Commit Graph

841 Commits

Author SHA1 Message Date
Mark Brown 8a9dab1a55 ASoC: Update name of debugfs root symbol to snd_soc_
Everything else is using snd_soc_ so we should use it here too.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-10 22:25:21 +00:00
Stephen Warren faff4bb067 ASoC: Export debugfs root dentry
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.

Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-10 22:20:04 +00:00
Clemens Ladisch 9600732b6c ALSA: core, oxygen, virtuoso: add an enum control info helper
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10 16:46:53 +01:00
Dimitris Papastamos 24ff33ac69 ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
This new type is a virtual version of snd_soc_dapm_mux.  It is used
when a backing register value is not necessary for deciding which
input path to connect.  A simple virtual enumeration control e.g.
SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used
to choose which path to connect.

The snd_soc_dapm_virt_mux type ensures that during the initial
path setup, the first (which is also the default) input path will
be connected.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-17 17:36:28 +00:00
Mark Brown 97404f2e03 ASoC: Do DAPM control updates in the middle of DAPM sequences
Attempt to minimise audible effects from mixer and mux updates by
implementing the actual register changes between powering down widgets
that have become unused and powering up widgets that are newly used.

This means that we're making the change with the minimum set of widgets
powered, that the input path is connected when we're powering up widgets
(so things like DC offset correction can run with their signal active)
and that we bring things down to cold before switching away.  Since
hardware tends to be designed for the power on/off case more than for
dynamic reconfiguration this should minimise pops and clicks during
reconfiguration while active.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-17 11:18:04 +00:00
Jarkko Nikula 7be31be880 ASoC: Extend DAPM to handle power changes on cross-device paths
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.

This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.

DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.

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>
2010-12-15 18:01:34 +00:00
Jarkko Nikula 97c866defc ASoC: Move widgets from DAPM context to snd_soc_card
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.

This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.

Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.

An example below shows a path that connects MONO out of A into Line In of B:

static const struct snd_soc_dapm_route mapA[] = {
	{"MONO", NULL, "DAC"},
};

static const struct snd_soc_dapm_route mapB[] = {
	{"Line In", NULL, "MONO"},
};

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>
2010-12-15 18:01:01 +00:00
Jarkko Nikula 8ddab3f510 ASoC: Move DAPM paths from DAPM context to snd_soc_card
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.

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>
2010-12-15 18:00:41 +00:00
Mark Brown 656d4b1ede ASoC: Remove unused DAPM_DOUBLE control types
There are no users of these and it's not clear what they would do given
the mono flow modelling which DAPM does. If need arises we can add them
again.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-15 14:18:07 +00:00
Olaya, Margarita d88429a695 ASoC: dapm: Add output driver widget
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.

Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.

Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-14 11:12:11 +00:00
Takashi Iwai 354d14b3f5 Merge branch 'topic/workq-update' into topic/misc 2010-12-13 09:29:52 +01:00
Takashi Iwai 20aeeb356b Merge branch 'topic/workq-update' into topic/asoc
Conflicts:
	sound/soc/codecs/wm8350.c
	sound/soc/codecs/wm8753.c
	sound/soc/sh/fsi.c
	sound/soc/soc-core.c
2010-12-13 09:28:43 +01:00
Seungwhan Youn 05d209ad3b ASoC: Remove unnecessary structure definitions
This patch removes some legacy structure definitions which are not using
in current ASoC drivers.

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>
2010-12-09 11:27:57 +00:00
Dimitris Papastamos 0d735eaa2c ASoC: soc-cache: Add optional cache name member to snd_soc_cache_ops
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.

Remove redundant newline in source code.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-06 14:13:46 +00:00
Mark Brown 1badabd980 ASoC: Add post-CODEC bias level callback for machine driver
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-06 12:41:30 +00:00
Mark Brown 001ae4c035 ASoC: Constify struct snd_soc_codec_driver
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-03 16:37:55 +00:00
Dimitris Papastamos fdf0f54dab ASoC: soc-core: Allow machine drivers to override compress_type
This patch allows machine drivers to override the compression type
provided by the codec driver.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-03 16:37:32 +00:00
Dimitris Papastamos 3335ddca93 ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary.  This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-03 16:37:06 +00:00
Dimitris Papastamos ff819b8357 ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf
The snd_soc_codec_conf struct now holds codec specific configuration
information.

A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.

In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead.  In all other cases the one
set in the snd_soc_codec_conf struct takes effect.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-03 16:36:45 +00:00
Dimitris Papastamos 119bd789f6 ASoC: Change the base value of compress_type
Ensure that the base value of compress_type starts at 1 so that
we know whether the machine driver has provided a compress_type
for overriding the codec supplied one.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-03 16:36:23 +00:00
Dimitris Papastamos 23bbce34f4 ASoC: Add compress_type as a member to snd_soc_codec
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with.  The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.

Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-03 16:36:03 +00:00
Mark Brown c3acec2671 ASoC: Move active copy of CODEC read and write into runtime structure
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-03 12:18:17 +00:00
Mark Brown 1ee46ebd04 ASoC: Make the DAI ops constant in the DAI structure
Neither drivers nor the core should be fiddling with the actual ops
structure at runtime.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-03 12:17:43 +00:00
Florian Faber 28b26e1553 ALSA: hdsp - Add support for RPM io box
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.

Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-01 12:14:47 +01:00
Jarkko Nikula 2eea392d0a ASoC: Add support for optional auxiliary dailess codecs
This makes possible to register auxiliary dailess codecs in a machine
driver. Term dailess is used here for amplifiers and codecs without DAI or
DAI being unused.

Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs
are probed after initializing the DAI links. There are no major differences
between DAI link codecs and dailess codecs in ASoC core point of view. DAPM
handles them equally and sysfs and debugfs directories for dailess codecs
are similar except the pmdown_time node is not created.

Only suspend and resume functions are modified to traverse all probed codecs
instead of DAI link codecs.

Example below shows a dailess codec registration.

struct snd_soc_aux_dev foo_aux_dev[] = {
	{
		.name = "Amp",
		.codec_name = "codec.2",
		.init = foo_init2,
	},
};

static struct snd_soc_card card = {
	...
	.aux_dev = foo_aux_dev,
	.num_aux_devs = ARRAY_SIZE(foo_aux_dev),
};

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-30 14:39:00 +00:00
Dimitris Papastamos df0701bb86 ASoC: soc-cache: Ensure consistent cache naming
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-29 12:43:52 +00:00
Kuninori Morimoto d4bc99b977 ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMI
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
2010-11-24 15:29:56 +09:00
Kay Sievers 03cfe6f57d ALSA: support module on-demand loading for seq and timer
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and
/dev/snd/timer the usual static minors, and export specific
module aliases to generate udev module on-demand loading
instructions:

  $ cat /lib/modules/2.6.33.4-smp/modules.devname
  # Device nodes to trigger on-demand module loading.
  microcode cpu/microcode c10:184
  fuse fuse c10:229
  ppp_generic ppp c108:0
  tun net/tun c10:200
  uinput uinput c10:223
  dm_mod mapper/control c10:236
  snd_timer snd/timer c116:33
  snd_seq snd/seq c116:1

The last two lines instruct udev to create device nodes, even
when the modules are not loaded at that time.

As soon as userspace accesses any of these nodes, the in-kernel
module-loader will load the module, and the device can be used.

The header file minor calculation needed to be simplified to
make __stringify() (supports only two indirections) in
the MODULE_ALIAS macro work.

This is part of systemd's effort to get rid of unconditional
module load instructions and needless init scripts.

Cc: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-24 05:53:25 +01:00
Jarkko Nikula 851cad5aa1 ASoC: Remove cyclic dependency between soc.h and soc-dapm.h/soc-dai.h
There is no need anymore to include soc.h in soc-dapm.h and soc-dai.h as
drivers are converted to include only soc.h.

Thanks to Lars-Peter Clausen <lars@metafoo.de> for pointing out the issue.

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>
2010-11-22 14:05:05 +00:00
Clemens Ladisch ab69a4904b ALSA: pcm: support for period wakeup disabling
This patch allows to disable period interrupts which are
not needed when the application relies on a system timer
to wake-up and refill the ring buffer. The behavior of
the driver is left unchanged, and interrupts are only
disabled if the application requests this configuration.
The behavior in case of underruns is slightly different,
instead of being detected during the period interrupts the
underruns are detected when the application calls
snd_pcm_update_avail, which in turns forces a refresh of the
hw pointer and shows the buffer is empty.

More specifically this patch makes a lot of sense when
PulseAudio relies on timer-based scheduling to access audio
devices such as HDAudio or Intel SST. Disabling interrupts
removes two unwanted wake-ups due to period elapsed events
in low-power playback modes. It also simplifies PulseAudio
voice modules used for speech calls.

To quote Lennart "This patch looks very interesting and
desirable. This is something have long been waiting for."

Support for this in hardware drivers is optional.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-11-22 08:13:16 +01:00
Jarkko Nikula ead9b9199c ASoC: Add optional name_prefix for codec kcontrol, widget and route names
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.

This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.

Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.

static const struct snd_soc_dapm_route map0[] = {
	{"Spk", NULL, "MONO"},
};

static const struct snd_soc_dapm_route map1[] = {
	{"Vibra", NULL, "foo MONO"},
};

static struct snd_soc_prefix_map codec_prefix[] = {
	{
		.dev_name = "codec.2",
		.name_prefix = "foo",
	},
};

static struct snd_soc_card card = {
	...
	.prefix_map = codec_prefix,
	.num_prefixes = ARRAY_SIZE(codec_prefix),
};

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>
2010-11-15 15:24:58 +00:00
Dimitris Papastamos a7f387d5af ASoC: soc-cache: Add support for rbtree based register caching
This patch adds support for rbtree compression when storing the
register cache.  It does this by not adding any uninitialized registers
(those whose value is 0).  If any of those registers is written
with a nonzero value they get added into the rbtree.

Consider a sample device with a large sparse register map.  The
register indices are between [0, 0x31ff].  An array of 12800 registers
is thus created each of which is 2 bytes.  This results in a 25kB
region.  This array normally lives outside soc-core, normally in the
driver itself.  The original soc-core code would kmemdup this region
resulting in 50kB total memory.  When using the rbtree compression
technique and __devinitconst on the original array the figures are
as follows.  For this typical device, you might have 100 initialized
registers, that is registers that are nonzero by default.  We build
an rbtree with 100 nodes, each of which is 24 bytes.  This results
in ~2kB of memory.  Assuming that the target arch can freeup the
memory used by the initial __devinitconst array, we end up using
about ~2kB bytes of actual memory.  The memory footprint will increase
as uninitialized registers get written and thus new nodes created in
the rbtree.  In practice, most of those registers are never changed.
If the target arch can't freeup the __devinitconst array, we end up
using a total of ~27kB.  The difference between the rbtree and the LZO
caching techniques, is that if using the LZO technique the size of
the cache will increase slower as more uninitialized registers get
changed.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-11 15:59:22 +00:00
Dimitris Papastamos cc28fb8e7d ASoC: soc-cache: Add support for LZO register caching
This patch adds support for LZO compression when storing the register
cache.  The initial register defaults cache is marked as __devinitconst
and the only change required for a driver to use LZO compression is
to set the compress_type member in codec->driver to SND_SOC_LZO_COMPRESSION.

For a typical device whose register map would normally occupy 25kB or 50kB
by using the LZO compression technique, one can get down to ~5-7kB.  There
might be a performance penalty associated with each individual read/write
due to decompressing/compressing the underlying cache, however that should not
be noticeable.  These memory benefits depend on whether the target architecture
can get rid of the memory occupied by the original register defaults cache
which is marked as __devinitconst.  Nevertheless there will be some memory
gain even if the target architecture can't get rid of the original register
map, this should be around ~30-32kB instead of 50kB.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-11 15:59:01 +00:00
Dimitris Papastamos 7a30a3db34 ASoC: soc-cache: Add support for flat register caching
This patch introduces the new caching API and migrates the
old caching interface into the new one.  The flat register caching
technique does not use compression at all and it is equivalent to
the old caching technique.  One can still access codec->reg_cache
directly but this is not advised as that will not be portable
across different caching strategies.

None of the existing drivers need to be changed to adapt to this
caching technique.  There should be no noticeable overhead associated
with using the new caching API.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-11 15:58:41 +00:00
Jarkko Nikula 3a45b8672d ASoC: Move pop time from DAPM context to sound card
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.

debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:35 -04:00
Jarkko Nikula a605215494 ASoC: Add sound card directory under debugfs/asoc/
There will be need to have sound card specific debugfs entries. This patch
introduces a new debugfs/asoc/{card->name}/ directory but does not add yet
any entries there.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:35 -04:00
Liam Girdwood ce6120cca2 ASoC: Decouple DAPM from CODECs
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:29 -04:00
Mark Brown c375370799 ASoC: Push snd_soc_write() and snd_soc_read() into the source file
Facilitating adding trace type stuff. For a first pass add some dev_dbg()
statements into them.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-11-03 13:41:53 -04:00
Mark Brown 9e3be1edbe Merge branch 'for-2.6.37' into HEAD
WARN() fix from Joe moved.

Conflicts:
	sound/soc/codecs/wm_hubs.c
2010-11-02 09:58:49 -04:00
Mark Brown 6e1bd1ab1d Merge branch 'for-2.6.37' into for-2.6.38 2010-11-01 13:58:18 -04:00
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
Arnaud Patard (Rtp) 6f4bc952c6 ASoC: add support for alc562[123] codecs
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-10-21 13:51:13 -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
Mark Brown 50ae8384cd ASoC: Remove unused DAPM suspend flag
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.

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 10:35:55 +01:00
Krzysztof Helt a20971b201 ALSA: Merge es1688 and es968 drivers
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.

Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.

Also, a new PnP id is added for the card I acquired (the change
was tested on this card).

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 09:49:30 +02:00
Krzysztof Helt 396fa82726 ALSA: es1688: allocate snd_es1688 structure as a part of snd_card structure
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-10 09:48:59 +02:00
Peter Ujfalusi 826e962c46 Revert "ASoC: tpa6130a2: Support for limiting gain"
This reverts commit 6f3991152f.

Since core has now support for limiting the volume on controls this
patch is not needed.  Furthermore, this patch actually prevents the core
to set new volume on the TPA.

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>
2010-05-07 16:42:23 +01:00
Peter Ujfalusi 637d3847ba ASoC: core: Support for limiting the volume
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)

If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:

snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);

This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.

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>
2010-05-07 16:41:33 +01:00
Takashi Iwai aeb29a82de Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-05-06 17:06:27 +02:00
Peter Ujfalusi 6f3991152f ASoC: tpa6130a2: Support for limiting gain
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.

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>
2010-05-06 14:58:20 +01:00
Jarkko Nikula 5193d62f18 ASoC: tlv320aic3x: Add platform data and reset gpio handling
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-06 14:58:02 +01:00
Mark Brown 39b8eab7e7 ASoC: Add WM9090 amplifier driver
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.

Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control.  The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-30 16:12:44 +01:00