Since the core now includes deduplication in the name of CODEC
devices there's no need to add extra for the debugfs directory name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the debugfs directory is current per CODEC we should only init
it when the CODEC is initialised, otherwise we end up with errors
being generated when an attempt is made to add duplicate debugfs
entries.
Since most of this stuff is actually for the card we should refactor
but this can come later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Any subsequent revisions will have these configuration changes applied
by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the chip defaults to optimise performance of some of the DSP
functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: add AD1980 obsolete information
ASoC: register cache should be 1 byte aligned for 1 byte long register
ALSA: hda - Adding support for new IDT 92HD87XX codecs
ASoC: Fix inverted mute controls for WM8580
ALSA: HDA: Use model=auto for LG R510
ALSA: hda - Update model entries in HD-Audio-Models.txt
ALSA: hda: document VIA models
ALSA: hda - patch_nvhdmi.c: Add missing codec IDs, unify names
ALSA: hda - add support for Conexant CX20584
ALSA: hda - New snd-hda-intel model/pin config for hp dv7-4000
ALSA: hda - Fix missing stream for second ADC on Realtek ALC260 HDA codec
ALSA: hda - Make converter setups sticky
ALSA: hda - Add support for Acer ZGA ALC271 (1025:047c)
sound/oss: Adjust confusing if indentation
sound: oss: au1550_ac97.c removed duplicated #include
ASoC: Fix for changed Eureka Kconfig symbol names
Nothing should be referencing this any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the provision of a struct device for the CODEC is now mandatory
we can use container_of() to locate the struct i2c_client and struct
spi_device for relevant devices, removing the need to manually set it
in each driver.
A further patch will automate selection of the control type based on
the bus_type of the struct device, further reducing the amount of
driver code required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for adding "status = disabled" to an SSI node to incidate that it
is not wired on the board. This replaces the not-so-intuitive previous method
of omitting a codec-handle property.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The error handling code in the OF probe function of the SSI driver is not
freeing all resources correctly.
Since the machine driver no longer calls the DMA driver to provide information
about the SSI, we don't need to keep a list of DMA objects any more. In
addition, the fsl_soc_dma_remove() function is incorrectly removing *all*
DMA objects when it should only remove one.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.
The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture. This function is just a front-end for
dma_alloc_coherent(). Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs. Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Goni board based on S5PV210.
The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Aquila board based on S5PC110.
The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
This patch modify the wrong field name on Aquila board.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Should use Capture rather than ADC so the UI tools can identify their
function more readily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Sadly these aren't soft controllable and can't be read back either :(
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Otherwise debugfs gets upset when we try to create filenames with /
in them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)
From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
of_device is just an alias for platform_device, so remove it entirely. Also
replace to_of_device() with to_platform_device() and update comment blocks.
This patch was initially generated from the following semantic patch, and then
edited by hand to pick up the bits that coccinelle didn't catch.
@@
@@
-struct of_device
+struct platform_device
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Reviewed-by: David S. Miller <davem@davemloft.net>
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch fixes the error path in wm9081_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8961_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8955_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
snd_soc_register_dai() fail to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
handling if snd_soc_register_dai failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 should be kfreed if ak4642_init() return error.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz. The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.
TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.
Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.
The patch extends the original THRESHOLD mode by doing the
following:
if (period_words <= max_threshold)
Current THRESHOLD mode configuration
Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size
With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Current FSI driver id is not only 0
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
WARNING: sound/soc/au1x/snd-soc-au1xpsc-i2s.o(.data+0xa8): Section mismatch in reference from the variable au1xpsc_i2s_driver to the function .init.text:au1xpsc_i2s_drvprobe()
The variable au1xpsc_i2s_driver references
the function __init au1xpsc_i2s_drvprobe()
If the reference is valid then annotate the
variable with __init* or __refdata (see linux/init.h) or name the variable:
*_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console,
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Select wm_hubs automatically for WM8994
ASoC: Remove duplicate AUX definition from WM8776
ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
ASoC: wm8727: add a missing return in wm8727_platform_probe
ASoC: fsi: fixup wrong value setting order of TDM
ASoC: fsi: fixup clock inversion operation
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.
This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.
platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.
Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.
Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.
This patch has been tested on DM644x and OMAP-L138 EVMs.
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise all machine drivers need to do so.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.
Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
snd_soc_unregister_codec is called twice if snd_soc_register_dai fail.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
otherwise the error path will always be executed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Annotate platform probe callback with __devinit instead of plain __init.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch didn't use dev_err,
because it is difficult to get struct device here.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
channel size should be set before setting register value
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clock inversion should be specified by each flags bit.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK. The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
i2s_accurate_sck switch can be used to have a better approximate
sampling frequency.
The clock is an externally visible bit clock and it is named
i2s continuous serial clock (I2S_SCK).
The trade off is between more accurate clock (fast clock)
and less accurate clock (slow clock).
The waveform will be not symmetric.
Probably it is possible to get a better algorithm for calculating
the divider, trying to keep a slower clock as possible.
This patch has been developed against the
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When McBSP peripheral gets the clock from an external pin,
there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
and MCBSP_CLKS.
evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
hardware connection and I use MCBSP_CLKS, so I have added
this possibility.
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm)
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added two clocking options for dm365 McBSP peripheral when used
with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
from external pin and generates frame sync).
A slave clock management can be important when the external codec needs
the system clock and the bit clock synchronized (tested with uda1345).
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This adds sound support for the SmartQ board.
The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: usb/endpoint, fix dangling pointer use
ALSA: asihpi - Get rid of incorrect "long" types and casts.
ASoC: DaVinci: Fix McASP hardware FIFO configuration
ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
ALSA: usb-audio: fix UAC2 control value queries
ALSA: usb-audio: parse UAC2 sample rate ranges correctly
ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
ALSA: hda - Don't check capture source mixer if no ADC is available
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only
SND_JACK_VIDEOOUT type is reported. More types could be reported after
getting more audio features supported and necessary drivers integrated for
implementing automated accessory detection.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used
as headphone, headset or audio-video connector. This patch implements the
control 'Jack Function' which is used to select the desired function.
At the moment only TV-out without audio is supported.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.
in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF
Signed-off-by: Eric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds ASoC support for the qi_lb60 board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the JZ4740 internal codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds ASoC support for JZ4740 SoCs I2S module.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.
Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.
The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.
Solution: Mask the result so that it "wraps around", emulating
sign-extension.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* tdm slot has to be configured to get sound working on i.MX25
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.
Consequently when I2C is not set, the compilation fails [1]
This patch fixes this issues, by adding a depencdency on the related HW-
controller.
Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compiling in the MPC5200 sound drivers results in the following build error:
sound/soc/fsl/mpc5200_psc_ac97.o: In function `to_psc_dma_stream':
mpc5200_psc_ac97.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
sound/soc/fsl/efika-audio-fabric.o: In function `to_psc_dma_stream':
efika-audio-fabric.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
make[3]: *** [sound/soc/fsl/built-in.o] Error 1
make[2]: *** [sound/soc/fsl] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
This patch fixes it by declaring the inline function in the header file to
also be a static.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Cc: Jon Smirl <jonsmirl@gmail.com>
Tested-by: John Hilmar Linkhorst <John.Linkhorst@rwth-aachen.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf
Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)
During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.
https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).
The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The header contains an extern that isn't used by anything. Remove.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* master.kernel.org:/home/rmk/linux-2.6-arm:
ARM: 6164/1: Add kto and kfrom to input operands list.
ARM: 6166/1: Proper prefetch abort handling on pre-ARMv6
ARM: 6165/1: trap overflows on highmem pages from kmap_atomic when debugging
ARM: 6152/1: ux500 make it possible to disable localtimers
[ARM] pxa/spitz: Correctly register WM8750
[ARM] pxa/palmtc: storage class should be before const qualifier
ARM: 6146/1: sa1111: Prevent deadlock in resume path
ARM: 6145/1: ux500 MTU clockrate correction
ARM: 6144/1: TCM memory bug freeing bug
ARM: VFP: Fix vfp_put_double() for d16-d31
This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for i2s audio on Bluewater Systems Snapper CL15 module
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove break after return, it is not needed.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
IMX_SSI_NET : enable Network Mode
IMX_SSI_SYN : enable Synchronous Mode
IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer
This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.
I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.
We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.
There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding support for openrd client platforms. It's using
the cs42l51 codec and has one mic and one speaker plugs.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables support for the i2s controller available on kirkwood
platforms
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.ul>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for the power down wrapper.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reset most of the codec registers to their chip reset
value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch registers the WM8750 codec on a proper place on the SPITZ machine
after the WM8750 driver was converted to new API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: snd-usb-caiaq: Bump version number to 1.3.21
ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
ALSA: snd-usb-caiaq: Simplify single case to an 'if'
ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
ALSA: hda: Use LPIB for a Shuttle device
ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
ALSA: hda: Use LPIB for Sony VPCS11V9E
ALSA: usb-audio: fix feature unit parser for UAC2
ALSA: asihpi - Minor code cleanup
ALSA: asihpi - Add support for new ASI8800 family
ALSA: asihpi - Fix bug preventing outstream_write preload from happening
ALSA: asihpi - Fix imbalanced lock path in hw_message
ALSA: asihpi - Remove support for old ASI8800 family
ALSA: asihpi - Add hd radio blend functions
ALSA: asihpi - Remove unused io map functions
ALSA: usb-audio: add support for UAC2 pitch control
ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
ALSA: usb-audio: fix return values
ALSA: usb-audio: parse more format descriptors with structs
sound: Add missing spin_unlock
...
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These scales should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These should be regular rather than linear scales.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
This updates the i.MX SSI driver to make it compatible with the ASoC tree
following the move of DMA parameters from the DAI to the audio substream
object.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
