.com> Cc: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 04 7月, 2011 2 次提交
-
-
由 Peter Ujfalusi 提交于
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. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: NSamuel Ortiz <sameo@linux.intel.com>
-
由 Peter Ujfalusi 提交于
Rename the driver, and header file from twl4030-codec to twl4030-audio. To avoid breakage change depending drivers at the same time. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> CC: Misael Lopez Cruz <misael.lopez@ti.com> Acked-by: NSamuel Ortiz <sameo@linux.intel.com>
-
- 27 5月, 2011 1 次提交
-
-
由 Samuel Ortiz 提交于
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. Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Cc: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg@slimlogic.co.uk> Cc: Dmitry Torokhov <dtor@mail.ru> Signed-off-by: NSamuel Ortiz <sameo@linux.intel.com>
-
- 31 3月, 2011 1 次提交
-
-
由 Lucas De Marchi 提交于
Fixes generated by 'codespell' and manually reviewed. Signed-off-by: NLucas De Marchi <lucas.demarchi@profusion.mobi>
-
- 23 3月, 2011 1 次提交
-
-
由 Andres Salomon 提交于
The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: NAndres Salomon <dilinger@queued.net> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NSamuel Ortiz <sameo@linux.intel.com>
-
- 15 12月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
twl4030 series of codecs supports S32_LE with msbits=24. Replace the S24_LE with S32_LE format, and add constraint for 24msbit in case of 32 S32_LE format. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 22 11月, 2010 1 次提交
-
-
由 Jarkko Nikula 提交于
There is no need to include soc-dapm.h since soc.h includes it. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 19 11月, 2010 1 次提交
-
-
由 Axel Lin 提交于
As we allocate memory for twl4030 in twl4030_codec_probe(), twl4030_codec_remove() is a better place to free the memory. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 06 11月, 2010 1 次提交
-
-
由 Liam Girdwood 提交于
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 26 10月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
Change the busy loop delays with usleep_range or msleep calls. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 15 9月, 2010 1 次提交
-
-
由 Jarkko Nikula 提交于
This assignment is done by the snd_soc_register_codec so there is no need to redo it in probe function of a codec driver. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 12 8月, 2010 2 次提交
-
-
由 Peter Ujfalusi 提交于
Add back the register restore call, when the codec driver is removed. This does not affect normal operation, but it is usefull when debugging audio through the twl4030 class codecs. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Liam Girdwood 提交于
This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: NTimur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: NRyan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com> Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com> Reviewed-by: NJassi Brar <jassi.brar@samsung.com> Signed-off-by: NSeungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: NTimur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 04 8月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
Fix the ordering problem in DAPM domain, when the user changes between digital and analog sources during active capture (or loopback) scenario. Before this patch, when the user changed from analog source to digital there were a short time, when the codec enabled analog mic bias (2.2 volts) instead of the correct digital mic bias (1.8 volts) to the digital microphones. This behaviour caused by the former implementation of selecting the correct type of bias. This was done at the POST_REG event of the DAPM_MUX_E("TXx Capture Route") widget. By moving the bias type selection as DAPM_SUPPLY and connecting it to the corresponding digimic widget the problematic situation can be avoided. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 27 7月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
There is no need to handle POST_PMU, POST_PMD event with the Capture Route widget. It is enough to handle POST_REG event, since that will come when the user changes the routing, and we will switch the needed bits in the registers. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 21 7月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 13 7月, 2010 2 次提交
-
-
由 Peter Ujfalusi 提交于
Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 31 5月, 2010 8 次提交
-
-
由 Peter Ujfalusi 提交于
Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
It seams at least on twl5031 that the ARXR2_APGA_CTL register does not have the same default value as it is written in the TRM. Since the codec part of the PM chip has not been actually changed according to TI, assuming, that all version has the same problem, so writing there the TRM value. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Add means for machine drivers to select the path for offset cancellation. Reset the reg cache value to the chip reset value at the same time. Machine drivers can specify which path need to be used for offset cancellation via the twl4030_setup.offset_cncl_path. For paths use the defines from include/linux/mfd/twl4030-codec.h: TWL4030_OFFSET_CNCL_SEL_* Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
There is no need for the power down wrapper. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Reset most of the codec registers to their chip reset value. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 12 5月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 10 5月, 2010 1 次提交
-
-
由 Mark Brown 提交于
The core will ensure that the device is in either STANDBY or OFF bias before suspending, restoring the bias in the driver is unneeded. Some drivers doing slightly more roundabout things have been left alone for now. Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 03 5月, 2010 2 次提交
-
-
由 Peter Ujfalusi 提交于
OUTL/R are leftovers from the original driver, and they are no longer needed. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 20 4月, 2010 1 次提交
-
-
由 Takashi Iwai 提交于
Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files in the previous commit. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
-
- 17 4月, 2010 1 次提交
-
-
由 Mark Brown 提交于
One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-