For CODEC to CODEC links we should only immediately power down if both CODECs
are configured to ignore the power down delay. Factor the logic for this
into a helper function that can be used for both compressed and normal PCMs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We now no longer have any users of hw_read() in the kernel so remove the
code in order to prevent any new users being added. Users should be using
regmap.
Signed-off-by: Mark Brown <broonie@linaro.org>
With the upcoming consolidation of normal MUXs and virtual MUXs we need to be
able to distinguish between enums with and without a backing register at the
enum level. Use the same approach as used for virtual mixer controls by setting
the reg field of the enum to SND_SOC_NOPM for enums without a backing register.
This patch adds a set of helper macros that can be used to define such enums.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The implementations for enum and value enum controls are almost identical. The
only difference is that the value enum uses an additional look-up table to map
the control value to the register value, while the enum control uses a direct
mapping. Enums and value enums can easily be distinguished at runtime, for value
enums the values field of the snd_soc_enum struct contains the look-up table,
while for enums it is NULL. This patch adds two new small helper functions
called snd_soc_enum_item_to_val() and snd_soc_enum_val_to_item() which map
between register value and control item. If the items field of the snd_soc_enum
struct is NULL the function will do a direct mapping otherwise they'll use the
look-up table to do the mapping. Using these small helper functions it is
possible to use the same kcontrol handlers for both enums and value enums. The
functions are added a inline functions in soc.h so they can also be used by the
DAPM code to accomplish similar consolidation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This
patch adds the slot parsing from DT supports.
TDM slot properties:
dai-tdm-slot-num : Number of slots in use.
dai-tdm-slot-width : Width in bits for each slot.
For instance:
dai-tdm-slot-num = <2>;
dai-tdm-slot-width = <8>;
And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
to specify a explicit mapping of the channels and the slots. If it's absent
the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
tx and rx masks.
For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
for an active slot as default, and the default active bits are at the LSB of
the masks.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that all users have been converted to regmap we can eliminate the ASoC
level wrapper for I2C I/O reducing the amount of duplicated functionality.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acquiring the DAPM mutex is necessary before using several DAPM
functions and dereference is quite ugly. This patch provides a helper
function to simplify this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since these macros are supposed to be used for decalring const
objects, let's add the const modifier there.
The doubled const appearing in usages will be cleaned by later
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The name "max" in struct soc_enum is rather confusing since it
actually takes the number of items. With "max", one might try to
assign (nitems - 1) value.
Rename the field to a more appropriate one, "items", which is also
used in struct snd_ctl_elem_info, too.
This patch also rewrites some code like "if (x > e->nitems - 1)" with
"if (x >= e->nitems)". Not only the latter improves the readability,
it also fixes a potential bug when e->items is zero.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
All ASoC CODEC drivers that use SPI have now been converted to use regmap
so we can delete SND_SOC_SPI, preventing any new users being added.
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds snd_soc_of_parse_audio_simple_widgets() and supports
below style of widgets name on DT:
"template-wname", "user supplied wname"
For instance:
simple-audio-widgets =
"Microphone", "Microphone Jack",
"Line", "Line In Jack",
"Line", "Line Out Jack",
"Headphone", "Headphone Jack",
"Speaker", "Speaker External";
The "template-wname" currently includes: "Microphone", "Line", "Headphone"
and "Speaker".
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a macro for signed mixer with two registers and tlv array.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some codecs use signed volume control representation with non standard
register sizes, e.g. 6 or 7 bit signed integers.
This patch adds generic signed register volume control logic to
soc-core. Instead of a fixed width signed register control, this
implementation uses a 'min' value and the signed bit location to translate
it to an absolute volume. Using the 'sign_bit' we can calculate a
correct mask for the register values and translate it back into signed
integers of standard size.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently compressed audio streams are statically routed from the /dev
to the DAI link. Some DSPs can route compressed data to multiple BE DAIs
like they do for PCM data.
Add support to allow dynamically routed compressed streams using the existing
DPCM infrastructure. This patch adds special FE versions of the compressed ops
that work out the runtime routing.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB
explicitly with ifdef, and this causes build errors on some drivers
such as:
sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type
Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.
In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.
Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.
Thus, this patch applys symmetry for channels and sample bits.
Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name()
callback on each component drivers.
But required behavior on almost all these drivers is
just returns its indexed driver's name.
This patch adds this feature as default behavior.
.of_xlate_dai_name() can overwrite it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for virtual DAPM mixer controls. They are similar to
virtual DAPM enums. There is no hardware register backing the control, so
changing the control's value wont have any direct effect on the hardware. But it
still influences the DAPM graph by causing the path it sits on to be connected
or disconnected. This in turn can cause power changes for some of the widgets on
the DAPM graph, which will then modify the hardware state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC sound driver requires CPU/CODEC drivers for probing,
and each CPU/CODEC has some DAI on it.
Then, "dai name matching" have been used to identify
CPU-CODEC DAI pair on ASoC.
But, the "dai port number matching" is now required from DeviceTree.
The solution of this issue is to replace
the dai port number into dai name.
Now, CPU/CODEC are based on struct snd_soc_component,
and it can care above as common issue.
This patch adds .of_xlate_dai_name callback interface
on struct snd_soc_component_driver,
and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name.
Then, #sound-dai-cells which enables DAI specifier is required
on CPU/CODEC device tree properties.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Codec includes component by this patch,
and component moved to upside of codec
to avoid extra declaration.
Codec dai will be registered via component
by this patch.
Current component register function
is used for cpu, and it is using
dai/dais functions properly to keep
existing cpu dai name.
And now, it will be used from codec also.
But codec driver had been used dais function only
even though it was single dai.
This patch adds new flag which can selects
dai/dais function on component register
function to keep existing codec dai name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify error handling and remove repetitive (and rarely executed) code
for unregistration by providing a devm_snd_soc_register() card.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Since with the wider use of devres many drivers are now only calling
snd_soc_unregister_component() in their remove functions providing a
managed version will save a reasonable amount of code.
Signed-off-by: Mark Brown <broonie@linaro.org>
The only cache type left is the flat cache and new other cache types won't be
added since new drivers are supposed to use regmap directly for IO and caching.
This patch removes the snd_soc_cache_ops indirection that was added to support
multiple cache types and modifies the code to always use the flat cache
directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The reg_size field is calculated in snd_soc_register_codec() and then used
exactly once in snd_soc_flat_cache_init(). Since it is calculated based on other
fields from the codec struct just move the calculation to
snd_soc_flat_cache_init() and remove the 'reg_size' field from the codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
reg_def_copy was introduced in commit 3335ddca ("ASoC: soc-cache: Use
reg_def_copy instead of reg_cache_default") to keep a copy of the register
defaults around in case the register defaults where placed in the __devinitdata
section. With the __devinitdata section gone we effectivly keep the same data
around twice. This patch removes reg_def_copy and uses reg_cache_default
directly instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of snd_soc_bulk_write_raw() are left and new drivers are going to use
regmap directly for this, so the function can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of reg_access_defaults are left and new drivers are going to use regmap
for this, so support for it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.
The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysfs_registered field was added to the snd_soc_codec struct in commit
f0fba2ad ("ASoC: multi-component - ASoC Multi-Component Support"), but has never
been used.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAPM context struct has its own field where it stores the pointer to the
DAPM debugfs entry. The debugfs_dapm field in the snd_soc_platform and
snd_soc_codec structs are completely unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The control_type field was used by the core to track which raw IO methods to
use, but when switching to regmap this was no longer necessary and so the last
user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map
API for ASoC generic physical I/O"). The field is now completely unused and can
be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds generic ac97 reset functions using pincontrol and gpio
parsed from devicetree.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The
only difference is that snd_soc_info_enum_double() is also able to handle stereo
controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext()
for the SOC_ENUM_EXT control's info callback allows us to remove
snd_soc_info_enum_ext().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info
callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The
snd_soc_info_volsw_ext() function has been unused ever since then, so remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some devices have the problem that if a internal audio signal source is disabled
the output of the source becomes undefined or goes to a undesired state (E.g.
DAC output goes to ground instead of VMID). In this case it is necessary, in
order to avoid unwanted clicks and pops, to disable any mixer input the signal
feeds into or to active a mute control along the path to the output. Often it is
still desirable to expose the same mixer input control to userspace, so cerain
paths can sill be disabled manually. This means we can not use conventional DAPM
to manage the mixer input control. This patch implements a method for letting
DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable
the control if the path on which the control sits becomes inactive. Userspace
will then only see a cached copy of the controls state. Once DAPM powers the
path up again it will sync the userspace setting with the hardware and give
control back to userspace.
To implement this a new widget type is introduced. One widget of this type will
be created for each DAPM kcontrol which has the auto-disable feature enabled.
For each path that is controlled by the kcontrol the widget will be connected to
the source of that path. The new widget type behaves like a supply widget,
which means it will power up if one of its sinks are powered up and will only
power down if all of its sinks are powered down. In order to only have the mixer
input enabled when the source signal is valid the new widget type will be
disabled before all other widget types and only be enabled after all other
widget types.
E.g. consider the following simplified example. A DAC is connected to a mixer
and the mixer has a control to enable or disable the signal from the DAC.
+-------+
+-----+ | |
| DAC |-----[Ctrl]-| Mixer |
+-----+ : | |
| : +-------+
| :
+-------------+
| Ctrl widget |
+-------------+
If the control has the auto-disable feature enabled we'll create a widget for
the control. This widget is connected to the DAC as it is the source for the
mixer input. If the DAC powers up the control widget powers up and if the DAC
powers down the control widget is powered down. As long as the control widget
is powered down the hardware input control is kept disabled and if it is enabled
userspace can freely change the control's state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The update field of a DAPM context is only assigned while the card's dapm_mutex
is locked, the field is also cleared again while the mutex is stil locked. So
there will only ever be one DAPM context at a time with a non-NULL update field.
So it is safe to move the update field from the DAPM context struct to the card
struct. Doing so will allow further cleanups in this area.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is useful for drivers who want to grab a pointer to
snd_kcontrol outside of the kcontrol callbacks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we can only have a single platform built in with AC'97 support
due to the use of a global variable to provide the bus operations. Fix
this by making that variable a pointer and having the bus drivers set the
operations prior to registering.
This is not a particularly good or nice approach but it avoids blocking
multiplatform and a real fix involves fixing the fairly deep problems
with AC'97 support - we should be converting it to a real bus.
Acked-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and
snd_soc_unregister_platform with the difference that they won't allocate and
free the snd_soc_platform structure.
Also add snd_soc_lookup_platform which looks up a platform by the device it has
been registered for.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's compr_ops structure. Making it
const allows ASoC platform drivers to declare their snd_compr_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's ops structure. Making it const
allows ASoC platform drivers to declare their snd_pcm_ops struct as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does no not modify the driver of a platform. Making it const
allows ASoC platform drivers to declare the snd_soc_platform_driver struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds .name member on snd_soc_component_driver.
But this patch doesn't care about whether cmpnt_drv was NULL,
and/or its name was NULL in snd_soc_register_component()
at this point.
Because, it is easy to switch over to
snd_soc_register_component() from snd_soc_register_dais()
if it doesn't care cmpnt_drv was NULL.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.
As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.
This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This field was added in commit 2e72f8e ("ASoC: New enum type: value_enum"), but
has never been used since. Considering that the soc_enum struct is usually
shared between all instances of a CODEC, it also doesn't make much sense to have
a pointer to DAPM specific data in it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds snd_soc_of_parse_daifmt() and supports below style on DT.
[prefix]format = "i2c";
[prefix]clock-gating = "continuous";
[prefix]bitclock-inversion;
[prefix]bitclock-master;
[prefix]frame-master;
Each driver can use specific [prefix]
(ex simple-card,cpu,dai,format = xxx;)
This sample will be
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT |
SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core does not modify these fields, so they can be made const. This allows
drivers to declare their op tables as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Here we update the asoc structures to add compress stream definations
First the struct snd_soc_dai_driver adds a new member to indicate if the dai is
compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in
the struct snd_soc_dai_link. This is to be used for machine driver to perform
any opertaions required for setting up compressed audio streams
next is the compressed data operations, they are added using struct
snd_compr_ops in the struct snd_soc_platform_driver.
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code handles this fine already, we just need new macros in the header
for drivers to create the controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Control type added for cases where a specific range of values
within a register are required for control.
Added convenience macros:
SOC_SINGLE_RANGE
SOC_SINGLE_RANGE_TLV
Added accessor implementations:
snd_soc_info_volsw_range
snd_soc_put_volsw_range
snd_soc_get_volsw_range
Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prior to this patch, the CPU side of a DAI link was specified using a
single name. Often, this was the result of calling dev_name() on the
device providing the DAI, but in the case of a CPU DAI driver that
provided multiple DAIs, it needed to mix together both the device name
and some device-relative name, in order to form a single globally unique
name.
However, the CODEC side of the DAI link was specified using separate
fields for device (name or OF node) and device-relative DAI name.
This patch allows the CPU side of a DAI link to be specified in the same
way as the CODEC side, separating concepts of device and device-relative
DAI name.
I believe this will be important in multi-codec and/or dynamic PCM
scenarios, where a single CPU driver provides multiple DAIs, while also
booting using device tree, with accompanying desire not to hard-code the
CPU side device's name into the original .cpu_dai_name field.
Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link()
would now be identical. However, two things prevent that at present:
1) The need to save rtd->codec for the CODEC side, which means we have
to search for the CODEC explicitly, and not just the CODEC side DAI.
2) Since we know the CODEC side DAI is part of a codec, and not just
a standalone DAI, it's slightly more efficient to convert .codec_name/
.codec_of_node into a codec first, and then compare each DAI's .codec
field, since this avoids strcmp() on each DAI's CODEC's name within
the loop.
However, the two loops are essentially semantically equivalent.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debugFS files for DPCM link management information.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
e.g. pcm:0,0 routing digital data to 2 external codecs.
FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0
+--> BE (McPDM.0) ----> CODEC 1
e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.
FE pcm:0,0 ---
+--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---
The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.
DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.
Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.
This patch adds the core DPCM code and contains :-
o The FE and BE PCM operations.
o FE and BE DAI link support.
o FE and BE PCM creation.
o BE support API.
o BE and FE link management.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).
This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.
Added convenience macro.
SOC_SINGLE_STROBE
Added accessor implementations.
snd_soc_get_strobe
snd_soc_put_strobe
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.
Added convenience macro.
SOC_SINGLE_XR_SX
Added accessor implementations.
snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct. If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.
This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.
This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there. It is expected that the bias
level callbacks will be used for clock configuration.
Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute(). This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here. We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.
At present we are also restricted to a single DAPM link for the entire
DAI. Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle.
Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros.
Add single register macro : SOC_SINGLE_SX_TLV.
Use snd_soc_info_volsw for .info
Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double.
kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros.
The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In version 3.4 the driver core acquired probe deferral which is a core way
of doing essentially the same thing as ASoC has been doing since forever
to make sure that all the devices needed to make up the card are present
without needing open coding in the subsystem.
Make basic use of this probe deferral mechanism for the cards, removing the
need to handle partially instantiated cards. We should be able to remove
even more code than this, though some of the checks we're currently doing
should stay since they're about things like suppressing unneeded DAPM runs
rather than deferring probes.
In order to avoid robustness issues with our teardown paths (which do need
quite a bit of TLC) add a check for aux_devs prior to attempting to set
things up, this means that we've got a reasonable idea that everything will
be there before we start. As with the removal of partial instantiation
support more work will be needed to make this work neatly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently operations on jack reporting take the CODEC mutex both to protect
the current jack status and also to protect the DAPM run which is triggered
on status updates. Since the addition of a DAPM-specific lock we no longer
need to worry about locking DAPM as it has its own finer grained lock so
create a per jack lock to take care of the jack status.
This is both cleaner where the jack isn't specifically associated with a
CODEC and clearer as it's much more obvious what the lock is protecting.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better
describe all uses for this mutex subclass and align with DAPM too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It has now become necessary to use a DAPM mutex instead of the codec
mutex to lock the DAPM operations. This is due to the recent multi
component support and forth coming Dynamic PCM updates.
Currently we lock DAPM operations with the codec mutex of the calling
RTD context. However, DAPM operations can span the whole card context
and all components.
This patch updates the DAPM operations that use the codec mutex to
now use the DAPM mutex PCM subclass for all DAPM ops.
We also add a mutex subclass for DAPM init and PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the first part of a change that is intended to improve
ASoC locking protection for DAPM and PCM operations.
This part of the series adds a mutex class for the soc_card mutex. The
SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the
SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic
PCM operations. The new mutex classes are required otherwise we will see a false
positive mutex deadlock warning between the card initialisation and the PCM
operations (something that would never deadlock in real life).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add mutex support for platform IO operations. e.g. can be used
for platform DAPM widget IO ops.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Chip designers frequently include things like the enable and disable
controls for algorithms in the register blocks which also hold the
coefficients. Since it's desirable to split out the enable/disable
control from userspace the plain SND_SOC_BYTES() isn't optimal for
these devices.
Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word
of the block to be excluded from the control. This supports the needs
of devices I've looked at and lets us have a reasonably simple API.
Further controls can be added in future if that's needed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow devices to export blocks of registers to the application layer,
intended for use for reading and writing coefficient data which can't
usefully be worked with by the kernel at runtime (for example, due to
requiring complex and expensive calculations or being the results of
callibration procedures). Currently drivers are using platform data to
provide configurations for coefficient blocks which isn't at all
convenient for runtime management or configuration development.
Currently only devices using regmap are supported, an error will be
generated for any attempt to work with a byte control on a non-regmap
device. There's no fundamental block to other devices so support could
be added if required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Neater and avoids warnings when used in other places where const strings
are desired.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow platform widgets to be visible in debugfs like codec widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is usually not a use case dependant flag anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.
This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-
snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()
This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().
It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().
Finally, it updates the existing machine drivers that register controls to either :-
1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.
In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a driver is using regmap directly ensure that we're coherent with
non-ASoC register updates by using the regmap API directly to do our
read/modify/write cycles. This will bypass the ASoC cache but drivers
using regmap directly should not be using the ASoC cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Most devices accept data in formats that don't correspond directly to
their internal format. ALSA allows us to set a msbits constraint which
tells userspace about this in case it finds it useful (for example, in
order to avoid wasting effort dithering bits that will be ignored when
raising the sample size of data) so provide a mechanism for drivers to
specify the number of bits that are actually significant on a DAI and
add the appropriate constraints along with all the others.
This is done slightly awkwardly as the constraint is specified per sample
size - we loop over every possible sample size, including ones that the
device doesn't support and including ones that have fewer bits than are
actually used, but this is harmless as the upper layers do the right thing
in these cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The device model needs a release() function so it can free devices when
they become dereferenced. Do that for rtds.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that everything is seeing the same declaration by moving it to
a header file rather than putting the declaration in soc-core.c
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DAI link endpoints and platform (DMA) devices are currently specified
by name. When instantiating sound cards from device tree, it may be more
convenient to refer to these devices by phandle in the device tree, and
for code to describe DAI links using the "struct device_node *"
("of_node") those phandles map to.
This change adds new fields to snd_soc_dai_link which can "name" devices
using of_node, enhances soc_bind_dai_link() to allow binding based on
of_node, and enhances snd_soc_register_card() to ensure that illegal
combinations of name and of_node are not used.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement snd_soc_of_parse_audio_routing(), a utility function that can
parses a simple DAPM route table from device tree.The machine driver
specifies the DT property to use, since this is binding-specific.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement snd_soc_of_parse_card_name(), a utility function that sets a
card's name from device tree. The machine driver specifies the DT
property to use, since this is binding-specific.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A card is fully routed if the DAPM route table describes all connections on
the board.
When a card is fully routed, some operations can be automated by the ASoC
core. The first, and currently only, such operation is described below, and
implemented by this patch.
Codecs often have a large number of external pins, and not all of these pins
will be connected on all board designs. Some machine drivers therefore call
snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core
never to activate them.
However, when a card is fully routed, the information needed to derive the
set of unused pins is present in card->dapm_routes. In this case, have
the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused
codec pin.
This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no current users and new drivers ought to be using the regmap
API and its cache implementation directly so just delete the ASoC copy.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My usual technique for finding definitions is to search for "name {"
which breaks with the extra space.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this flag, each dai_link in machine driver can choose
to ignore pmdown_time during DAPM shut down sequence.
If the ignore_pmdown_time is set, the DAPM for corresponding DAI
will be executed immediately.
Signed-off-by: Ramesh Babu K V <ramesh.babu@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this flag codec drivers can indicate that it is desired
to ignore the pmdown_time for DAPM shutdown sequence when
playback stream is stopped.
The DAPM sequence will be executed without delay in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By accident few places still uses the _2r calls from
the core.
This is a quick fix, the drivers using the old callbacks
going to be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We do not have users for snd_soc_put_volsw_2r anymore.
It can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the put_volsw/put_volsw_2r in one function.
To avoid build breakage in twl6040 keep the
snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the get_volsw/get_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the info_volsw/info_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the
bits are within one register.
Assign .rreg to be the same as .reg for these types.
With this change we can tell if the mixer in question:
is mono:
mc->reg == mc->rreg && mc->shift == mc->rshift
is stereo, within single register:
mc->reg == mc->rreg && mc->shift != mc->rshift
is stereo, in two registers:
mc->reg != mc->rreg
The patch provide a small inline function to query, if the mixer
is stereo, or mono.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to reduce the number of DAPM power checks we run keep a list of
widgets which have been changed since the last DAPM run and iterate over
that rather than the full widget list. Whenever we change the power state
for a widget we add all the source and sink widgets it has to the dirty
list, ensuring that all widgets in the path are checked.
This covers more widgets than we need to as some of the neighbour widgets
won't be connected but it's simpler as a first step. On one system I tried
this gave:
Power Path Neighbour
Before: 207 1939 2461
After: 114 1066 1327
which seems useful.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the new macro we can remove duplicated code
for the SOC_DOUBLE_R type of controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the new macro we can remove duplicated code
for the SOC_DOUBLE type of controls.
We can also remap the SOC_SINGLE_VALUE macro to
SOC_DOUBLE_VALUE
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.
Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.
We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices that need this exist; obviously the newer regmap defaults
mechanism will deal with this more happily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow drivers to set up their own regmap API structures. This is mainly
useful with MFDs where the core driver will have set up regmap at the
minute, though it may make sense to push the existing regmap setup out
of the core into the drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove all the ASoC specific physical I/O code and replace it with calls
into the regmap API. The bulk write code can only be used safely if all
regmap calls are locked with the CODEC lock, we need to add bulk support
to the regmap API or replace the code with an open coded loop (though
currently it has no users...).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In preparation for Dynamic PCM (AKA DSP) support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs
after completion of stream events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform driver widgets to perform any IO required for DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow ASoC machine drivers to register a driver name
and a longname. This allows user space to determine
the flavour of machine driver.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The enum texts are supposed to be const char * const []. Without the
second const, it gets compile warnings like
sound/soc/codecs/max98095.c:607:2: warning: initialization discards qualifiers from pointer target type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Those should not be modified (and are not) by the core code, so make them const.
This also makes them consistent with the same members of snd_soc_codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide the top level ASoC core functions for indicating whether
a given register is readable or writable.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using struct snd_soc_reg_access for the read/write/vol attributes
of the registers, we provide callbacks that automatically determine whether
a given register is readable/writable or volatile.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is mainly used by the soc-cache code to easily determine the
currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we
can distinguish it if it is not initialized or set.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As it has become more common to have to write firmware or similar
large chunks of data to the hardware, add a function to perform
raw bulk writes that bypass the cache. This only handles volatile
registers as we should avoid getting out of sync with the actual
cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This is run after the DAPM widgets and routes are added, allowing setup
of things like jacks using the routes. The main card probe() is run before
anything else so can't be used for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
These will be added after all devices are registered and allow most DAI
init functions in machine drivers to be replaced by simple data.
Regular controls are not supported as the registration function still
works in terms of CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This means that rather than adding the board specific DAPM widgets to a
random CODEC DAPM context they can be added to the card itself which is
a bit cleaner. Previously there only was one DAPM context and it was
tied to the single supported CODEC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds soc-jack support for adding voltage zones and for
detecting jack type
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide driver data for cards within the card structure. To simplify the
implementation of the PM operations we don't use the struct device driver
data as this is used by the core to retrieve the card in callbacks from
the device model and PM core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allows drivers to distinguish which subsequence is being notified when
they get called back.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Could just as well live in sysfs but sysfs doesn't have the simple
value export helpers debugfs does.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow hookup of cards registered directly with the core to the PM
operations by exporting the device power management operations to
modules, also exporting the default PM operations since it is
expected that most cards will end up using exactly the same setup.
Note that the callbacks require that the driver data for the card be
the snd_soc_card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We generally refer to registers as unsigned ints (including in the
underlying CODEC driver operation).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is primarily needed to avoid writing back to the cache
whenever we are syncing the cache with the hardware. This gives a
performance benefit especially for large register maps.
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>
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The machine driver can't register the card directly and need to do this thru
soc-audio device creation
This patch allows the register and unregister card to be directly called by
machine drivers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the soc_probe initializes the card hence it does the card list
initialzation. But if machines directly register the card they would need to
do these steps, so putting them as inline would save lot of code
This patch adds an inline to do list initialzation
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <harsha.priya@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that all calls to readable_register()/volatile_register() go via
the snd_soc_codec function pointers.
If the default register access table has been given but no functions
for handling readable()/volatile() registers, use the default ones provided
by soc-cache.
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>
For common scenarios, device drivers can provide a table of all the
registers that are at least either readable/writable/volatile. The idea
is that if a register lookup fails, all of its read/write/vol members
will be zero and will be treated as default. This also reduces the
size of the register access array.
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>
Simplify the use of reg_size, by calculating it once and storing it in
the codec structure for later reference. The value of reg_size is
reg_cache_size * reg_word_size.
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This 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>
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>
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>
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>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.
Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.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>
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.
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>
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Several shortcuts for popular uses of some of SOC_ENUM_* and
SOC_VALUE_ENUM_* macros.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps CODECs with sparse register maps work better with the
register cache display interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for
double controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double
controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses. However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.
This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended. At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They are now only accessed within dapm_power_widgets() so can be local
to that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Should be no impact on the generated code but it helps the compiler
print clearer messages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a macro for double controls with special callback functions.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.
This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.
[Edited to raise priority of error log message and document parameters.
-- broonie]
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a jack reporting interface to ASoC. This wraps the ALSA
core jack detection functionality and provides integration with DAPM to
automatically update the power state of pins based on the jack state.
Since embedded platforms can have multiple detecton methods used for a
single jack (eg, separate microphone and headphone detection) the report
function allows specification of which bits are being updated on a given
report.
The expected usage is that machine drivers will create jack objects and
then configure jack detection methods to update that jack.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge the recently introduced soc_value_enum structure to the soc_enum.
The value based enums are still handled separately from the normal enum types,
but with the merge some of the newly introduced functions can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces a new enum type.
In this enum type each enumerated items referred with a value.
This new enum type can handle enums encoded in bitfield, or any other
weird ways. twl4030 codec has several mux selection register, where the
input/output mux is coded in a bitfield. With the normal enum type this type
of mux can not be handled in a clean way.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 factors most of the contents of soc.h out into separate headers,
including soc-dai.h for the DAI. Factor the existing DAI API out into
this file in order to prepare for backporting of the ASoC v2 DAI API.
Also backport some of Liam's improvements to the documentation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASOC: convert use of uint to unsigned int
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most of the ASoC controls refer to the maximum value that can be set for
a control as mask but there is no actual requirement for all bits to be
set at the highest possible value making the name mask misleading.
Change the code to use max instead.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert bitfields in ASoC into full int width. This is a
simple mechanical conversion. Two places in the DAPM code
were fixed to properly use mask.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some codecs have unusual features in their register maps such as very
large registers representing arrays of coefficients. Support these
codecs in the register cache sysfs file by allowing them to provide a
function register_display() overriding the default output for register
contents.
Also ensure that we don't overflow PAGE_SIZE while writing out the
register dump.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.
This merger removes duplication in both DAI structures and simplifies
the API for other users.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On OpenMoko soc-audio resume is taking 700ms of the whole resume time of
1.3s, dominated by writes to the codec over I2C. This patch shunts the
resume guts into a workqueue which then is done asynchronously.
The "card" is locked using the ALSA power state APIs as suggested by
Mark Brown.
[Added fix for race with resume to suspend and fixed a couple of nits
from checkpatch -- broonie.]
Signed-off-by: Andy Green <andy@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mike Montour <mail@mmontour.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.
Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.
Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The CPU and codec DAI operations differ only in the presence of the
digital mute operation for the codec so they may as well be the same
type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I suspect that snd_ctl_boolean_mono should have been
snd_ctl_boolean_mono_info instead. This fixes the build for magician.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This header file exists only for some hacks to adapt alsa-driver
tree. It's useless for building in the kernel. Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it. This should be really killed in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clean up codes using the new common snd_ctl_boolean_*_info() callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes a bug whereby AC97 bus device data was being clobbered
when AC97 codecs using the generic ac97_codec.c driver were being
registered. Codecs that didn't use the generic driver were unaffected
(e.g. WM9712, WM9713).
Changes:-
o Add new AC97 codec class for custom (or need bus dev registration)
AC97 codecs.
o Only register/deregister this custom codec device with the AC97 bus.
The generic AC97 driver already does this for generic codec devices.
This may be related to bug #3038 :-
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch updates the API's to include the new DAI configuration and
clocking architecture.
Changes:-
o Removed DAI automatic matching and capabilities structure (struct
snd_soc_dai_mode) and macros.
o Added DAI operations for codec and CPU interfaces.
o Removed config_sysclk() function and struct snd_soc_clock_info. No
longer needed as clocking is now configured manually in the machine
drivers. Also removed other clocking data from structures.
o Updated version to 0.13
o Added shift to SOC_SINGLE_EXT kcontrol macro.
Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix the changes realted to delayed_work in soc/codecs/wm8750.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
From: Andrew Morton <akpm@osdl.org>
I converted the workqueues to per-device while I was there. It seems
strange to create a new kernel thread (on each CPU!) and to then only
have a single global work to ever be queued upon it.
Plus without this, I'd have to use the _NAR stuff, gawd help me.
Does that workqueue really need to be per-cpu?
Does that workqueue really need to exist? Why not use keventd?
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the ASoC and DAPM headers.
Features:-
o Defines Digital Audio Interface (DAI) API
o Defines Codec, Platform and Machine API
o Defines Dynamic Audio Power Management API
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
