For CODEC controls snd_kcontrol_chip() currently returns a pointer to the
CODEC that registered the control. With the upcoming consolidation of
platform and CODEC controls this will change. Prepare for this by introducing
the snd_soc_kcontrol_codec() helper function that will hide the implementation
details of how the CODEC for a control can be obtained. This will allow us to
change this easily in the future.
The patch also updates all CODEC drivers to use the new helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the default regmap I/O setting to snd_soc_register_codec() while
the CODEC is initialising, which will be called by CODEC driver device
probe(), and then we can make XXX_set_cache_io() go away entirely from each
CODEC ASoC probe.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Separate HW initialization from device creation.
This is needed for suspend/resume support.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from last week - use of the tdm_slot mapping from
Xiubo plus a few smaller fixes and cleanups.
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Merge tag 'asoc-v3.15-4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Final updates for v3.15 merge window
A few more updates from last week - use of the tdm_slot mapping from
Xiubo plus a few smaller fixes and cleanups.
The 'of_' is not appropriate here for there hasn't any DT parsing.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few more updates for the merge window:
- Fixes for the simple-card DAI format DT mess.
- A new driver for Cirrus cs42xx8 devices.
- DT support for a couple more devices.
- A revert of a previous buggy fix for soc-pcm, plus a few more fixes
and cleanups.
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Merge tag 'asoc-v3.15-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
A few more updates for the merge window:
- Fixes for the simple-card DAI format DT mess.
- A new driver for Cirrus cs42xx8 devices.
- DT support for a couple more devices.
- A revert of a previous buggy fix for soc-pcm, plus a few more fixes
and cleanups.
Add a helper function to cast back from a component struct to the CODEC struct
it is embedded in. This is useful in situations where we know that a certain
component is a CODEC and want to get access to some CODEC specific properties.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is mostly a few additional fixes from Lars-Peter, a new driver and
cleaning up a git failure with merging the Intel branch (combined with
an xargs failure to pay attention to error codes). The history lists a
bunch of additional commits for the branch but the content of those
commits is actually present already but not recorded in history due to
git failing. Unfortunately xargs is used in the merge script and it
doesn't do a good job of noticing errors from the commands it invokes.
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Merge tag 'asoc-v3.15-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
This is mostly a few additional fixes from Lars-Peter, a new driver and
cleaning up a git failure with merging the Intel branch (combined with
an xargs failure to pay attention to error codes). The history lists a
bunch of additional commits for the branch but the content of those
commits is actually present already but not recorded in history due to
git failing. Unfortunately xargs is used in the merge script and it
doesn't do a good job of noticing errors from the commands it invokes.
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
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Merge tag 'asoc-v3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
Now that all users have been converted to regmap and the config.reg_bits
and config.val_bits can be setted by each user through regmap core API.
So these two params are redundant here.
Since the only control type that left is SND_SOC_REGMAP, so remove it. Drop
the control params and add struct regmap *regmap to simplify the code.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 83fc3bc095.
sh-specific "CCR" and "CCR2" have been prefixed by "SH_" in commit
a5f6ea29f9 ('sh: prefix sh-specific "CCR" and
"CCR2" by "SH_"').
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that every DAI has a component we can track the DAIs on a per component
basis. This simplifies the DAI lookup when we are only interested in DAIs of a
specific component and also makes it possible to have multiple components with
the same parent device and also register DAIs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In preparation for componentization move the ignore_pmdown_time field from the
snd_soc_codec struct to the snd_soc_component struct. Set it to true for non
CODEC components for now.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no reason why active count tracking should only be done for CODECs but
not for other components. Moving the active count from the snd_soc_codec struct
to the snd_soc_component struct reduces the differences between CODECs and other
components and will eventually allow component to component DAI links (Which is
a prerequisite for converting CODECs to components).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Keep track of which component registered a DAI. We'll need this as
componentization progresses.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of directly checking the 'active' field of the CODEC struct add a new
helper function that will return either true or false depending on whether the
CODEC is active. This will make the migration to the component level easier.
The patch also updates all CODEC drivers that check the active attribute to use
the new helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have the same code that increments and decrements the active field of the
various PCM runtime components (all with the same bugs). Factor this out into
common helper functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
R-Car sound has SCU unit which has SRC/CTU/MIX/DVC,
and current rsnd driver has scu.c and scu module.
Current scu.c has SRC support only.
My first concept was control these feature on scu.c
but, it become difficult and un-understandable now.
This patch rename scu to src
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is assuming that SCU/SRU ID is
same as SSIU/SSI ID, because Gen1 can't select it.
But, Gen2 can select it.
The SCU/SRU/SSIU/SSI pair depends on the platform.
This patch get correct SCU ID from platform info.
To keep compatible, it still assuming SCU ID = SSI ID
if platform doesn't have info
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound DAI consists from SSI/SCU/SSIU/SRU...
Current R-Car sound DAI is decided from these settings,
but it is intuitively unclear, and is not good design for DT support.
This patch adds new rsnd_dai_platform_info to solve this issue.
But now, many platform is using this driver without
rsnd_dai_platform_info.
So, this patch still supports DAI settings via SSI to keep compatible.
It will be removed in next Linux version.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
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>
The hda codecs all use this struct and, with an HDA platform driver,
will be able to be built without PCI.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MUXs and virtual MUXs are almost identical, the only difference is that for
virtual MUX there is no hardware backing register in which setting is stored.
This patch adds code, which is similar to what we already do for DAPM mixer
controls to support virtual mixer controls, to DAPM enum controls. The new code
will check if the enum does a hardware backing register and skip over reading
and writing to the register if it has not. This allows us to use the same code
path for both MUXs and virtual MUXs and a lot of nearly identical code can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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>
MUXs and value MUXs are almost identical, the only difference is that a value
MUX uses a look-up table to map from the selected control item to a register
value, while MUXs use a direct mapping. This patch uses
snd_soc_enum_item_to_val() and snd_soc_enum_val_to_item(), which where earlier
introduced during the consolidation of enum and value enum controls, to hide
this difference. This allows us to use the same code path for both MUXs and
value MUXs and a lot of nearly duplicated code can be removed.
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>
A few code cleanups and optimizations. In addition, drop
snd_device_disconnect() that isn't used at all, and drop the return
values from snd_device_free*().
Another slight difference by this change is that now the device state
will become always SNDRV_DEV_REGISTERED no matter whether dev_register
ops is present or not. It's for better consistency. There should be
no impact for the current tree, as the state isn't checked.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Basically, the device type specifies the priority of the device to be
registered / freed, too. However, the priority value isn't well
utilized but only it's checked as a group. This results in
inconsistent register and free order (where each of them should be in
reversed direction).
This patch simplifies the device list management code by simply
inserting a list entry at creation time in an incremental order for
the priority value. Since we can just follow the link for register,
disconnect and free calls, we don't have to specify the group; so the
whole enum definitions are also simplified as well.
The visible change to outside is that the priorities of some object
types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object
is registered before others (control, PCM, etc) and, in return,
released after others. Similarly, SNDRV_DEV_CODEC is in a lower
priority than SNDRV_DEV_BUS for ensuring the dependency.
Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and
SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like PCM, allow hwdep to be assigned to a different parent device
than the card. It'll be used for the HD-audio codec device in the
later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling each time device_create_file(), create the groups
of sysfs attribute files at once in a normal way. Add a new helper
function, snd_get_device(), to return the associated device object,
so that we can handle the sysfs addition locally.
Since the sysfs file addition is done differently now,
snd_add_device_sysfs_file() helper function is removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some CPU/CODEC DAI devices the slot information maybe needed. This
patch adds the slot information parsing for simple-card driver.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
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>
'snd_rawmidi_transmit_reset()' and 'snd_rawmidi_receive_reset()' are declared
but not defined.
This state has been continue over 10 years. So let us remove them.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We will often call sync after several functions that require the DAPM
mutex to be held. Rather than release and immediately relock the mutex
provide an unlocked function for this situation.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
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>
The snd_soc_dapm_xxxx_pin all require the dapm_mutex to be held when
they are called as they edit the dirty list, however very few of the
callers do so.
This patch adds unlocked versions of all the functions replacing the
existing implementations with one that holds the lock internally. We
also fix up the places where the lock was actually held on the caller
side.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.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>
Using __bitwise and typedefs for the attributes of snd_device struct
isn't so useful, and rather it worsens the readability. Let's drop
them and use the straightforward enum.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
For simplicity, introduce new helpers for pcm stream, pcm_err(), etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the own refcount but use the standard device refcounting via
get_device() and put_device(). Introduce a new completion to snd_card
instead of the wait queue for syncing the last release, which is used
in snd_card_free().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As prepared in the previous patch, we are ready to create a device
struct for the card object in snd_card_create() now. This patch
changes the scheme from the old style to:
- embed a device struct for the card object into snd_card struct,
- initialize the card device in snd_card_create() (but not register),
- registration is done in snd_card_register() via device_add()
The actual card device is stored in card->card_dev. The card->dev
pointer is kept unchanged and pointing to the parent device as before
for compatibility reason.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a part of preliminary works for modernizing the ALSA device
structure. So far, we set card->dev at later point after the object
creation. Because of this, the core layer doesn't always know which
device is being handled before it's actually registered, and it makes
impossible to show the device in error messages, for example. The
first goal is to achieve a proper struct device initialization at the
very beginning of probing.
As a first step, this patch introduces snd_card_new() function (yes
there was the same named function in the very past), in order to
receive the parent device pointer from the very beginning.
snd_card_create() is marked as deprecated.
At this point, there is no functional change other than that. The
actual change of the device creation scheme will follow later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last argument, name, of snd_oss_register_device() is nowhere
referred in the function in the current code. Let's drop it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The platform data structure contains information which is used only by
the driver, and the driver allocates platform information fields which
are of no use.
Move the driver specific data to a new private structure and cleanup
the platform data structure.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
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>
Renesas sound Gen2 has SRC (= Sampling Rate Converter)
which needs 2 DMAC.
The data path image when you use SRC on Gen2 is
[mem] -> Audio-DMAC -> SRC -> Audio-DMAC-peri-peri -> SSIU -> SSI
This patch support SRC and DMAEnine.
It is tested on R-Car H2 Lager board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas Chip is supporting multi pin sound,
but the HW setting is very difficult and confusable.
But driver is supporting it halfway.
Remove SYNC option at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This branch is reducing in size for every release since most board-related
changes have started happening in devicetrees now. Still, we have some things
going on here.
* Renesas platforms are still adding a bit more legacy device support, something
that should trail off shortly as they move to full DT.
* We group most defconfig updates into this branch out of old habits
* Removal of legacy OMAP2 platforms over to DT continues, and a handful of old
code is being removed here.
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Merge tag 'boards-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC board updates from Olof Johansson:
"This branch is reducing in size for every release since most
board-related changes have started happening in devicetrees now.
Still, we have some things going on here.
* Renesas platforms are still adding a bit more legacy device
support, something that should trail off shortly as they move to
full DT
* We group most defconfig updates into this branch out of old habits
* Removal of legacy OMAP2 platforms over to DT continues, and a
handful of old code is being removed here"
* tag 'boards-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (94 commits)
ARM: dts: OMAP2: fix interrupt number for rng
ARM: dts: Split omap3 pinmux core device
ARM: dts: Add omap specific pinctrl defines to use padconf addresses
ARM: bcm2835: bcm2835_defconfig updates
ARM: msm_defconfig: Enable restart driver
defconfig: msm_defconfig: Enable CONFIG_ARCH_MSM8974
ARM: msm: Add support for APQ8074 Dragonboard
ARM: exynos_defconfig: Enable S2MPS11 voltage regulator
ARM: tegra: Enable DRM panel support
ARM: shmobile: mackerel: Fix USBHS pinconf entry
ARM: shmobile: Let Koelsch multiplatform boot with Koelsch DTB
ARM: shmobile: Let Lager multiplatform boot with Lager DTB
ARM: shmobile: Remove non-multiplatform Koelsch reference support
ARM: shmobile: Remove non-multiplatform Lager reference support
ARM: shmobile: koelsch-reference: Instantiate clkdevs for SCIF and CMT
ARM: shmobile: lager-reference: Instantiate clkdevs for SCIF and CMT
ARM: shmobile: koelsch-reference: Remove duplicate CCF initialization
ARM: shmobile: lager-reference: Enable multiplaform kernel support
ARM: shmobile: armadillo: Set backlight enable GPIO
ARM: shmobile: Koelsch: add Ether support
...
Conflicts:
arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c
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>
The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A bit of special care is necessary when creating the intersection of two rate
masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and
SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two
rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a
specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of
discrete rates specified by a list constraint. For all other cases the supported
rates are specified directly in the rate mask.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Nowadays we have CMA for obtaining the contiguous memory pages
efficiently. Let's kill the old kludge for reserving the memory pages
for large buffers. It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Connect the DAPM graph through each BE DAI link to the componnent(s) on the
other side of the BE DAI link. This allows the graph to be walked on
both sides of the link when graph changes are made.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI.
This is for internal DAPM usage only and is used to determine whether to
insert a DAI link connection into the DAPM graph.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.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>
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
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Merge tag 'asoc-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.14
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add helpers for obtaining the width of a format directly from params
since this is expected to become a common operation in ASoC.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
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Merge tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas into next/boards
From Simon Horman:
Renesas ARM based SoC board updates for v3.14
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
* tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas: (75 commits)
ARM: shmobile: lager: add gpio/fixed regulator for SDHI
ARM: shmobile: bockw: remove unused RSND_SSI_CLK_FROM_ADG
ARM: shmobile: armadillo: fixup FSI address size
ARM: Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
ARM: shmobile: mackerel: Use pinconf API to configure pin pull-down
ARM: shmobile: Lager:add SPI FLASH support on QSPI
ARM: shmobile: mackerel: clk_round_rate() can return a zero to indicate an error
ARM: shmobile: Add pinctrl_register_mappings() for Koelsch
ARM: shmobile: Use ->init_late() on Lager
ARM: shmobile: Hook up SW30-SW36 on Koelsch
ARM: shmobile: koelsch: mark GPIO keys as wake-up sources
ARM: shmobile: Use ->init_late() on Koelsch
ARM: shmobile: lager: mark GPIO keys as wake-up sources
ARM: shmobile: r7s72100 Genmai Multiplatform
ARM: shmobile: r7s72100 Genmai DT reference C bits
ARM: shmobile: r7s72100 Genmai DT reference DTS bits
ARM: shmobile: Initial r8a7791 and Koelsch multiplatform support
ARM: shmobile: koelsch: set .debounce_interval
ARM: shmobile: lager: set .debounce_interval
ARM: shmobile: bockw: add pin pull-up setting for SDHI
...
Signed-off-by: Olof Johansson <olof@lixom.net>
spear_pcm_request_chan() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:
a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
pointer rather than some custom type.
b) dma_data->filter_data rather than dma_data should be passed to
snd_dmaengine_pcm_request_channel() as the filter data.
Make minor changes to the SPEAr DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When running a 32bit kernel the hda_intel driver is still reporting
a 64bit dma_mask if the HW supports it.
From sound/pci/hda/hda_intel.c:
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
else {
pci_set_dma_mask(pci, DMA_BIT_MASK(32));
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32));
}
which means when there is a call to dma_alloc_coherent from
snd_malloc_dev_pages a machine address bigger than 32bit can be returned.
This can be true in particular if running the 32bit kernel as a pv dom0
under the Xen Hypervisor or PAE on bare metal.
The problem is that when calling setup_bdle to program the BLE the
dma_addr_t returned from the dma_alloc_coherent is wrongly truncated
from snd_sgbuf_get_addr if running a 32bit kernel:
static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= PAGE_MASK;
return addr + offset % PAGE_SIZE;
}
where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr.
Without this patch the HW will fetch the 32bit truncated address,
which is not the one obtained from dma_alloc_coherent and will result
to a non working audio but can corrupt host memory at a random location.
The current patch apply to v3.13-rc3-74-g6c843f5
Signed-off-by: Stefano Panella <stefano.panella@citrix.com>
Reviewed-by: Frediano Ziglio <frediano.ziglio@citrix.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
92eba04e4b
(ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG) removed
RSND_SSI_CLK_FROM_ADG, it is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Simon Horman <horms+renesas@verge.net.au>
Add fields to struct snd_dmaengine_pcm_config to allow custom:
- DMA channel names.
This is useful when the default "tx" and "rx" channel names don't
apply, for example if a HW module supports multiple channels, each
having different DMA channel names. This is the case with the FIFOs
in Tegra's AHUB. This new facility can replace
SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.
- DMA device
This allows requesting DMA channels for a device other than the device
which is registering the "PCM" driver. This is quite unusual, but is
currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
module contains its own FIFOs which DMA writes to. However, in Tegra30,
the DMA FIFOs were split out AHUB HW module, which then routes the data
through a cross-bar, and into the DAI HW modules. However, the current
ASoC driver structure does not expose this detail, and acts as if the
FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
driver is registered with the DAI HW module, yet the DMA channels must
be looked up in the AHUB HW module's device tree node. This new config
field allows that to happen. Eventually, the Tegra drivers will be
reworked to fully expose the AHUB, and this config field can be
removed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
MICA/B Single-Ended input selection depends on mica/b config so lets
make the mixer controls for them only show for selected mic's
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch reworks the MICA an MICB config for single-ended or
differential and the selection of which MIC for the single config
Signed-off-by: Brian Austin <brian.austin@cirrus.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>
SND_SOC_DAPM_MUX() doesn't currently initialize the .mask field. This
results in the mux never affecting HW, since no bits are ever set or
cleared. Fix SND_SOC_DAPM_MUX() to use SND_SOC_DAPM_INIT_REG_VAL() to
set up the reg, shift, on_val, and off_val fields like almost all other
SND_SOC_xxx() macros. It looks like this was a "typo" in the fixed
commit linked below.
This makes the speakers on the Toshiba AC100 (PAZ00) laptop work again.
Fixes: de9ba98b6d ("ASoC: dapm: Make widget power register settings more flexible")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: <stable@vger.kernel.org> # v3.12+
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>
The drain and drain_notify callback were blocked by low level driver
until the draining was complete. Due to this being invoked with big
fat mutex held, others ops like reading timestamp, calling pause, drop
were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would
be required to be invoked by low level driver when drain or partial
drain has been completed by the DSP. Thus we make the drain and
partial_drain callback as non blocking and driver returns immediately
after notifying DSP. The waiting is done while releasing the lock so
that other ops can go ahead.
[ The commit 917f4b5cba was wrongly applied from the preliminary
patch. This commit corrects to the final version.
Sorry for inconvenience! -- tiwai ]
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
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Merge tag 'asoc-v3.13-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.13
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
The size of the register cache array is actually 6 instead of 7,
as it caches up to AK4114_REG_INT1_MASK. This resulted in unexpected
access out of array range, although most of them aren't so serious
(just reading one more byte on the stack at snd_ak4114_create()).
Also, the check of cache size was wrongly done by checking with
sizeof() instead of ARRAY_SIZE(). Fixed this together.
(And yes, hardcoded numbers are bad, but I keep the coding style as is
for making it clear what this patch actually does.)
Spotted by coverity among several CIDs, e.g. 711621.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds platform data support for a reset GPIO.
Also uses reset_gpio to toggle reset of the CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
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Merge tag 'asoc-v3.13' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.13
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
The drain and drain_notify callback were blocked by low level driver untill the
draining was complete. Due to this being invoked with big fat mutex held, others
ops like reading timestamp, calling pause, drop were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would be required
to be invoked by low level driver when drain or partial drain has been completed
by the DSP. Thus we make the drain and partial_drain callback as non blocking
and driver returns immediately after notifying DSP.
The waiting is done while relasing the lock so that other ops can go ahead.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that we can't use gen_pool_*() functions on archs
without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since
linux/genalloc.h doesn't provide dummy functions for all. We'd be
able to fix linux/genalloc.h size, but I take an easier path for
now...
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have more than just simple TX and RX DMA channels, for example
modern Samsung I2S IPs support a secondary transmit DMA stream which is
mixed into the primary stream during playback. Allow such devices to
specify the names of the channels to be requested in their dma_data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Now it's quite common that an SoC contains its on-chip internal RAM.
By using this RAM space for DMA buffer during audio playback/record,
we can shutdown the voltage for external RAM to save power.
So add new DEV type with iram malloc()/free() and accordingly modify
current default mmap() for the iram circumstance.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current SSI needs RSND_SSI_DEPENDENT flag to
decide dependent/independent mode.
And SCU needs RSND_SCU_USE_HPBIF flag
to decide HPBIF is enable/disable.
But these 2 means same things.
This patch adds new rsnd_scu_hpbif_is_enable()
function, and merges above methods.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow DMA data to be set at probe time for devices that can do that,
avoiding the need to do it every time we start a stream and supporting
non-DT dmaengine users using the helpers.
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently each platform making use the the generic dmaengine PCM driver still
needs to provide a custom snd_pcm_hardware struct which specifies the
capabilities of the DMA controller, e.g. the maximum period size that can be
supported. This patch adds code which uses the newly introduced
dma_get_slave_caps() API to query this information from the dmaengine driver.
The new code path will only be taken if the 'pcm_hardware' field of the
snd_dmaengine_pcm_config struct is NULL.
The patch also introduces a new 'fifo_size' field to the
snd_dmaengine_dai_dma_data struct which is used to initialize the
snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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>
Add support for RST GPIO and Charge Pump Freq in platform data
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a comment to the trigger function in snd_soc_dai_ops struct about
possible command sequences.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rcar is using rsnd_is_gen1/gen2() to checking its
IP generation, but it needs data mask.
This patch fixes it up.
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>
Some codec drivers when running in slave mode require that BCLK to sample rate ratio
is explicitly set by the machine driver as it may not be exactly rate * frame size.
Extend the DAI API by adding :-
int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.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>
Pull media updates from Mauro Carvalho Chehab:
"This series contains:
- Exynos s5p-mfc driver got support for VP8 encoder
- Some SoC drivers gained support for asynchronous registration
(needed for DT)
- The RC subsystem gained support for RC activity LED;
- New drivers added: a video decoder(adv7842), a video encoder
(adv7511), a new GSPCA driver (stk1135) and support for Renesas
R-Car (vsp1)
- the first SDR kernel driver: mirics msi3101. Due to some troubles
with the driver, and because the API is still under discussion, it
will be merged at staging for 3.12. Need to rework on it
- usual new boards additions, fixes, cleanups and driver
improvements"
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (242 commits)
[media] cx88: Fix regression: CX88_AUDIO_WM8775 can't be 0
[media] exynos4-is: Fix entity unregistration on error path
[media] exynos-gsc: Register v4l2 device
[media] exynos4-is: Fix fimc-lite bayer formats
[media] em28xx: fix assignment of the eeprom data
[media] hdpvr: fix iteration over uninitialized lists in hdpvr_probe()
[media] usbtv: Throw corrupted frames away
[media] usbtv: Fix deinterlacing
[media] v4l2: added missing mutex.h include to v4l2-ctrls.h
[media] DocBook: upgrade media_api DocBook version to 4.2
[media] ml86v7667: fix compile warning: 'ret' set but not used
[media] s5p-g2d: Fix registration failure
[media] media: coda: Fix DT driver data pointer for i.MX27
[media] s5p-mfc: Fix input/output format reporting
[media] v4l: vsp1: Fix mutex double lock at streamon time
[media] v4l: vsp1: Add support for RT clock
[media] v4l: vsp1: Initialize media device bus_info field
[media] davinci: vpif_capture: fix error return code in vpif_probe()
[media] davinci: vpif_display: fix error return code in vpif_probe()
[media] MAINTAINERS: add entries for adv7511 and adv7842
...
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>
snd_soc_dapm_new_widgets() works on the ASoC card as a whole not on a specific
DAPM context. The DAPM context that is passed as the parameter is only used to
look up the pointer to the card. This patch updates the signature of
snd_soc_dapm_new_widgets() to take the card directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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>
Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not
defined. Add a wrapper macro around printk_limit() that is conditionally defined by
CONFIG_SND_DEBUG.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move include/sound/tea575x-tuner.h to include/media/tea575x.h and update files that include it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
Use snd_dmaengine_dai_dma_data for passing the dma parameters from
clients to the pxa pcm lib. This does no functional change, it's just an
intermedia step to migrate the pxa bits over to dmaengine.
The calculation of dcmd is a transition hack which will be removed again
in a later patch. It's just there to make the transition more readable.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Mark Brown <broonie@linaro.org>
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>
This patch adds BUSIF support for R-Car sound DMAEngine transfer.
The sound data will be transferred via FIFO which can cover blank time
which will happen when DMA channel is switching.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DMAEngine transfer on SSI.
But, it transfers sound data from memory to SSI directly
without using HPBIF at this time.
It will be updated soon
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is using struct rsnd_dai_platform_info
so that indicate sound DAI information (playback/capture SSI ID).
But, SSI settings were also required separately.
Thus, platform settings was very un-understandable.
This patch adds dai_id to SSI
settings, and removed rsnd_dai_platform_info.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
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>
Currently the DAPM code is limited to only setting or clearing a single bit in a
register to power a widget up or down. This patch extends the DAPM code to be
more flexible in that regard and allow widgets to use arbitrary values to be
used to put a widget in either on or off state.
Since the snd_soc_dapm_widget struct already contains a on_val and off_val field
no additional fields need to be added and in fact the invert field can even be
removed. Also the generated code is slightly smaller.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we store for each path which control (if any at all) is associated
with that control. But we are only ever interested in the reverse relationship,
i.e. we want to know all the paths a certain control is associated with. This is
currently implemented by always iterating over all paths. This patch updates the
code to keep a list for each control which contains all the paths that are
associated with that control. This improves the run time of e.g.
soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n
being the number of paths for the card) to O(1).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The 'value' field is really per control and not per widget. Currently it is only
used for virtual MUXes, which only have one control per widget. So in that case
there is not so much of a difference between whether it is stored per widget or
per control. Moving the 'value' field from the widget to the control will allow
us to use it also for cases where we have more than one control per widget. E.g.
for mixers with multiple input controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places.
This patch puts them into a common helper function. Having this encapsulated in
a helper function will also make it more easier to eventually change the data
layout of the kcontrol's private data.
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>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
As 1st protype, this patch adds SSI feature on this driver.
But, it is PIO sound playback support only at this point.
The DMA transfer, and capture feature will be supported in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds ADG feature which controls sound clock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds SCU feature on this driver.
But, it defines SCU style only, does nothing at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
The main difference between Gen1 and Gen2 are
1) register offset, 2) data path
In order to control Gen1/Gen2 by same method,
this patch adds gen.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuits are different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2).
(Actually, there are many difference in Generation1 chips)
Basically, for the future, Renesas R-Car series will use
Gen2 style sound circuit, but driver should care Gen1 also.
The main differences between Gen1 and Gen2 peripheral
are 1) register offset, 2) data path.
This patch adds basic (core) feature for R-Car
series sound driver as prototype
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move HW initialization to separate function to allow using the code without
the v4l parts. This is needed for use in the bttv driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
In order to avoid race conditions the assignment of dapm->update should happen
while card->dapm_mutex is being held. To allow CODEC drivers to run a register
update when using snd_soc_dapm_mux_update_power() or
snd_soc_dapm_mixer_update_power() add a update parameter to these two functions.
The update parameter will be assigned to dapm->update while card->dapm_mutex is
locked.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when updating a control that is shared between multiple widgets the
whole power-up/power-down sequence is being run once for each widget. The
control register is updated during the first run, which means the CODEC internal
routing is also updated for all widgets during this first run. The input and
output paths for each widgets are only updated though during the respective run
for that widget. This leads to a slight inconsistency between the CODEC's
internal state and ASoC's state, which causes non optimal behavior in regard to
click and pop avoidance.
E.g. consider the following setup where two MUXs share the same control.
+------+
A1 ------| |
| MUX1 |----- C1
B1 ------| |
+------+
|
control ---+
|
+------+
A2 ------| |
| MUX2 |----- C2
B2 ------| |
+------+
If the control is updated to switch the MUXs from input A to input B with the
current code the power-up/power-down sequence will look like this:
Run soc_dapm_mux_update_power for MUX1
Power-down A1
Update MUXing
Power-up B1
Run soc_dapm_mux_update_power for MUX2
Power-down A2
(Update MUXing)
Power-up B2
Note that the second 'Update Muxing' is a no-op, since the register was already
updated.
While the preferred order for avoiding pops and clicks should be:
Run soc_dapm_mux_update_power for control
Power-down A1
Power-down A2
Update MUXing
Power-up B1
Power-up B2
This patch changes the behavior to the later by running the updates for all
widgets that the control is attached to at the same time.
The new code is also a bit simpler since callers of
soc_dapm_{mux,muxer}_update_power don't have to loop over each widget anymore
and neither do we need to keep track for which of the kcontrol's widgets the
current update is.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
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Merge tag 'asoc-v3.11-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.11
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
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>
Introduce a new helper function, snd_ctl_sync_vmaster(), which updates
the slave put callbacks forcibly as well as calling the hook. This
will be used in the upcoming patch in HD-audio codec driver for
toggling the mute in vmaster slaves.
Along with the new function, the old snd_ctl_sync_vmaster_hook() is
replaced as a macro calling with the argument hook_only=true.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
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Merge tag 'asoc-v3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.11
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
* for-linus: (635 commits)
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
ALSA: hda - Fix pin configurations for MacBook Air 4,2
ALSA: usb-audio: work around Android accessory firmware bug
ALSA: hda - Headset mic support for three more machines
Linux 3.10-rc6
smp.h: Use local_irq_{save,restore}() in !SMP version of on_each_cpu().
powerpc: Fix missing/delayed calls to irq_work
powerpc: Fix emulation of illegal instructions on PowerNV platform
powerpc: Fix stack overflow crash in resume_kernel when ftracing
snd_pcm_link(): fix a leak...
use can_lookup() instead of direct checks of ->i_op->lookup
move exit_task_namespaces() outside of exit_notify()
fput: task_work_add() can fail if the caller has passed exit_task_work()
xfs: don't shutdown log recovery on validation errors
xfs: ensure btree root split sets blkno correctly
xfs: fix implicit padding in directory and attr CRC formats
xfs: don't emit v5 superblock warnings on write
mei: me: clear interrupts on the resume path
mei: nfc: fix nfc device freeing
mei: init: Flush scheduled work before resetting the device
...
Since commit 85762e71 ("ASoC: dapm: Implement mixer control sharing") the
long_name field of the snd_soc_dapm_path struct is unused. All of the name
handling now happens entirely in dapm_create_or_share_mixmux_kcontrol(). So we
can remove the long_name field from the snd_soc_dapm_path struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the ALC5640 codec driver.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.
Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.
Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently ALSA supports up to 32 card instances when the dynamic minor
is used. While 32 cards are usually big enough for normal use cases,
there are sometimes weird requirements with more card support.
Actually, this limitation, 32, comes from the index option, where you
can pass the bit mask to assign the card. Other than that, we can
actually give more cards up to the minor number limits (currently 256,
which can be extended more, too).
This patch adds a new Kconfig to specify the max card numbers, and
changes a few places to accept more than 32 cards.
The only incompatibility with high card numbers would be the handling
of index option. The index option can be still used to pass the
bitmask for card assignments, but this works only up to 32 slots.
More than 32, no bitmask style option is available but only a single
slot can be specified via index option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ops field of the snd_pcm_substream struct is never modified inside the ALSA
core. Making it const allows drivers to declare their snd_pcm_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices may benefit from being able to start some parts of the widget
power up/down sequence earlier on in the sequence than the point at which
the final power state is committed. Support these by providing events which
are called before any power state changes are done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
Userspace is not meant to have to handle all strange dB ranges,
so add a specification comment.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
Fix typo in printk and comments within various drivers.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some platforms which are half-duplex share the same DMA channel between the
playback and capture stream. Add support for this to the generic dmaengine PCM
driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Use the generic dmaengine PCM driver instead of a custom implemention. There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
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>
Add support for platforms which don't use devicetree yet or have to optionally
support a non-devicetree way to request the DMA channel. The patch adds the
compat_request_channel and compat_filter_fn callbacks to the
snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it
will be used to request the DMA channel. If not dma_request_channel with
compat_filter_fn as the filter function will be used to request the channel.
The patch also exports the snd_dmaengine_pcm_request_chan() function, since
compat platforms will want to use it to request their DMA channel.
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>
This patch adds a generic dmaengine PCM driver. It builds on top of the
dmaengine PCM library and adds the missing pieces like DMA channel management,
buffer management and channel configuration. It will be able to replace the
majority of the existing platform specific dmaengine based PCM drivers.
Devicetree is used to map the DMA channels to the PCM device.
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>
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>
Refactor the dmaengine PCM library to allow the DMA channel to be requested
before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA
channel instead of a filter function and filter parameter as its parameters.
snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows
a dmaengine based PCM driver to request its channels before the substream is
opened.
The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan()
and snd_dmaengine_pcm_close_release_chan(), which have the same signature and
behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new
variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are
updated to use snd_dmaengine_pcm_open_request_chan() and
snd_dmaengine_pcm_close_release_chan().
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>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
This patch adds a common DMA data struct which can be used by DAI drivers to
communicate their DMA configuration requirements to the DMA pcm driver. Having
a common data structure for this allows us to implement common functions on top
of them, which can be used by multiple platforms.
This patch also introduces a new function to initialize certain fields of a
dma_slave_config struct from the common DAI DMA data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
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>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions were initially added to be able to support some oddball dma
drivers, but all users have been updated to deal with the situation without the
help of snd_dmaengine_pcm_{set,get}_data, so these two functions can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Source files shouldn't have the executable bit set.
Signed-off-by: Joe Perches <joe@perches.com>
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>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG
is set leads to frequent bugs, since other similar macros in the kernel
have different behavior. Let's make snd_BUG_ON() act like those macros
so it will stop being accidentally misused.
Signed-off-by: Christine Spang <christine.spang@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a driver for TI's TA5086 6-channel PWM processor.
This chip has a very unusual register layout, specifically because the
registers are of unequal size, and multi-byte registers require bulk
writes to take effect. Regmap does not support these kind of mappings.
Currently, the driver does not touch any of the registers >= 0x20, so
it doesn't matter, because the register map is mapped to an 8-bit array.
In case more features will be added in the future that require access
to higher registers, the entire regmap H/W I/O routines have to be
open-coded.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 497098be ("ASoC: dapm: Remove bodges for no-widget CODECs") removed the
last user of the n_widgets field. Currently it is incremented for each widget
added, but the value is never used, so we can remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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>
Tegra only supports, and always enables, device tree. Remove all runtime
checks for DT support from the driver.
This allows removal of the hard-coded Harmony ASoC mapping table, since
Harmony only boots with DT now.
All board-specific configuration now comes from device tree, so there is
no need to have a platform_data structure. Rework the driver to parse the
device tree directly into struct tegra_wm8903.
Also some slight re-ordering of probe() so that the code more closely
resembles other drivers for easier comparison. Inparticular, the GPIO DT
parsing and initial programming are moved together for each GPIO.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This branch contains changes for OMAP that came in late during the release
staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs (Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and revision
detection, as well as removal of some dead code (CONFIG_32K_TIMER_HZ)
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Merge tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC late OMAP changes from Olof Johansson:
"This branch contains changes for OMAP that came in late during the
release staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs
(Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and
revision detection, as well as removal of some dead code
(CONFIG_32K_TIMER_HZ)"
* tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (34 commits)
ARM: omap2: include linux/errno.h in hwmod_reset
ARM: OMAP2+: fix some omap_device_build() calls that aren't compiled by default
ARM: OMAP4: hwmod data: Enable AESS hwmod device
ARM: OMAP4: hwmod data: Update AESS data with memory bank area
ARM: OMAP4+: AESS: enable internal auto-gating during initial setup
ASoC: TI AESS: add autogating-enable function, callable from architecture code
ARM: OMAP2+: hwmod: add enable_preprogram hook
ARM: OMAP4: clock data: Add missing clkdm association for dpll_usb
ARM: OMAP2+: PM: Fix the dt return condition in pm_late_init()
ARM: OMAP2: am33xx-hwmod: Fix "register offset NULL check" bug
ARM: OMAP2+: AM33xx: hwmod: add missing HWMOD_NO_IDLEST flags
ARM: OMAP: AM33xx hwmod: Add parent-child relationship for PWM subsystem
ARM: OMAP: AM33xx hwmod: Corrects PWM subsystem HWMOD entries
ARM: DTS: AM33XX: Add nodes for OCMC RAM and WKUP-M3
ARM: OMAP2+: AM33XX: Update the hardreset API
ARM: OMAP2+: AM33XX: hwmod: Update the WKUP-M3 hwmod with reset status bit
ARM: OMAP2+: AM33XX: hwmod: Fixup cpgmac0 hwmod entry
ARM: OMAP2+: AM33XX: hwmod: Update TPTC0 hwmod with the right flags
ARM: OMAP2+: AM33XX: hwmod: Register OCMC RAM hwmod
ARM: OMAP2+: AM33XX: CM/PRM: Use __ASSEMBLER__ macros in header files
...
Currently if a path loops back on itself we correctly skip over it to
avoid going into an infinite loop but this causes us to ignore the need
to power up the path as we don't count the loop for the purposes of
counting inputs and outputs. This means that internal loopbacks within a
device that have powered devices on them won't be powered up.
Fix this by treating any path that is currently in the process of being
recursed as having a single input or output so that it is counted for
the purposes of power decisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
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Merge tag 'asoc-3.9-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.9
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
This patch adds support for the Dialog DA7213 audio codec.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
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Merge tag 'asoc-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.9
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
Add a basic header file for the TI AESS IP block, located in the OMAP4
Audio Back-End subsystem.
Currently, this header file only contains a function to enable the
AESS internal clock auto-gating. This will be used by a subsequent
patch to ensure that the AESS won't block the entire chip
low-power-idle mode. We wish to be able to place the AESS into idle
even when no AESS driver has been compiled in.
Signed-off-by: Paul Walmsley <paul@pwsan.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Péter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: Jerry Wong <jerry.wong@maximintegrated.com>
Tested-by: Matthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: Ralph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some audio drivers are calling snd_dma_continuous_data(GFP_KERNEL)
which makes "sparse" give a warning:
$ make C=2 M=sound/usb modules
...
sound/usb/6fire/pcm.c:625:25: warning: cast from restricted gfp_t
sound/usb/caiaq/audio.c:845:41: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:997:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:1001:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:774:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:778:54: warning: cast from restricted gfp_t
Add __force to the cast to silence the warning.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8).
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>
Current simple-card driver calls asoc_simple_card_dai_init()
if platform had a asoc_simple_card_dai_init pointer.
And, this initialization function works only
when platform has an applicable initial value for each dai settings.
And basically, almost all sound card requires certain initialization.
This means that almost all platform has initialization settings,
and driver do nothing if it doesn't have settings.
And additionally, current simple-card supports sysclk settings but it was
only for codec. In order to abolish deviation between cpu and codec,
and in order to simplify processing,
this patch adds asoc_simple_dai, and removed pointless
struct asoc_simple_dai_init_info which was trigger of
calling asoc_simple_card_dai_init().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All MXS users have been converted to device tree and the board files have been
removed.
No need to keep platform data in the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Empty files can get deleted by the patch program, so remove empty Kbuild
files and their links from the parent Kbuilds.
Signed-off-by: David Howells <dhowells@redhat.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
FSI driver's flag usage was changed/removed by
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
ab6f6d8521
(ASoC: fsi: add master clock control functions)
And unused flags had been removed on FSI driver,
but the definition had been kept to avoid compile error.
It is possible to cleanup sh_fsi.h now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ab6f6d8521
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.
One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.
This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.
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>
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
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Merge tag 'asoc-3.8p1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
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>
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
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Merge tag 'asoc-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.8
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Yet again like previous two commits, drop the old hwdep user-space
firmware code from vx driver (snd-vxpocket and snd-vx222).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CS4271 has a feature to sync its analog mute flags, so one mute
circuitry can be used for both channels.
Give users access to this feature with a new DT property and a flag in
the platform data.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Pull media updates from Mauro Carvalho Chehab:
"The first part of the media updates for Kernel 3.7.
This series contain:
- A major tree renaming patch series: now, drivers are organized
internally by their used bus, instead of by V4L2 and/or DVB API,
providing a cleaner driver location for hybrid drivers that
implement both APIs, and allowing to cleanup the Kconfig items and
make them more intuitive for the end user;
- Media Kernel developers are typically very lazy with their duties
of keeping the MAINTAINERS entries for their drivers updated. As
now the tree is more organized, we're doing an effort to add/update
those entries for the drivers that aren't currently orphan;
- Several DVB USB drivers got moved to a new DVB USB v2 core; the new
core fixes several bugs (as the existing one that got bitroted).
Now, suspend/resume finally started to work fine (at least with
some devices - we should expect more work with regards to it);
- added multistream support for DVB-T2, and unified the API for
DVB-S2 and ISDB-S. Backward binary support is preserved;
- as usual, a few new drivers, some V4L2 core improvements and lots
of drivers improvements and fixes.
There are some points to notice on this series:
1) you should expect a trivial merge conflict on your tree, with the
removal of Documentation/feature-removal-schedule.txt: this series
would be adding two additional entries there. I opted to not
rebase it due to this recent change;
2) With regards to the PCTV 520e udev-related breakage, I opted to
fix it in a way that the patches can be backported to 3.5 even
without your firmware fix patch. This way, Greg doesn't need to
rush backporting your patch (as there are still the firmware cache
and firmware path customization issues to be addressed there).
I'll send later a patch (likely after the end of the merge window)
reverting the rest of the DRX-K async firmware request, fully
restoring its original behaviour to allow media drivers to
initialize everything serialized as before for 3.7 and upper.
3) I'm planning to work on this weekend to test the DMABUF patches
for V4L2. The patches are on my queue for several Kernel cycles,
but, up to now, there is/was no way to test the series locally.
I have some concerns about this particular changeset with regards
to security issues, and with regards to the replacement of the old
VIDIOC_OVERLAY ioctl's that is broken on modern systems, due to
GPU drivers change. The Overlay API allows direct PCI2PCI
transfers from a media capture card into the GPU framebuffer, but
its API is crappy. Also, the only existing X11 driver that
implements it requires a XV extension that is not available
anymore on modern drivers. The DMABUF can do the same thing, but
with it is promising to be a properly-designed API. If I can
successfully test this series and be happy with it, I should be
asking you to pull them next week."
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (717 commits)
em28xx: regression fix: use DRX-K sync firmware requests on em28xx
drxk: allow loading firmware synchrousnously
em28xx: Make all em28xx extensions to be initialized asynchronously
[media] tda18271: properly report read errors in tda18271_get_id
[media] tda18271: delay IR & RF calibration until init() if delay_cal is set
[media] MAINTAINERS: add Michael Krufky as tda827x maintainer
[media] MAINTAINERS: add Michael Krufky as tda8290 maintainer
[media] MAINTAINERS: add Michael Krufky as cxusb maintainer
[media] MAINTAINERS: add Michael Krufky as lg2160 maintainer
[media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer
[media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer
[media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer
[media] MAINTAINERS: add Michael Krufky as tda18271 maintainer
[media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap
[media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend()
[media] exynos-gsc: Add missing static storage class specifiers
[media] exynos-gsc: Remove <linux/version.h> header file inclusion
[media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs()
[media] s5p-tv: Fix potential NULL pointer dereference error
[media] s5k6aa: Fix possible NULL pointer dereference
...
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Additional updates for v3.7
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
Convert #include "..." to #include <path/...> in kernel system headers.
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This patch adds support for Dialog semiconductor's DA9055 audio codec.
This has been tested on DA9055 EVB with Samsung SMDK6410 board.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <david.chen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow regulators managed via DAPM to make use of the bypass support that
has recently been added to the regulator API by setting a flag
SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will
be put into bypass mode before being disabled, allowing the regulator to
fall into bypass mode if it can't be disabled due to other users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing struct snd_dma_buffer pointer instead, so that they work no
matter whether real SG buffer is used or not.
This is a preliminary work for the HD-audio DSP loader code.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.
It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
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>
Add support for tuning AM (on devices with the necessary additional
hardware components), and advertise the available bands using the new
VIDIOC_ENUM_FREQ_BANDS ioctl.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
For following the standard, define more channel map positions and
shuffle the items a bit:
- As both PulseAudio and gstreamer define MONO channel position
explicitly, we should follow that, too. The mono streams point to
this channel position unless they are explicitly assigned to certain
channel positions.
- Top-front-* and Top-rear-* positions are added, carried from
PulseAudio's definitions.
- Move NA and MONO definitions at the top of table right after
UNKNOWN, since these are more abstract in comparison with other
practical positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be used to enable additional control of the regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
There is already a set of channel position definitions in alsa-lib
mixer.h, and it'd be more practical to keep the same order for the
PCM channel map, too. The value is shifted with 1 to keep zero for
UNKNOWN.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC650 has a channel swap option between surround and CLFE channels,
so we need to tweak the channel maps dynamically depending on the
register bit.
Now struct snd_ac97 can contain chmap pointers for playback and
capture. The driver may store these and let ac97 driver changing the
channel mapping dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the basic data types for the standard channel
mapping API handling.
- The definitions of the channel positions and the new TLV types are
added in sound/asound.h and sound/tlv.h, so that they can be
referred from user-space.
- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
control elements representing the channel maps for each PCM
(sub)stream.
- Some standard pre-defined channel maps are provided for
convenience.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since bypass paths aren't part of DAPM streams and we may not have any
DAPM streams there may not be anything that triggers a DAPM sync for
them. Mark all input and output widgets as dirty and then sync to do so
at the end of suspend and resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove
current PGA control handling").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the Tegra+WM8903 ASoC platform data header out of
arch/arm/mach-tegra, as a pre-requisite of single zImage.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the main ALSA version number from the kernel ALSA driver.
The ALSA driver package release diverges from the upstream. This may
confuse users to see the same ALSA version for many kernel releases
and this version lost it's original purpose and connection.
The "ioctl" APIs have own version numbers, so the user space may check
for specific API changes only.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM0010 is a compact digital signal processor that has been
highly optimised for low-power audio applications. Extensive memory
resources and core optimisation allow the device to manage all audio
processing algorithms efficiently and autonomously, while the host
processor sleeps or performs other tasks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes the analogue circuitry connected to the microphone needs some
time to settle after power up. Allow systems to configure this delay in
the platform data, the driver will then insert the required delay during
power up of paths that involve the microphone.
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>
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
Implement suspend/resume support for AD1816 chips.
Tested with Terratec SoundSystem Base-1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card_ad1816a is only set but the values are never used then.
Removing it allows struct snd_card's private_data to be used for
struct snd_ad1816a, simplifying the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix kernel-doc warning in <sound/pcm.h> and add function name to make
the kernel-doc notation complete.
Warning(include/sound/pcm.h:1081): No description found for parameter 'substream'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move snd_legacy_find_free_ioport() function back to initval.h as it is used
by two drivers.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes that have been found recently.
Most of the commits are regression fixes in HD-audio and some other
random drivers.
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes that have been found recently. Most of
the commits are regression fixes in HD-audio and some other random
drivers."
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: snd-usb: fix clock source validity index
ALSA: hda - Fix mute-LED GPIO initialization for IDT codecs
ALSA: hda - Add descriptions for missing IDT 92HD83x models
ALSA: hda - Fix polarity of mute LED on HP Mini 210
ALSA: es1688 - freeup resources on init failure
ALSA: hda - Workaround for silent output on VAIO Z with ALC889
ALSA: hda - Fix WARNING from HDMI/DP parser
ALSA: hda - Detach from converter at closing in patch_hdmi.c
ALSA: hda - Fix mute-LED GPIO setup for HP Mini 210
ALSA: mpu401: Fix missing initialization of irq field
ALSA: hda - Fix invalid D3 of headphone DAC on VT202x codecs
Pull second set of media updates from Mauro Carvalho Chehab:
- radio API: add support to work with radio frequency bands
- new AM/FM radio drivers: radio-shark, radio-shark2
- new Remote Controller USB driver: iguanair
- conversion of several drivers to the v4l2 core control framework
- new board additions at existing drivers
- the remaining (and vast majority of the patches) are due to
drivers/DocBook fixes/cleanups.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (154 commits)
[media] radio-tea5777: use library for 64bits div
[media] tlg2300: Declare MODULE_FIRMWARE usage
[media] lgs8gxx: Declare MODULE_FIRMWARE usage
[media] xc5000: Add MODULE_FIRMWARE statements
[media] s2255drv: Add MODULE_FIRMWARE statement
[media] dib8000: move dereference after check for NULL
[media] Documentation: Update cardlists
[media] bttv: add support for Aposonic W-DVR
[media] cx25821: Remove bad strcpy to read-only char*
[media] pms.c: remove duplicated include
[media] smiapp-core.c: remove duplicated include
[media] via-camera: pass correct format settings to sensor
[media] rtl2832.c: minor cleanup
[media] Add support for the IguanaWorks USB IR Transceiver
[media] Minor cleanups for MCE USB
[media] drivers/media/dvb/siano/smscoreapi.c: use list_for_each_entry
[media] Use a named union in struct v4l2_ioctl_info
[media] mceusb: Add Twisted Melon USB IDs
[media] staging/media/solo6x10: use module_pci_driver macro
[media] staging/media/dt3155v4l: use module_pci_driver macro
...
Conflicts:
Documentation/feature-removal-schedule.txt
Some devices which use the tea575x tuner chip don't allow direct control
over the IO pins, and thus cannot mute the audio output.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Some devices which use the tea575x tuner chip don't allow bit banging the
lines, instead they offer a method to directly set / get the contents of the
25 bit shift-register in the chip. Notably the Griffin radioSHARK USB radio
receiver does this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
Add a DECLARE_TLV_DB_RANGE() macro so that dB range information
can be specified without having to count the items manually for
TLV_DB_RANGE_HEAD().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the DECLARE_TLV_CONTAINER() macro to allow having static
TLVs containing more than one item.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add helper macros with a little bit of preprocessor magic to
automatically compute the length of a TLV item. This lets us avoid
having to compute this by hand, and will allow to use items that do
not use a fixed length.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're now relying on DAPM for things like enabling clocks when we
reparent the clocks for widgets we need to either use conditional routes
(which are expensive) or remove routes at runtime. Add a route removal
API to support this use case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
They aren't modified by the core so the drivers can declare them const.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a bit clean up of public sound header directory.
Some header files in include/sound aren't really necessary to be
located there but can be moved to their local directories gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull media fixes from Mauro Carvalho Chehab.
Trivial conflict due to new USB HID ID's being added next to each other
(Baanto vs Axentia).
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (44 commits)
[media] smia: Fix compile failures
[media] Fix VIDIOC_DQEVENT docbook entry
[media] s5p-fimc: Fix control creation function
[media] s5p-mfc: Fix checkpatch error in s5p_mfc_shm.h file
[media] s5p-mfc: Fix setting controls
[media] v4l/s5p-mfc: added image size align in VIDIOC_TRY_FMT
[media] v4l/s5p-mfc: corrected encoder v4l control definitions
[media] v4l: mem2mem_testdev: Fix race conditions in driver
[media] s5p-mfc: Bug fix of timestamp/timecode copy mechanism
[media] cxd2820r: Fix an incorrect modulation type bitmask
[media] em28xx: Show a warning if the board does not support remote controls
[media] em28xx: Add remote control support for Terratec's Cinergy HTC Stick HD
[media] USB: Staging: media: lirc: initialize spinlocks before usage
[media] Revert "[media] media: mx2_camera: Fix mbus format handling"
[media] bw-qcam: driver and pixfmt documentation fixes
[media] cx88: fix firmware load on big-endian systems
[media] cx18: support big-endian systems
[media] ivtv: fix support for big-endian systems
[media] tuner-core: return the frequency range of the correct tuner
[media] v4l2-dev.c: fix g_parm regression in determine_valid_ioctls()
...
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>
This patch implements the spdif IN driver for ST peripheral
Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support for the SPEAr ASoC pcm layer in ASoC
framework. The pcm layer uses common snd_dmaengine framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support for synopsys I2S controller as per the ASoC
framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch addresses the issue by
implementing support for querying the current stream position directly from the
dmaengine driver. Since not all dmaengine drivers support reporting the stream
position yet the old period counting implementation is kept for now.
Furthermore the new mechanism allows to report the stream position with a
sub-period granularity, given that the dmaengine driver supports this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch renames the current implementation
and documents its shortcomings and that it should not be used anymore in new
drivers.
The next patch will introduce a new snd_dmaengine_pcm_pointer which will be
implemented based on querying the current stream position from the dma device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is essentially the reverse of snd_pcm_rate_to_rate_bit().
This is generally useful as the Compress API uses the rate bit
directly and it helps to be able to map back to the actual sample
rate.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before this patch the owner field of the /dev/radio# device fops was set to
the snd-tea575x-tuner module itself. Meaning that the module which was using
it could be rmmod-ed while the device is open, and then BAD things happen.
I know, as I found out the hard way :)
Note that there is no need to also somehow increase the refcount of the
snd-tea575x-tuner module itself, since any drivers using it will have
symbolic references to it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.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>
Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds a function getting the stream-name as a string for
a specific stream.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
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Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute updates
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
SupherH FSI2 can use special data transfer,
but it depends on CPU-FSI2 connection style.
We can use 16bit data stream mode if it was valid connection,
and it is required for 16bit data DMA transfer / SPDIF sound output.
We can use 24bit data transfer if it was invalid connection.
We can select connection type if CPU is SH7372,
and it is always valid connection if latest SuperH.
This patch adds new bus_option and fsi_bus_setup()
for supporting these feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.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>
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.
This patchs adds/changes the following :-
o Adds DPCM runtime update core.
o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
to return if a change has occured rather than 0. No other users check
atm.
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>
In preparation for ASoC DSP support.
Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.
This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
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>
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.
This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Supports larger register maps, not using unsigned ints for the full 32
bit as we rely on checking for negative registers.
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>
There are no users any more and new drivers should be using supply widgets
which fully replace it anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
This change adds the logic to support using the jack detect mechanism built
in to the codec to detect both when a jack was inserted and what type of
jack is present.
This change also supports the use of an external mechanism for headphone
detection. If this mechanism exists, when the max98095_jack_detect function
is called, the hp_jack is simply passed NULL.
This change supports both simple headphones, powered headphones, microphones
and headsets with both headphones and a mic.
Signed-off-by: Rhyland Klein <rklein@nvidia.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>
Currently DAPM widgets use the private data for their regulator.
Add a regulator * for widgets to use instead of private data.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename SND_SOC_DAPM_CLASS_PCM to SND_SOC_DAPM_CLASS_RUNTIME to
better match the usage and align with card mutex too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
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>
Currently stream events are only perfomed on codec stream widgets only.
There is now a need to be able to perform stream events on platform
widgets too.
e.g. we have the ABE platform driver with several DAI links
to dummy codecs. We need to be able to perform stream events on any
of the dummy codec DAI links.
This patch also removes the snd_soc_dai * parameter since it's already
contained within the rtd * parameter.
Finally makle stream event return void since no one checks it anyway.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add platform driver support for CPU DAI DAPM widgets.
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>
"[RFC PATCH 0/2] audit of linux/device.h users in include/*"
https://lkml.org/lkml/2012/3/4/159
--
Nearly every subsystem has some kind of header with a proto like:
void foo(struct device *dev);
and yet there is no reason for most of these guys to care about the
sub fields within the device struct. This allows us to significantly
reduce the scope of headers including headers. For this instance, a
reduction of about 40% is achieved by replacing the include with the
simple fact that the device is some kind of a struct.
Unlike the much larger module.h cleanup, this one is simply two
commits. One to fix the implicit <linux/device.h> users, and then
one to delete the device.h includes from the linux/include/ dir
wherever possible.
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Merge tag 'device-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux
Pull <linux/device.h> avoidance patches from Paul Gortmaker:
"Nearly every subsystem has some kind of header with a proto like:
void foo(struct device *dev);
and yet there is no reason for most of these guys to care about the
sub fields within the device struct. This allows us to significantly
reduce the scope of headers including headers. For this instance, a
reduction of about 40% is achieved by replacing the include with the
simple fact that the device is some kind of a struct.
Unlike the much larger module.h cleanup, this one is simply two
commits. One to fix the implicit <linux/device.h> users, and then one
to delete the device.h includes from the linux/include/ dir wherever
possible."
* tag 'device-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux:
device.h: audit and cleanup users in main include dir
device.h: cleanup users outside of linux/include (C files)
Merge second batch of patches from Andrew Morton:
- various misc things
- core kernel changes to prctl, exit, exec, init, etc.
- kernel/watchdog.c updates
- get_maintainer
- MAINTAINERS
- the backlight driver queue
- core bitops code cleanups
- the led driver queue
- some core prio_tree work
- checkpatch udpates
- largeish crc32 update
- a new poll() feature for the v4l guys
- the rtc driver queue
- fatfs
- ptrace
- signals
- kmod/usermodehelper updates
- coredump
- procfs updates
* emailed from Andrew Morton <akpm@linux-foundation.org>: (141 commits)
seq_file: add seq_set_overflow(), seq_overflow()
proc-ns: use d_set_d_op() API to set dentry ops in proc_ns_instantiate().
procfs: speed up /proc/pid/stat, statm
procfs: add num_to_str() to speed up /proc/stat
proc: speed up /proc/stat handling
fs/proc/kcore.c: make get_sparsemem_vmemmap_info() static
coredump: add VM_NODUMP, MADV_NODUMP, MADV_CLEAR_NODUMP
coredump: remove VM_ALWAYSDUMP flag
kmod: make __request_module() killable
kmod: introduce call_modprobe() helper
usermodehelper: ____call_usermodehelper() doesn't need do_exit()
usermodehelper: kill umh_wait, renumber UMH_* constants
usermodehelper: implement UMH_KILLABLE
usermodehelper: introduce umh_complete(sub_info)
usermodehelper: use UMH_WAIT_PROC consistently
signal: zap_pid_ns_processes: s/SEND_SIG_NOINFO/SEND_SIG_FORCED/
signal: oom_kill_task: use SEND_SIG_FORCED instead of force_sig()
signal: cosmetic, s/from_ancestor_ns/force/ in prepare_signal() paths
signal: give SEND_SIG_FORCED more power to beat SIGNAL_UNKILLABLE
Hexagon: use set_current_blocked() and block_sigmask()
...
This addresses some header check warnings. DRM headers which include
"drm.h" have been excluded, as they indirectly include types.h.
Signed-off-by: Bobby Powers <bobbypowers@gmail.com>
Cc: Chris Ball <cjb@laptop.org>
Cc: Dave Airlie <airlied@linux.ie>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull media updates from Mauro Carvalho Chehab:
- V4L2 API additions to better support JPEG compression control
- media API additions to properly support MPEG decoders
- V4L2 API additions for image crop/scaling
- a few other V4L2 API DocBook fixes/improvements
- two new DVB frontend drivers: m88rs2000 and rtl2830
- two new DVB drivers: az6007 and rtl28xxu
- a framework for ISA drivers, that removed lots of common code found
at the ISA radio drivers
- a new FM transmitter driver (radio-keene)
- a GPIO-based IR receiver driver
- a new sensor driver: mt9m032
- some new video drivers: adv7183, blackfin, mx2_emmaprp, sii9234_drv,
vs6624
- several new board additions, driver fixes, improvements and cleanups.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (295 commits)
[media] update CARDLIST.em28xx
[media] partially reverts changeset fa5527c
[media] stb0899: fix the limits for signal strength values
[media] em28xx: support for 2304:0242 PCTV QuatroStick (510e)
[media] em28xx: support for 2013:0251 PCTV QuatroStick nano (520e)
[media] -EINVAL -> -ENOTTY
[media] gspca - sn9c20x: Cleanup source
[media] gspca - sn9c20x: Simplify register write for capture start/stop
[media] gspca - sn9c20x: Add automatic JPEG compression mechanism
[media] gspca - sn9c20x: Greater delay in case of sensor no response
[media] gspca - sn9c20x: Optimize the code of write sequences
[media] gspca - sn9c20x: Add the JPEG compression quality control
[media] gspca - sn9c20x: Add a delay after Omnivision sensor reset
[media] gspca - sn9c20x: Propagate USB errors to higher level
[media] gspca - sn9c20x: Use the new video control mechanism
[media] gspca - sn9c20x: Fix loss of frame start
[media] gspca - zc3xx: Lack of register 08 value for sensor cs2102k
[media] gspca - ov534_9: Add brightness to OmniVision 5621 sensor
[media] gspca - zc3xx: Add V4L2_CID_JPEG_COMPRESSION_QUALITY control support
[media] pvrusb2: fix 7MHz & 8MHz DVB-T tuner support for HVR1900 rev D1F5
...
* tag 'v3.3': (1646 commits)
Linux 3.3
Don't limit non-nested epoll paths
netfilter: ctnetlink: fix race between delete and timeout expiration
ipv6: Don't dev_hold(dev) in ip6_mc_find_dev_rcu.
nilfs2: fix NULL pointer dereference in nilfs_load_super_block()
nilfs2: clamp ns_r_segments_percentage to [1, 99]
afs: Remote abort can cause BUG in rxrpc code
afs: Read of file returns EBADMSG
C6X: remove dead code from entry.S
wimax/i2400m: fix erroneous NETDEV_TX_BUSY use
net/hyperv: fix erroneous NETDEV_TX_BUSY use
net/usbnet: reserve headroom on rx skbs
bnx2x: fix memory leak in bnx2x_init_firmware()
bnx2x: fix a crash on corrupt firmware file
sch_sfq: revert dont put new flow at the end of flows
ipv6: fix icmp6_dst_alloc()
MAINTAINERS: Add Serge as maintainer of capabilities
drivers/video/backlight/s6e63m0.c: fix corruption storing gamma mode
MAINTAINERS: add entry for exynos mipi display drivers
MAINTAINERS: fix link to Gustavo Padovans tree
...
The tea575x-tuner module has been updated to use the latest V4L2 framework
functionality. This also required changes in the drivers that rely on it.
The tea575x changes are:
- The drivers must provide a v4l2_device struct to the tea module.
- The radio_nr module parameter must be part of the actual radio driver,
and not of the tea module.
- Changed the frequency range to the normal 76-108 MHz range instead of
50-150.
- Add hardware frequency seek support.
- Fix broken rxsubchans/audmode handling.
- The application can now select between stereo and mono.
- Support polling for control events.
- Add V4L2 priority handling.
And radio-sf16fmr2.c now uses the isa bus kernel framework.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Thanks-to: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
The <linux/device.h> header includes a lot of stuff, and
it in turn gets a lot of use just for the basic "struct device"
which appears so often.
Clean up the users as follows:
1) For those headers only needing "struct device" as a pointer
in fcn args, replace the include with exactly that.
2) For headers not really using anything from device.h, simply
delete the include altogether.
3) For headers relying on getting device.h implicitly before
being included themselves, now explicitly include device.h
4) For files in which doing #1 or #2 uncovers an implicit
dependency on some other header, fix by explicitly adding
the required header(s).
Any C files that were implicitly relying on device.h to be
present have already been dealt with in advance.
Total removals from #1 and #2: 51. Total additions coming
from #3: 9. Total other implicit dependencies from #4: 7.
As of 3.3-rc1, there were 110, so a net removal of 42 gives
about a 38% reduction in device.h presence in include/*
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
Allows the constraint lists to be declared const by drivers which seems
reasonable; there's plenty of other constification we could do if we were
being complete but this was easy and quick.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a hook to vmaster control to be called at each time
when the master value is changed. It'd be handy for an additional
mute LED control following the Master switch, for example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
There's now core code which falls back to global CODEC operations for
DAI calls that needs to be able to tell if it's dealing with a CPU or
CODEC DAI and given the small number of DAIs in a typical system and
overall memory usage pattern saving a pointer per DAI is really not
worth the effort.
Reported-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds a set of functions which are intended to be used when
implementing a dmaengine based sound PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is intended to facilitate the merge of the two jack detection
mechanisms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This patch supports DMAEngine to FSI driver.
It supports only Tx case at this point.
If platform/cpu doesn't support DMAEngine, FSI driver will
use PIO transfer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow us to do smarter things with DAI links create DAPM
widgets which directly represent the DAIs in the DAPM graph. These are
automatically created from the DAIs as we probe the card with references
held in both directions between the widget and the DAI.
The widgets are not made available for direct instantiation by drivers,
they are created automatically from the DAIs. Drivers should be updated
to create stream routes using DAPM maps rather than by annotating AIF
and DAC widgets with streams.
In order to ease transition to this model from existing drivers we
automatically create DAPM routes between the DAI widgets and the existing
stream widgets which are started and stopped by the DAI widgets, though
the old stream handling mechanism is still in place. This also has the
nice effect of removing non-DAPM devices as any device with a DAI
acquires a widget automatically which will allow future simplifications
to the core DAPM logic.
The intention is that in future the AIF and DAI widgets will gain the
ability to interact such that we are able to manage activity on
individual channels independantly rather than powering up and down the
entire AIF as we do currently.
Currently we only generate these for CODECs, mostly as I have no systems
with non-CODEC DAPM to integrate with. It should be a simple matter of
programming to add the additional hookup for these.
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>
In order to allow us to do something smarter than iterate through widgets
doing strcmp() to work out what to power up for stream events change the
interface used to generate them to be based on the combination of a DAI
and a stream direction rather than just a simple string identifying the
stream.
At some point we'll probably want a set of channels too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Everything now uses snd_soc_dapm_new_controls() instead so we don't need
to make it part of the external API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We never modify it and this lets us use a const string as the name without
warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
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>
.. the number of the half-beast?
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Merge tag 'v3.3-rc3' as we've got several bugfixes in there which are
colliding annoyingly with development.
Linux 3.3-rc3
.. the number of the half-beast?
Conflicts:
sound/soc/codecs/wm5100.c
sound/soc/codecs/wm8994.c
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>
The new ASoC dynamic PCM core needs to create PCMs and substreams that are
for use by internal ASoC drivers only and not visible to userspace for
direct IO. These new PCMs are similar to regular PCMs expect they have no
device nodes or procfs entries. The ASoC component drivers use them in exactly
the same way as regular PCMs for PCM and DAI operations.
The intention is that a dynamic PCM based driver will register both regular
PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace
however the internal PCMs will be used by the driver to route digital audio
through numerous back end DAI links (with potentially a DSP providing different
hw_params, DAI formats based on the regular front end PCM params) to devices
like CODECs, MODEMs, Bluetooth, FM, DMICs, etc
This patch adds a new snd_pcm_new_internal() API call to create the internal PCM
without device nodes or procfs. It also adds adds a new internal flag to snd_pcm.
[fixed minor coding-style issues by tiwai]
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow for the operation of custom mixer and mux DAPM widgets that can call
snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly
after updating their status. This is useful with complex DAPM Mixer operations
where we need to do additional work in addition to setting a few mixer register
bits.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
Current FSI got each PortA/B parameter by porta_flags/portb_flags from platform.
And .set_rate function was shared for PortA/B.
This structure was not readable and not flexible.
This patch adds sh_fsi_port_info, and its own settings was added on each platform.
it is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a driver supporting the volume control and the mute pin. Shdn pin
and DAPM are not taken care of yet.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes below build warning when CONFIG_PCI is not set.
CC sound/sound_core.o
In file included from sound/sound_core.c:15:
include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list
include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM2200 is a low power mobile CODEC with enhanced Wolfson myZone
Ambient Noise Cancellation (ANC) intended for mobile telephony
applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modern devices allow systems to enable and disable individual supplies on
the device, allowing additional power saving by switching off regulators
which power portions of the device which are not currently in use. Add a
new SND_SOC_DAPM_REGULATOR_SUPPLY widget type factoring out the code for
managing such widgets from individual drivers.
The widget name will be used as the supply name when requesting the
regulator from the regulator API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the hardware is configured with one or both of the IN4 inputs used
for DC measurement (with no DC blocking capacitor connected) then we can
improve power consumption slightly in idle modes by applying a register
write sequence. Provide platform data to enable this, implemented using
a regmap patch.
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>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
ALSA: usb-audio: add Yamaha MOX6/MOX8 support
ALSA: virtuoso: add S/PDIF input support for all Xonars
ALSA: ice1724 - Support for ooAoo SQ210a
ALSA: ice1724 - Allow card info based on model only
ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
ALSA: hdspm - Provide unique driver id based on card serial
ASoC: Dynamically allocate the rtd device for a non-empty release()
ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
ALSA: hda/cirrus - support for iMac12,2 model
ASoC: cx20442: add bias control over a platform provided regulator
ALSA: usb-audio - Avoid flood of frame-active debug messages
ALSA: snd-usb-us122l: Delete calls to preempt_disable
mfd: Put WM8994 into cache only mode when suspending
...
Fix up trivial conflicts in:
- arch/arm/mach-s3c64xx/mach-crag6410.c:
renamed speyside_wm8962 to tobermory, added littlemill right
next to it
- drivers/base/regmap/{regcache.c,regmap.c}:
duplicate diff that had already come in with other changes in
the regmap tree
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>
Again, a lot of platforms have changes in here: pxa, samsung, omap,
at91, imx, ...
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Merge tag 'drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Driver specific changes
Again, a lot of platforms have changes in here: pxa, samsung, omap,
at91, imx, ...
* tag 'drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (54 commits)
ARM: sa1100: clean up of the clock support
ARM: pxa: add dummy clock for sa1100-rtc
RTC: sa1100: support sa1100, pxa and mmp soc families
RTC: sa1100: remove redundant code of setting alarm
RTC: sa1100: Clean out ost register
Input: zylonite-wm97xx - replace IRQ_GPIO() with gpio_to_irq()
pcmcia: pxa: replace IRQ_GPIO() with gpio_to_irq()
ARM: EXYNOS: Modified files for SPI consolidation work
ARM: S5P64X0: Enable SDHCI support
ARM: S5P64X0: Add lookup of sdhci-s3c clocks using generic names
ARM: S5P64X0: Add HSMMC setup for host Controller
ARM: EXYNOS: Add USB OHCI support to ORIGEN board
USB: Add Samsung Exynos OHCI diver
ARM: EXYNOS: Add USB OHCI support to SMDKV310 board
ARM: EXYNOS: Add USB OHCI device
net: macb: fix build break with !CONFIG_OF
i2c: tegra: Support DVC controller in device tree
i2c: tegra: Add __devinit/exit to probe/remove
net/at91_ether: use gpio_is_valid for phy IRQ line
ARM: at91/net: add macb ethernet controller in 9g45/9g20 DT
...
Export compress_offload.h and compress_params.h for userland to use
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the header files for ioctl definitions and header file for
driver APIs for lower level device drivers to use
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the various definations used to define the encoder
and decoder parameters
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the minor numbers 2 and 3 for audio compressed offload devices.
Also add support for these devices in core
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
wm8903_platform_data.gpio_cfg[] was intended to be interpreted as follows:
0: Don't touch this GPIO's configuration register
1..7fff: Write that value to the GPIO's configuration register
8000: Write zero to the GPIO's configuration register
other: Undefined (invalid)
The rationale is that platform data is usually global data, and a value of
zero means that the field wasn't explicitly set to anything (e.g. because
the field was new to the pdata type, and existing users weren't update to
initialize it) and hence the value zero should be ignored. 0x8000 is an
explicit way to get 0 in the register.
The code worked this way until commit 7cfe561 "ASoC: wm8903: Expose GPIOs
through gpiolib", where the behaviour was changed due to my lack of
awareness of the above rationale.
This patch reverts to the intended behaviour, and updates all in-tree users
to use the correct scheme. This also makes WM8903 consistent with other
devices that use a similar scheme.
WM8903_GPIO_NO_CONFIG is also renamed to WM8903_GPIO_CONFIG_ZERO so that
its name accurately reflects its purpose.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Cc: Olof Johansson <olof@lixom.net>
Cc: Colin Cross <ccross@android.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 signal generator behaves as an input would but is not considered for
any of the special behaviour associated with external input pins. This
is especially useful when automatically working out not connected widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Add master_mode and master_id in platfrom_data since it's board
specific and board knows it.
Then we can remove the function pointer in platfrom_data to make
the driver more devicetree friendly.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Current fsi-ak4642 was using id_entry name in order to specify
FSI port and ak464x codec.
But it was no sense, no flexibility.
Platform can specify FSI/ak464x pair by this patch.
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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>
Now move the helper function for creating and reporting the jack-detection
to the common place. The driver that needs this functionality should
select CONFIG_SND_KCTL_JACK kconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sta32x resets and loses all configuration during ESD test.
Work around by polling the CONFA register once a second
and restore all coeffcients and registers when CONFA
changes unexpectedly.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a structure for platform specific configuration and use it,
thereby removing a few FIXMEs which marked hard-coded values.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
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>
* 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux: (230 commits)
Revert "tracing: Include module.h in define_trace.h"
irq: don't put module.h into irq.h for tracking irqgen modules.
bluetooth: macroize two small inlines to avoid module.h
ip_vs.h: fix implicit use of module_get/module_put from module.h
nf_conntrack.h: fix up fallout from implicit moduleparam.h presence
include: replace linux/module.h with "struct module" wherever possible
include: convert various register fcns to macros to avoid include chaining
crypto.h: remove unused crypto_tfm_alg_modname() inline
uwb.h: fix implicit use of asm/page.h for PAGE_SIZE
pm_runtime.h: explicitly requires notifier.h
linux/dmaengine.h: fix implicit use of bitmap.h and asm/page.h
miscdevice.h: fix up implicit use of lists and types
stop_machine.h: fix implicit use of smp.h for smp_processor_id
of: fix implicit use of errno.h in include/linux/of.h
of_platform.h: delete needless include <linux/module.h>
acpi: remove module.h include from platform/aclinux.h
miscdevice.h: delete unnecessary inclusion of module.h
device_cgroup.h: delete needless include <linux/module.h>
net: sch_generic remove redundant use of <linux/module.h>
net: inet_timewait_sock doesnt need <linux/module.h>
...
Fix up trivial conflicts (other header files, and removal of the ab3550 mfd driver) in
- drivers/media/dvb/frontends/dibx000_common.c
- drivers/media/video/{mt9m111.c,ov6650.c}
- drivers/mfd/ab3550-core.c
- include/linux/dmaengine.h
The <linux/module.h> pretty much brings in the kitchen sink along
with it, so it should be avoided wherever reasonably possible in
terms of being included from other commonly used <linux/something.h>
files, as it results in a measureable increase on compile times.
The worst culprit was probably device.h since it is used everywhere.
This file also had an implicit dependency/usage of mutex.h which was
masked by module.h, and is also fixed here at the same time.
There are over a dozen other headers that simply declare the
struct instead of pulling in the whole file, so follow their lead
and simply make it a few more.
Most of the implicit dependencies on module.h being present by
these headers pulling it in have been now weeded out, so we can
finally make this change with hopefully minimal breakage.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
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>
We really should be doing this in the core, not in a driver...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
The number of connected input and output endpoints for a given widgets
can't change during a DAPM run so there is no need to redo the recursion
through branches of the tree we've already visited. Doing this on one of
my test systems gives an improvement of:
Power Path Neighbour
Before: 63 607 731
After: 63 141 181
which scales up well as more widgets are involved in paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handling of user control elements was implemented for all types except
ENUMERATED. This type will be needed for the device-specific mixers of
upcoming FireWire drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some widgets will get power_check() run on them more than once during a
DAPM run, most commonly due to supply widgets checking to see if their
consumers are powered up. It's wasteful to do this so cache the result
of power_check() during a run. For one system I tested this on I got an
improvement of:
Power Path Neighbour
Before: 106 970 1186
After: 69 727 905
from this.
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>
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Lars-Peter Clausen