Commit Graph

311740 Commits

Author SHA1 Message Date
Takashi Iwai 1558eb838f Merge branch 'topic/pm-convert' into for-next
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
2012-07-19 08:21:57 +02:00
Takashi Iwai 4609ed6b1f ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
 terms, mostly just a few new drivers and updates:
 
 - Added the ability to add and remove DAPM paths dynamically, mostly for
   reparenting on clock changes.
 - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
   reference platform and ttc-dkp.
 - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
   Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
 - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
   Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for 3.6

This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:

- Added the ability to add and remove DAPM paths dynamically, mostly for
  reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
  reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
  Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
  Isabelle and Wolfson Microelectronics WM5102 and WM5110
2012-07-19 08:03:20 +02:00
Torben Hohn 409b78cc17 ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
The MX31 Audmux also has 7 Ports.
This patch adds the missing define, and makes the debugfs code iterate
over that port too.

Signed-off-by: Torben Hohn <torbenh@linutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-18 22:09:40 +01:00
Mark Brown 31a2239a5a ASoC: littlemill: Add userspace control of the WM1250 I/O
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-18 22:09:23 +01:00
Mark Brown b4046d013b ASoC: wm8994: Update micdet for irqdomain conversion
The conversion of the core driver to irqdomains means that we don't need
and irq_base to have working interrupts so use wm8994_request_irq() to
deal with looking up the interrupt number for the micdet IRQ.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-18 22:09:23 +01:00
David Henningsson 639aa4bd58 ALSA: hda - make sure alc268 does not OOPS on codec parse
A recent commit made patch_alc268 call snd_hda_pick_fixup with
NULL quirk pointer. Make sure we do not reference that NULL pointer.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-18 18:07:03 +02:00
Takashi Iwai f0913cd16e Merge branch 'topic/misc' into for-next
Generic updates for sound 3.6
2012-07-18 13:53:29 +02:00
Takashi Iwai 61eab000f3 Merge branch 'topic/hda' into for-next
This is a large chunk merge for 3.6 HD-audio things
2012-07-18 13:53:06 +02:00
David Henningsson 4e01ec636e ALSA: hda - Add support for Realtek ALC282
This codec has a separate dmic path (separate dmic only ADC),
and thus it looks mostly like ALC275.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1025377
Tested-by: Ray Chen <ray.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-18 07:51:26 +02:00
Takashi Iwai f46c329644 ALSA: hda - Fix index number conflicts of phantom jacks
Since some jack controls may be renamed as phantom jacks, the existing
check for index conflicts doesn't work because it simply compares the
name with the last used name, assuming that the controls with the same
name continue.  Thus, it would result in the duplicated controls when
two or more phantom jacks with the very same type exist, and the
driver gives up with an error.

This patch fixes the problem by checking the index number conflicts
more intensively (but dumbly).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-17 16:55:56 +02:00
Takashi Iwai 59b1f084ab ALSA: opti9xx: Fix section mismatch by PM support
In the previous commit, snd_opti9xx_configure() is called from the
resume handler but it's still marked as __devinit.  Fix it.

Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-17 12:15:49 +02:00
Ondrej Zary 5dd250728a ALSA: snd-opti9xx: Implement suspend/resume
Implement suspend/resume support for Opti 92x and 93x chips.
Tested with Opti 929A+AD1848 and Opti 931.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-17 10:55:47 +02:00
Aaron Plattner 7ae48b56f8 ALSA: hda - Add new GPU codec ID to snd-hda
Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Andy Ritger <aritger@nvidia.com>
Reviewed-by: Daniel Dadap <ddadap@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-17 07:30:44 +02:00
Takashi Iwai bdbe34dece ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH
According to Xingchao,
  This works for HDMI audio, otherwise there's blocking issue.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 16:17:10 +02:00
Wang Xingchao 1c76684d27 ALSA: hda - add Haswell HDMI codec id
0x80862807 is HDMI id for Haswell HDA.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 16:03:11 +02:00
Wang Xingchao e926f2c850 ALSA: hda - Add DeviceID for Haswell HDA
this patch add proper id for Haswell HDA Controller.

[Added AZX_DCAPS_POSFIX_COMBO flag by tiwai]

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 16:01:57 +02:00
Ondrej Zary b4e2a16f99 ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx
Yamaha OPL3-SAx chips don't resume properly when playback is running -
garbage is played after resume. Restoring the CS4231_PLAYBK_FORMAT register
last fixes the problem.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 11:24:05 +02:00
Ondrej Zary ca3273fb59 ALSA: wss_lib: fix suspend/resume
By setting SNDRV_PCM_INFO_RESUME, wss_lib claims that it can restore the card
state fully on resume. But in fact, it can't as DMA is not restored so any
playback/capture running during suspend will fail to continue after resume.

Remove SNDRV_PCM_INFO_RESUME flag from pcm info field to fix the problem.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 11:23:45 +02:00
Clemens Ladisch 093eef4166 ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE
Instead of the hard-to-mantain TLV_DB_RANGE_HEAD macro, use
DECLARE_TLV_DB_RANGE, which computes its size automatically.

(Also make this data const on the way.)

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 10:02:19 +02:00
Clemens Ladisch bf1d1c9b61 ALSA: tlv: add DECLARE_TLV_DB_RANGE()
Add a DECLARE_TLV_DB_RANGE() macro so that dB range information
can be specified without having to count the items manually for
TLV_DB_RANGE_HEAD().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 10:02:18 +02:00
Clemens Ladisch 570aef5de1 ALSA: tlv: add DECLARE_TLV_CONTAINER()
Add the DECLARE_TLV_CONTAINER() macro to allow having static
TLVs containing more than one item.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 10:02:16 +02:00
Clemens Ladisch b5b9eb5467 ALSA: tlv: compute TLV_*_ITEM lengths automatically
Add helper macros with a little bit of preprocessor magic to
automatically compute the length of a TLV item.  This lets us avoid
having to compute this by hand, and will allow to use items that do
not use a fixed length.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-16 10:02:10 +02:00
Sylwester Nawrocki 601787c232 ASoC: Convert S3C2412 I2S driver to gpiolib API
The s3c2410_gpio* calls are obsolete and have been scheduled for
removal since several kernel releases. Remove them and use common
gpiolib API.
This patch is a prerequisite for removal of the obsolete S3C24XX
SoC GPIO definitions.

Compile tested only.

Cc: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-15 22:21:42 +01:00
Sylwester Nawrocki 0eed8a1869 ASoC: Convert S3C24XX I2S driver to gpiolib API
The s3c2410_gpio* calls are obsolete and have been scheduled for
removal since several kernel releases. Remove them and use common
gpiolib API.
This patch is a prerequisite for removal of the obsolete S3C24XX
SoC GPIO definitions.

Tested on Micro2440-SDK.

Cc: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-15 22:21:42 +01:00
Marek Belisko 98b3cf1290 ASoC: dapm: Fix compilation warning
Fix following:
sound/soc/soc-dapm.c: In function ‘dapm_clock_event’:
sound/soc/soc-dapm.c:1021:1: warning: control reaches end of non-void function [-Wreturn-type]

Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-13 15:43:41 +01:00
Daniel Mack 68e67f40b7 ALSA: snd-usb: move calls to usb_set_interface
The rework of the snd-usb endpoint logic moved the calls to
snd_usb_set_interface() into the snd_usb_endpoint implemenation. This
changed the order in which these calls are issued to the device, and
thereby caused regressions for some webcams.

Fix this by moving the calls back to pcm.c for now to make it work again
and use snd_usb_endpoint_activate() to really tear down all remaining
URBs in the flight, consequently fixing another regression caused by USB
packets on the wire after altsetting 0 has been selected.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net>
Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-13 09:31:42 +02:00
Mark Brown e4dd76788c ASoC: wm8962: Redo early init of the part on resume
Ensure robust startup of the part by going through the reset procedure
prior to resyncing the full register cache, avoiding potential intermittent
faults in some designs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-07-12 18:41:02 +01:00
Matthias Kaehlcke b761c0ca2e ASoC: Free memory in the error paths of soc_of_parse_audio_routing()
Release the memory of the routing table before leaving the function upon errors
in the device tree

Signed-off-by: Matthias Kaehlcke <matthias@kaehlcke.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-11 18:40:57 +01:00
Mark Brown 5c6af635fd ASoC: wm5110: Add audio CODEC driver
The WM5110 is a highly integrated low power audio subsystem for
smartphones, tablets and other portable audio devices. It combines
an advanced DSP feature set with a flexible, high performance audio
hub CODEC.

This patch adds the audio CODEC driver for the device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-11 11:00:26 +01:00
Dan Carpenter f96985e3b3 ASoC: STA529: fix an error message
GCC complains that "ret" is uninitialized here.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-By: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-11 10:37:15 +01:00
Mark Brown 2b4d39fc2a ASoC: arizona: Support variable FLL VCO multipliers
Some Arizona chips have a higher frequency for the FLL VCO, support this
in the common code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-10 17:17:35 +01:00
Jiri Prchal a1f34af0ec ASoC: tlv320aic3x: add input clock selection
This patch adds input selection of main codec clock - from what pin.
Both registers set same value since codec uses clock divider or pll at one time.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-10 14:41:48 +01:00
Jiri Prchal bb1daa803c ASoC: tlv320aic3x: add AGC settings
This patch adds AGC target level and times settings for TLV320AIC3x.
Enums uses small arrays of two channels left and right since it uses different registers.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-10 14:41:48 +01:00
Jiri Prchal 774441915d ASoC: tlv320aic3x: add deemphasis switch
This patch adds missing deemphasis switch.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-09 19:36:13 +01:00
Mark Brown bc9dce5853 ASoC: wm5102: Fix cut'n'paste for digital volume registers
The analogue PGA shifts were used; this makes no practical difference as
the values are the same.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-09 19:35:09 +01:00
Mark Brown c9c56fd0b7 ASoC: arizona: Add IN4 to the mixer tables
Some devices have four input structures rather than three.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-09 19:35:04 +01:00
Mark Brown a837987e7b ASoC: arizona: Export dai_ops
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-09 12:16:41 +01:00
Mark Brown 410837a7a2 ASoC: arizona: Change DAPM routes for AIF clocks when we change them
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-07-06 19:07:00 +01:00
Mark Brown efcc3c61b9 ASoC: dapm: Allow routes to be deleted at runtime
Since we're now relying on DAPM for things like enabling clocks when we
reparent the clocks for widgets we need to either use conditional routes
(which are expensive) or remove routes at runtime. Add a route removal
API to support this use case.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-07-06 19:06:59 +01:00
Mark Brown fabd03842b ASoC: dapm: Mark widgets as dirty when a route is added
If we add a new route at runtime then we'll need to recheck the connections
to the affected widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-07-06 19:06:59 +01:00
Liam Girdwood 3ac3f5ca91 ASoC: dpcm: Allow FE to be opened without valid BE routes.
Some userspace will open a PCM device and then configure mixers
for routing before triggering. This patch allows userspace to do
this sequence.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-06 19:06:58 +01:00
Simon Wilson 3e4536546b ASoC: twl6040: fix spelling mistake
Fix spelling mistake in "High-Performance" option of twl6040 power mode.

Signed-off-by: Simon Wilson <simonwilson@google.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-06 19:06:50 +01:00
Liam Girdwood 5cb9b74822 ASoC: pcm: Clean up logging in soc_new_pcm()
Use dev_ style logging throughout soc_new_pcm()

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-06 18:59:15 +01:00
Takashi Iwai 9e9b594661 ALSA: usb-audio: Fix the first PCM interface assignment
In the new PCM streaming logic, the interface number is assigned to
usb stream instance (subs->interface) after the format and rate setups
are succeeded, but some codes are still passing subs->interface as the
reference to helper functions.  This leads to initializing with an
invalid iface number (-1).

This patch replaces the wrong references with the ones from the target
fmt correctly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-06 08:11:43 +02:00
Mark Brown bcbf4a69ee ASoC: wm1250-ev1: Flag all supported rates in the DAI
Not previously noticed due to normal usage being with CODEC<->CODEC links.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-05 20:49:37 +01:00
Mark Brown 9498822d75 ASoC: wm5102: Allow routing through the ASRCs
This enables common telephony use cases.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-05 17:44:19 +01:00
Mark Brown c013b27a17 ASoC: arizona: Enable ASYNCCLK domain for audio interfaces
If an audio interface is configured to use ASYNCCLK then update the
asynchronous sample rate rather than one of our primary sample rates.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-05 17:44:17 +01:00
Fabio Estevam 42810d1622 ASoC: imx-mc13783: Add audmux settings for mx27pdk
mx27pdk board also has a mc13783 codec.

Add support for it and do a run-time machine type check to perform the correct
audiomux settings.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-05 13:48:25 +01:00
Hui Wang ef3207c503 ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000
If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR
register (means Tx clock pin is input), we don't need to set
IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and
i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL
settings are ignored".

Signed-off-by: Hui Wang <jason77.wang@gmail.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-07-05 13:47:11 +01:00
Mark Brown 1464189f8c ALSA: pcm: Make constraints lists const
They aren't modified by the core so the drivers can declare them const.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-05 14:19:39 +02:00