Commit Graph

7678 Commits

Author SHA1 Message Date
Timur Tabi
1a3c5a491a asoc/multi-component: fsl: add support for 36-bit physical addresses
Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.

The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture.  This function is just a front-end for
dma_alloc_coherent().  Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-12 14:00:16 +01:00
Timur Tabi
6e6f66226f powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs.  Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.

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>
2010-08-12 14:00:15 +01:00
Chanwoo Choi
f51582fd8d ASoC: multi-component - Add Goni sound driver
This patch add sound support for the Goni board based on S5PV210.

The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.

The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
  : When TV-OUT cable is inserted on Goni board,
  the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
  : When TV-OUT cable is inserted on Goni board,
  the TV-OUT cable is connected to television.

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>
2010-08-12 14:00:15 +01:00
Chanwoo Choi
3782a52897 ASoC: multi-component - Add Aquila sound driver
This patch add sound support for the Aquila board based on S5PC110.

The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.

The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
  : When TV-OUT cable is inserted on Aquila board,
  the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
  : When TV-OUT cable is inserted on Aquila board,
  the TV-OUT cable is connected to television.

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>
2010-08-12 14:00:14 +01:00
Liam Girdwood
f0fba2ad1b ASoC: multi-component - ASoC Multi-Component Support
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
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>
2010-08-12 14:00:00 +01:00
Peter Ujfalusi
bda7d2a862 ASoC: TWL4030: Capture route runtime DAPM ordering fix
Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.

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>
2010-08-04 00:42:39 +01:00
Axel Lin
116bcd9cf2 ASoC: wm9081: fix resource reclaim in wm9081_register error path
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>
2010-08-03 07:46:41 +01:00
Axel Lin
d484366bee ASoC: wm8978: fix a memory leak if a wm8978_register fail
There is a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.

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>
2010-08-03 07:46:27 +01:00
Axel Lin
4eaac50552 ASoC: wm8974: fix a memory leak if another WM8974 is registered
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).

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>
2010-08-03 07:46:07 +01:00
Axel Lin
6b5d071e8b ASoC: wm8961: fix resource reclaim in wm8961_register error path
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>
2010-08-03 07:45:52 +01:00
Axel Lin
8089a49d99 ASoC: wm8955: fix resource reclaim in wm8955_register error path
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>
2010-08-03 07:45:37 +01:00
Axel Lin
db1e18de98 ASoC: wm8940: fix a memory leak if wm8940_register return error
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.

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>
2010-08-03 07:45:20 +01:00
Axel Lin
62f5ad6733 ASoC: wm8904: fix resource reclaim in wm8904_register error path
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
   snd_soc_register_dai() fail to properly free resources.

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>
2010-08-03 07:45:00 +01:00
Axel Lin
2c2749de11 ASoC: wm8711: fix a memory leak if another WM8711 is registered
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).

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>
2010-08-03 07:44:29 +01:00
Axel Lin
ef99e9b5a1 ASoC: wm8523: fix resource reclaim in wm8523_register error path
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
   handling if snd_soc_register_dai failed.

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>
2010-08-03 07:44:10 +01:00
Axel Lin
085efd28b6 ASoC: da7210: fix a memory leak if failed to initialise da7210 audio codec
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.

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>
2010-08-03 07:43:52 +01:00
Axel Lin
7bcaad919b ASoC: ak4642: fix a memory leak if failed to initialise AK4642
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>
2010-08-03 07:40:41 +01:00
Axel Lin
fd3c8ac9cb ASoC: ad1836: fix a memory leak if another ad1836 is registered
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).

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>
2010-08-03 07:40:26 +01:00
Ian Lartey
992bee401c ASoC: Initial WM8741 CODEC driver
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz.  The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.

TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.

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>
2010-08-03 07:38:15 +01:00
Takashi Iwai
988b0dc154 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-08-02 12:10:52 +02:00
Peter Ujfalusi
998a8a69f3 ASoC: omap-mcbsp: Remove period size constraint in THRESHOLD mode
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.

Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)

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>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
cf80e15860 ASoC: omap-mcbsp: Support for sDMA packet mode
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.

The patch extends the original THRESHOLD mode by doing the
following:

if (period_words <= max_threshold)
Current THRESHOLD mode configuration

Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size

With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.

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>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
15d0143007 ASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_params
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.

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>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
81ec027e64 ASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_params
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.

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>
2010-08-02 10:37:43 +01:00
Kuninori Morimoto
3bc280708e ASoC: fsi: Add new funtion for SPDIF
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>
2010-07-29 10:28:49 -07:00
Kuninori Morimoto
265c770d03 ASoC: fsi: remove device id check
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>
2010-07-29 10:28:37 -07:00
Kuninori Morimoto
bced8f5a36 ASoC: fsi: remove unnecessary clock processing
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>
2010-07-29 10:28:27 -07:00
Peter Ujfalusi
a577b318fc ASoC: tlv320dac33: Add support for automatic FIFO configuration
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.

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>
2010-07-29 10:21:11 +01:00
Peter Ujfalusi
f430a27f05 ASoC: tlv320dac33: Revisit the FIFO Mode1 handling
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>
2010-07-29 10:21:04 +01:00
Peter Ujfalusi
b93cc9f19b ASoC: TWL4030: Capture route DAPM event fix
There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.

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>
2010-07-27 11:43:40 +01:00
Kuninori Morimoto
a7e7cd5bd7 ASoC: da7210: Add HeadPhone Playback Volume control
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>
2010-07-23 10:17:47 +01:00
Takashi Iwai
49e7042799 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-07-21 15:01:07 +02:00
Peter Ujfalusi
01ea6ba2bc ASoC: TWL4030: Add configurable delay after digimic enable
When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.

Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.

Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.

Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).

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>
2010-07-21 11:57:58 +01:00
Mark Brown
d1ce6b200c ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 10:27:05 +01:00
Sekhar Nori
48519f0ae0 ASoC: davinci: let platform data define edma queue numbers
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.

This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.

platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.

Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.

Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.

This patch has been tested on DM644x and OMAP-L138 EVMs.

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>
2010-07-20 09:57:20 +01:00
Chanwoo Choi
5c519767b6 ASoC:Support Samsung SoC(S5P) in I2Sv2
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>
2010-07-20 09:53:36 +01:00
Mark Brown
3b89b22358 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-20 09:52:25 +01:00
Chanwoo Choi
41f9a314af ASoC: Select wm_hubs automatically for WM8994
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>
2010-07-20 09:51:12 +01:00
Mark Brown
a3257ba869 ASoC: Implement WM8994 AIF1ADC2 paths
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 09:48:25 +01:00
Mark Brown
395e4b7362 ASoC: Explicitly disable DC servo on WM hubs headphone powerdown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 09:48:07 +01:00
Eric Bénard
8a0bbbeb58 ASoC: eukrea-tlv320: add support for cpuimx35sd
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>
2010-07-20 09:47:28 +01:00
Kulikov Vasiliy
50e8ce1469 ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

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>
2010-07-17 19:45:56 +01:00
Kulikov Vasiliy
51b6dfb627 ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

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>
2010-07-17 19:45:56 +01:00
Kulikov Vasiliy
55938b106f ASoC: davinci: check kzalloc() result (typo)
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).

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>
2010-07-17 19:45:56 +01:00
Kuninori Morimoto
3c2ef841c0 ASoC: fsi: Add specified ID for soc-audio
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>
2010-07-17 19:45:56 +01:00
Mark Brown
d947837410 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-17 19:45:43 +01:00
Mark Brown
3c0709396d ASoC: Remove duplicate AUX definition from WM8776
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
2010-07-17 19:44:40 +01:00
Jorge Eduardo Candelaria
0fad4ed7b2 ASoC: TWL6040: Correct widget handling for drivers
In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.

Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.

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>
2010-07-17 19:27:18 +01:00
Mark Brown
5164d74d74 ASoC: Handle read failures in codec_reg
When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-14 20:13:09 +01:00
Mark Brown
03b0dc02cf Merge branch 'for-2.6.35' into for-2.6.36 2010-07-14 20:12:57 +01:00