Commit Graph

750 Commits

Author SHA1 Message Date
Linus Torvalds
33081adf8b Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
  ALSA: hda - Disable sticky PCM stream assignment for AD codecs
  ALSA: usb - Creative USB X-Fi volume knob support
  ALSA: ca0106: Use card specific dac id for mute controls.
  ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
  ALSA: ca0106: Create a nice spot for mapping channels to dacs.
  ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
  ALSA: ca0106: Pull out dac powering routine into separate function.
  ALSA: ca0106 - add Sound Blaster 5.1vx info.
  ASoC: tlv320dac33: Use usleep_range for delays
  ALSA: usb-audio: add Novation Launchpad support
  ALSA: hda - Add workarounds for CT-IBG controllers
  ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
  ASoC: tpa6130a2: Error handling for broken chip
  ASoC: max98088: Staticise m98088_eq_band
  ASoC: soc-core: Fix codec->name memory leak
  ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
  ALSA: hda - Add some workarounds for Creative IBG
  ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
  ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
  ALSA: hda - Add alc_init_jacks() call to other codecs
  ...
2010-10-25 08:32:05 -07:00
Takashi Iwai
506ecbca71 Merge branch 'topic/hda' into for-linus 2010-10-25 10:40:05 +02:00
Takashi Iwai
aa5c14d5c0 Merge branch 'topic/asoc' into for-linus
Conflicts:
	arch/powerpc/platforms/85xx/p1022_ds.c
2010-10-25 10:00:30 +02:00
Kay Sievers
39aba963d9 driver core: remove CONFIG_SYSFS_DEPRECATED_V2 but keep it for block devices
This patch removes the old CONFIG_SYSFS_DEPRECATED_V2 config option,
but it keeps the logic around to handle block devices in the old manner
as some people like to run new kernel versions on old (pre 2007/2008)
distros.

Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Cc: Jens Axboe <axboe@kernel.dk>
Cc: Stephen Hemminger <shemminger@vyatta.com>
Cc: "Eric W. Biederman" <ebiederm@xmission.com>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: "James E.J. Bottomley" <James.Bottomley@suse.de>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: Randy Dunlap <randy.dunlap@oracle.com>
Cc: Tejun Heo <tj@kernel.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2010-10-22 10:16:43 -07:00
Mark Brown
e86e1244a4 ASoC: Restore MAX98088 CODEC driver
This reverts commit f6765502f8 and adds
the missing include file.

Signed-off-by: Peter Hsiang <Peter.Hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-10-18 16:46:27 -07:00
Nobuhiro Iwamatsu
83fc3bc095 ALSA: emu10k1: Fix warning: "CCR" redefined
CCR is defined in emu10k1, but SuperH is defined too.
If user use this driver with SuperH, it becomes a double definition.

Signed-off-by: Nobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>
Cc: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-10-18 07:45:44 +02:00
Takashi Iwai
c08d91695b ALSA: tlv - Define numbers in sound/tlv.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-10-17 10:46:14 +02:00
Mike Frysinger
363129ea90 ALSA: fix unused warnings with snd_power_get_state
If we compile the ASoC code with PM disabled, we hit stuff like:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_suspend_check':
sound/soc/soc-dapm.c:440: warning: unused variable 'codec'

So tweak the stub macro to avoid these issues.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-10-17 10:08:45 +02:00
Mika Westerberg
0562f7882d ASoC: don't register AC97 devices twice
With generic AC97 ASoC glue driver (codec/ac97.c), we get following warning when
the device is registered (slightly stripped the backtrace):

kobject (c5a863e8): tried to init an initialized object, something is seriously
                    wrong.
[<c00254fc>] (unwind_backtrace+0x0/0xec)
[<c014fad0>] (kobject_init+0x38/0x70)
[<c0171e94>] (device_initialize+0x20/0x70)
[<c017267c>] (device_register+0xc/0x18)
[<bf20db70>] (snd_soc_instantiate_cards+0x924/0xacc [snd_soc_core])
[<bf20e0d0>] (snd_soc_register_platform+0x16c/0x198 [snd_soc_core])
[<c0175304>] (platform_drv_probe+0x18/0x1c)
[<c0174454>] (driver_probe_device+0xb0/0x16c)
[<c017456c>] (__driver_attach+0x5c/0x7c)
[<c0173cec>] (bus_for_each_dev+0x48/0x78)
[<c0173600>] (bus_add_driver+0x98/0x214)
[<c0174834>] (driver_register+0xa4/0x130)
[<c001f410>] (do_one_initcall+0xd0/0x1a4)
[<c0062ddc>] (sys_init_module+0x12b0/0x1454)

This happens because the generic AC97 glue driver creates its codec->ac97 via
calling snd_ac97_mixer(). snd_ac97_mixer() provides own version of
snd_device.register which handles the device registration when
snd_card_register() is called.

To avoid registering the AC97 device twice, we add a new flag to the
snd_soc_codec: ac97_created which tells whether the AC97 device was created by
SoC subsystem.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-10-13 10:35:17 +01:00
Mark Brown
4c14d78e8a ASoC: Use delayed work for debounce of GPIO based jacks
Rather than block the workqueue by sleeping to do the debounce use delayed
work to implement the debounce time. This should also means that we extend
the debounce time on each new bounce, potentially allowing shorter debounce
times for clean insertions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-10-07 12:58:56 -07:00
Mark Brown
3367b8d427 ASoC: Add support for WM8962 GPIO outputs
The WM8962 features five GPIOs, add support for controlling their output
state via gpiolib.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-10-02 14:23:04 -07:00
Mark Brown
a4f28c001d ASoC: Provide microphone bias configuration for WM8962
Add the widget for MICBIAS power control and allow configuration of the
microphone bias setup via the platform data for the WM8962. When
microphone status signals are brought out to GPIO this should be
sufficient to enable microphone detection.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-09-30 09:36:13 -07:00
Mark Brown
45e655047f ASoC: Initial WM8962 IRQ support
Provide an initial hookup for interrupts on the WM8962. Currently we simply
report error status via log messages if an IRQ is provided for the device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-09-29 00:32:36 -07:00
Mark Brown
831853c87f ALSA: Add more jack button slots
Some devices have more flexible microphone detection and can detect
a wider range of buttons.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-09-07 08:04:38 +02:00
Mark Brown
ea0d09de13 ASoC: Add event variants of the AIF widgets
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-09-06 11:34:29 +01:00
Kuninori Morimoto
7522948b1b ASoC: fsi: modify compile error
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>
2010-08-31 13:28:13 +01:00
Jarkko Nikula
4e48541676 ASoC: Swap bias level enumeration
Swapping the bias level enumeration is only meant to help debugging. It is
easier if number 0 means bias off and bigger number means bigger bias level.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-31 13:06:40 +01:00
Jaroslav Kysela
bd76af0f87 ALSA: pcm midlevel code - add time check for double interrupt acknowledge
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.

This code uses jiffies to check the right time window without any
performance impact.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-08-18 15:18:02 +02:00
Jaroslav Kysela
56385a12d9 ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter)
With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.

It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.

More information: Kernel bugzilla bug#16300

[A copmile warning fixed by tiwai]

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-08-18 15:10:59 +02:00
Mark Brown
e4862f2f6f Merge branch 'for-2.6.36' into for-2.6.37
Fairly simple conflicts, the most serious ones are the i.MX ones which I
suspect now need another rename.

Conflicts:
	arch/arm/mach-mx2/clock_imx27.c
	arch/arm/mach-mx2/devices.c
	arch/arm/mach-omap2/board-rx51-peripherals.c
	arch/arm/mach-omap2/board-zoom2.c
	sound/soc/fsl/mpc5200_dma.c
	sound/soc/fsl/mpc5200_dma.h
	sound/soc/fsl/mpc8610_hpcd.c
	sound/soc/pxa/spitz.c
2010-08-16 18:42:58 +01:00
Sam Ravnborg
60641aa1f3 include: replace unifdef-y with header-y
unifdef-y and header-y has same semantic.
So there is no need to have both.

Drop the unifdef-y variant and sort all lines again

Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
2010-08-14 22:26:51 +02:00
Mark Brown
cf7af01aa7 Merge branch 'topic/multi-component' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.37 2010-08-12 14:40:28 +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
Linus Torvalds
faa38b5e0e Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* '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
  ...
2010-08-07 17:07:31 -07:00
Mark Brown
9a76f1ff6e ASoC: Add initial WM8962 CODEC driver
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>
2010-08-05 13:38:57 +01:00
Takashi Iwai
74bf40f079 Merge branch 'topic/misc' into for-linus 2010-08-05 11:17:04 +02:00
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
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
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
James Bottomley
82f682514a pm_qos: Get rid of the allocation in pm_qos_add_request()
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request().  This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.

Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
2010-07-19 02:00:34 +02:00
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
Kuninori Morimoto
ccad7b44cc ASoC: fsi: Fixup for master mode
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>
2010-07-13 12:35:39 +01:00
Kuninori Morimoto
095687c48b ASoC: fsi: modify format area definition on flags
There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence number

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>
2010-07-13 12:35:35 +01:00
Takashi Iwai
65ee2ba310 Merge branch 'devel' of git://git.alsa-project.org/alsa-kernel into topic/misc 2010-07-05 15:37:27 +02:00
David Dillow
5daeba34d2 ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.

This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.

Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-28 09:42:09 +02:00
Vladimir Zapolskiy
cc3202f5da ASoC: uda134x: replace a macro with a value in platform struct.
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>
2010-06-25 12:29:01 +01:00
apatard@mandriva.com
ea762b047e ASoC: Add SND_SOC_DAPM_PRE_POST_PMD event
Some systems codecs need to configure some registers before and after
powering down some of their part. As a convenience add a macro for that.

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>
2010-05-31 12:20:01 +01:00
Mark Brown
e37c83c06c Merge commit 'v2.6.35-rc1' into for-2.6.36 2010-05-31 11:07:15 +01:00
Ben Collins
15c0cee6c8 ALSA: pcm: Define G723 3-bit and 5-bit formats
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.

I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.

Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 09:10:03 +02:00
Linus Torvalds
7f06a8b26a Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
  ALSA: hda: Storage class should be before const qualifier
  ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
  ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
  ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
  ASoC: TWL6040: Enable earphone path in codec
  ASoC: SDP4430: Add support for Earphone speaker
  ASoC: SDP4430: Add sdp4430 machine driver
  ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
  ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
  ALSA: sound/pci/asihpi: Use kzalloc
  ALSA: hdmi - dont fail on extra nodes
  ALSA: intelhdmi - add id for the CougarPoint chipset
  ALSA: intelhdmi - user friendly codec name
  ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
  ALSA: asihpi: incorrect range check
  ALSA: asihpi: testing the wrong variable
  ALSA: es1688: add pedantic range checks
  ARM: McBSP: Add support for omap4 in McBSP driver
  ARM: McBSP: Fix request for irq in OMAP4
  OMAP: McBSP: Add 32-bit mode support
  ...
2010-05-20 09:41:44 -07:00
Takashi Iwai
d71f4cece4 Merge branch 'topic/asoc' into for-linus
Conflicts:
	sound/soc/codecs/ad1938.c
2010-05-20 12:00:43 +02:00
Takashi Iwai
20406f9b67 Merge branch 'topic/jack' into for-linus 2010-05-20 11:59:37 +02:00
Takashi Iwai
5e8aa85253 Merge branch 'topic/misc' into for-linus 2010-05-20 11:59:29 +02:00
apatard@mandriva.com
b6f4bb383d ASoC: Add SOC_DOUBLE_R_SX_TLV control
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0

Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-16 18:04:46 +01:00
Daniel Mack
89485d4931 ALSA: include/sound/asound.h whitespace fixups
This fixes some whitespace/indentation flaws I stumbled over.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-11 22:41:50 +02:00
Peter Ujfalusi
d11bb4a925 ASoC: core: Fix for the volume limiting when invert is in use
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-11 09:34:11 +01:00
Mark Gross
ed77134bfc PM QOS update
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation.  I did this because request more
accurately represents what it actually does.

Also, I added a string based ABI for users wanting to use a string
interface.  So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface.  (someone asked me for it and I don't think
it hurts anything.)

This patch updates some documentation input I got from Randy.

Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
2010-05-10 23:08:19 +02:00
Mark Brown
3efab7dcc0 ASoC: Allow DAI links to be kept active over suspend
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked.  This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link.  It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:37:13 +01:00
Mark Brown
1547aba993 ASoC: Support leaving paths enabled over system suspend
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.

Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.

When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 10:36:48 +01:00