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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.11 (GNU/Linux)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=wLdE
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.15 (GNU/Linux)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=jhuo
-----END PGP SIGNATURE-----
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
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.14 (GNU/Linux)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=5zhn
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.15 (GNU/Linux)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=EOnb
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.14 (GNU/Linux)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=rVsK
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.14 (GNU/Linux)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=qzif
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=UZaY
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=gWf9
-----END PGP SIGNATURE-----
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
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)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=DMsb
-----END PGP SIGNATURE-----
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
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)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=sYiI
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=CgWu
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)
iQIcBAABAgAGBQJRb/60AAoJELSic+t+oim9vL8P+wfaXRBGDtxhzMnTCf/cR2sc
vlRnmSjA4s14vCoPffQJI0zaGLwDT5FnQtg6DAkP/8vXRoUz4Hgb0UwwDPiQNGED
6Wmqm7mU+XGWgo4bPBA1e3Bt/9phVVO62rNPMNEnNqcp/Fa3RrdFAfxy4EUz9sKa
lX4KJETCbIvLpOJmgq3H/WGtgYEnULHSCCNUQQ+fEY/VRQLsMtY5+tnZIJilMez0
Ff6B084kE5oQpMsxdf89q8O5Uqc8lB0Xleluh0yQ1YZK3lxELMgr1Z7BkitysaJh
uid+Ze8Vj2n5duI87OZcHN1Z2SibgTzqUwsd6YGCUKK3D3KVcSYgaYNn3zY09KNG
tYlckAOJgVXqe1jedsfyuKTraz2JBY+jWYcIf8cRbwxxZpItG4Oj3idIBAKw+FrE
/0DGqW7U9wXKx8pg7BH3dE6J6WVZ5uryaQX9d+nC8CGGjpcCla5L5jl/8stgGniW
StTk4ETB6PP6iApv11p/7CXaTqXi+9UHmlcHFo11oQKiJFx4kG21DKQCXS0ycocM
j0/gRGesWrVawYwJ86dhciUJjWlTHwproE/75i1JsTd3eRX6ybjBeNTTAI2ll/BJ
BFDTS7tbX7GVcNbwXCvxW6pKOPpqV9jh0yMgpaB4jtkXOTKV/Z73ThPEql5w27c5
OTBtONmiYeBcZGvgKQ3r
=MXrq
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=lx37
-----END PGP SIGNATURE-----
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)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.11 (GNU/Linux)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=/vBT
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=ivD5
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=tDkv
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=3VlW
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=2irt
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)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=uNqD
-----END PGP SIGNATURE-----
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).
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=nHb7
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=JSSt
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=X06H
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.19 (GNU/Linux)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=cSAm
-----END PGP SIGNATURE-----
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
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=JHE2
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=5L34
-----END PGP SIGNATURE-----
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>
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)
iQEbBAABAgAGBQJPi3XOAAoJEHm+PkMAQRiGnsUH9RjHwH4YFVyuP/DKtKa6zs74
wqkpT15yITQ5WWMog4JaJFFg5rJCUd8QZr7AS/HSn0ijDyZX5VU7Rcs9cMudDzNR
H/5K/AscS4fjb0HwWVqoltTWHRb9QGSwVN3+E3VCDLt9P89YJ0o3QztkkuEX5dkZ
jc7reVXTfRnCcILEa9jleOzrn+OLM3j/jAjQ2hGunl8EDLzD4b17HHPoli4jEZ/5
5ibpSVsPD+AqzN+glbXvYjVItl12D0IQos/JdOwfuZriCVWLxysSSwHZTbPCyvBZ
LHH4HR5T+XLSXbjJeNkUFHLzqU+d5gVRadIoWtJCxqxFjKbOs2YtzJ5Ai0nDiw==
=kTkC
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.11 (GNU/Linux)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=3j4+
-----END PGP SIGNATURE-----
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?
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)
iQEcBAABAgAGBQJPMzv8AAoJEHm+PkMAQRiGgeYIAJxxGCxSMLRIiPaGRmQbWifs
eE3z6AUWitGfYvhmYdWVz+C/aa4pPCVWr3bAFxUQhu+ESmiFIiqiefvlTRf5f84e
pcoJFobkFfhtK4gLvUr75WxXDFq/1HhKqW1OLzIW4gQsnhMdjGJ3lLK6ZoRYH8h4
cGoN3DRII9mVLFcB2sgbeRvbQBA/O4n0aOQ2ryZkCcPQQMIKhR/rUEEV2pe/ovHE
rumQgI3PVey816P4e5YnOvrEAllFZlaZ/F0ClJgNdfeQCVTZx9KNRbiNxHi6fcC5
6Zgg7e5EvQTAy7QqQZ0hmS4k4CFmnswX3Nm+ZabRNqBdLSjPEDD9hk1o1dodYCg=
=BBLC
-----END PGP SIGNATURE-----
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, ...
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.11 (GNU/Linux)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=vPhu
-----END PGP SIGNATURE-----
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>
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).
So now this flag is a NOOP and can be removed.
Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model_id is no longer needed within the platform_data
for the TPA driver since the model of TPA specified
with the device name (tpa6130a2/tpa6140a2).
Also update rx51 (the only affected user) to use the device name rather
than platform data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.
We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Add a helper function to allow drivers to disable hardware resampling
when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE
flag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.
The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.
2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive: To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero. At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller. This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.
With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.
This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter. As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices that need this exist; obviously the newer regmap defaults
mechanism will deal with this more happily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Misc fixes to improve code readability:
* rename struct pm_qos_request_list to struct pm_qos_request,
* rename pm_qos_req parameter to req in internal code,
consistenly use req in the API parameters,
* update the in-kernel API callers to the new parameters names,
* rename of fields names (requests, list, node, constraints)
Signed-off-by: Jean Pihet <j-pihet@ti.com>
Acked-by: markgross <markgross@thegnar.org>
Reviewed-by: Kevin Hilman <khilman@ti.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
The PM QoS implementation files are better named
kernel/power/qos.c and include/linux/pm_qos.h.
The PM QoS support is compiled under the CONFIG_PM option.
Signed-off-by: Jean Pihet <j-pihet@ti.com>
Acked-by: markgross <markgross@thegnar.org>
Reviewed-by: Kevin Hilman <khilman@ti.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for the Analog Devices ADAU1373 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My gmail account got disabled and I'm not going to reopen it.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow drivers to set up their own regmap API structures. This is mainly
useful with MFDs where the core driver will have set up regmap at the
minute, though it may make sense to push the existing regmap setup out
of the core into the drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove all the ASoC specific physical I/O code and replace it with calls
into the regmap API. The bulk write code can only be used safely if all
regmap calls are locked with the CODEC lock, we need to add bulk support
to the regmap API or replace the code with an open coded loop (though
currently it has no users...).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For marketing reasons the part will be called WM8996. In order to avoid
user confusion rename the driver to reflect this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (430 commits)
[media] ir-mce_kbd-decoder: include module.h for its facilities
[media] ov5642: include module.h for its facilities
[media] em28xx: Fix DVB-C maxsize for em2884
[media] tda18271c2dd: Fix saw filter configuration for DVB-C @6MHz
[media] v4l: mt9v032: Fix Bayer pattern
[media] V4L: mt9m111: rewrite set_pixfmt
[media] V4L: mt9m111: fix missing return value check mt9m111_reg_clear
[media] V4L: initial driver for ov5642 CMOS sensor
[media] V4L: sh_mobile_ceu_camera: fix Oops when USERPTR mapping fails
[media] V4L: soc-camera: remove soc-camera bus and devices on it
[media] V4L: soc-camera: un-export the soc-camera bus
[media] V4L: sh_mobile_csi2: switch away from using the soc-camera bus notifier
[media] V4L: add media bus configuration subdev operations
[media] V4L: soc-camera: group struct field initialisations together
[media] V4L: soc-camera: remove now unused soc-camera specific PM hooks
[media] V4L: pxa-camera: switch to using standard PM hooks
[media] NetUP Dual DVB-T/C CI RF: force card hardware revision by module param
[media] Don't OOPS if videobuf_dvb_get_frontend return NULL
[media] NetUP Dual DVB-T/C CI RF: load firmware according card revision
[media] omap3isp: Support configurable HS/VS polarities
...
Fix up conflicts:
- arch/arm/mach-omap2/board-rx51-peripherals.c:
cleanup regulator supply definitions in mach-omap2
vs
OMAP3: RX-51: define vdds_csib regulator supply
- drivers/staging/tm6000/tm6000-alsa.c (trivial)
Change locking to allow tea575x-radio device to be opened multiple times.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Convert tea575x-tuner to use the new V4L2 control framework. Also add
ext_init() callback that can be used by a card driver for additional
initialization right before registering the video device (for SF16-FMR2).
Also embed struct video_device to struct snd_tea575x to simplify the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Move the macros depending on snd_mask_min() and co out of pcm.h into
pcm_params.h. Otherwise using some params_*() macros will give comiple
errors without inclusion of pcm_params.h.
Also use hw_param_interval_c() and hw_param_mask_c() for const pointer.
Reported-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for Dynamic PCM (AKA DSP) support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs
after completion of stream events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (43 commits)
fs: Merge split strings
treewide: fix potentially dangerous trailing ';' in #defined values/expressions
uwb: Fix misspelling of neighbourhood in comment
net, netfilter: Remove redundant goto in ebt_ulog_packet
trivial: don't touch files that are removed in the staging tree
lib/vsprintf: replace link to Draft by final RFC number
doc: Kconfig: `to be' -> `be'
doc: Kconfig: Typo: square -> squared
doc: Konfig: Documentation/power/{pm => apm-acpi}.txt
drivers/net: static should be at beginning of declaration
drivers/media: static should be at beginning of declaration
drivers/i2c: static should be at beginning of declaration
XTENSA: static should be at beginning of declaration
SH: static should be at beginning of declaration
MIPS: static should be at beginning of declaration
ARM: static should be at beginning of declaration
rcu: treewide: Do not use rcu_read_lock_held when calling rcu_dereference_check
Update my e-mail address
PCIe ASPM: forcedly -> forcibly
gma500: push through device driver tree
...
Fix up trivial conflicts:
- arch/arm/mach-ep93xx/dma-m2p.c (deleted)
- drivers/gpio/gpio-ep93xx.c (renamed and context nearby)
- drivers/net/r8169.c (just context changes)
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All these are instances of
#define NAME value;
or
#define NAME(params_opt) value;
These of course fail to build when used in contexts like
if(foo $OP NAME)
while(bar $OP NAME)
and may silently generate the wrong code in contexts such as
foo = NAME + 1; /* foo = value; + 1; */
bar = NAME - 1; /* bar = value; - 1; */
baz = NAME & quux; /* baz = value; & quux; */
Reported on comp.lang.c,
Message-ID: <ab0d55fe-25e5-482b-811e-c475aa6065c3@c29g2000yqd.googlegroups.com>
Initial analysis of the dangers provided by Keith Thompson in that thread.
There are many more instances of more complicated macros having unnecessary
trailing semicolons, but this pile seems to be all of the cases of simple
values suffering from the problem. (Thus things that are likely to be found
in one of the contexts above, more complicated ones aren't.)
Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Add a convenience macro for external enumerated widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform driver widgets to perform any IO required for DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of ioctl definition in sound/sb16_csp.h contains the data size
over 8kB, and this causes build errors on architectures like MIPS,
which define _IOC_SIZEBITS=13.
For avoiding this build errors but keeping the compatibility, manually
expand with _IOC() instead of using _IOW() for the problematic ioctl.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kill tasklet usage in rawmidi core code. Use workq for the event callback
instead of tasklet (which is used only in core/seq/seq_midi.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.
Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800
among others.
Signed-off-by: Harald Welte <laforge@gnumonks.org>
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide real card and bus_info instead of hardcoded values.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card *card is present in struct snd_tea575x but never used.
Remove it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
freq_fixup is a constant, no need to hold it in struct snd_tea575x and set in
each driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow ASoC machine drivers to register a driver name
and a longname. This allows user space to determine
the flavour of machine driver.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Implement generic read/write functions to access TEA575x tuners. They're now
implemented 4 times (once in es1968 and 3 times in fm801).
This also allows mute to work on all cards.
Also improve tuner detection/initialization.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will allow multiple widgets to be affected by the same
control. For example, a single register bit that controls separate muxes
in both the L and R audio paths.
This change updates the code that handles relevant controls to be able
to iterate over a list of affected widgets. Note that only the put
functions need significant modification to implement the iteration; the
get functions do not need to iterate, nor unify the results, since all
affected widgets reference the same kcontrol.
When creating the list of widgets, always create a 1-sized list, since
the control sharing is not implemented in this change.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Future changes will need reference to the kcontrol created for a given
kcontrol_new. Store the created kcontrol values now.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.
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>
The enum texts are supposed to be const char * const []. Without the
second const, it gets compile warnings like
sound/soc/codecs/max98095.c:607:2: warning: initialization discards qualifiers from pointer target type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the equalizer and biquad filter controls.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
replace the tab with spaces,
make it align with other paragraphs
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those should not be modified (and are not) by the core code, so make them const.
This also makes them consistent with the same members of snd_soc_codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the MAX98095 CODEC driver.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide the top level ASoC core functions for indicating whether
a given register is readable or writable.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using struct snd_soc_reg_access for the read/write/vol attributes
of the registers, we provide callbacks that automatically determine whether
a given register is readable/writable or volatile.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is mainly used by the soc-cache code to easily determine the
currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we
can distinguish it if it is not initialized or set.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As it has become more common to have to write firmware or similar
large chunks of data to the hardware, add a function to perform
raw bulk writes that bypass the cache. This only handles volatile
registers as we should avoid getting out of sync with the actual
cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's a big no-no to use pgprot_noncached() when mmap'ing such buffers
into userspace since they are mapped cachable in kernel space.
This can cause all sort of interesting things ranging from to garbled
sound to lockups on various architectures. I've observed that usb-audio
is broken on powerpc 4xx for example because of that.
Also remove the now unused snd_pcm_lib_mmap_noncached(). It's
an arch business to know when to use uncached mappings, there's
already hacks for MIPS inside snd_pcm_default_mmap() and other
archs are supposed to use dma_mmap_coherent().
(See my separate patch that adds dma_mmap_coherent() to powerpc)
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This stops code that handles widgets generically from attempting to access
registers for these widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Add a function to dynamically replace a given control. If the
control does not already exist, a third parameter is used to determine
whether to actually add that control. This is useful in cases where
downloadable firmware at runtime can add or replace existing controls.
A separate patch needs to be made to allow ALSA Mixer to render the
replaced controls on the fly.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve tea575x-tuner with various good things from radio-maestro:
- extend frequency range to 50-150MHz
- fix querycap(): card name, CAP_RADIO
- improve g_tuner(): CAP_STEREO, stereo and tuned indication
- improve g_frequency(): tuner index checking and reading frequency from HW
- improve s_frequency(): tuner index and type checking
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new API function snd_ctl_activate_id() for activate / inactivate
the control element dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for tlv320aic3205 and tlv320aic3254 codecs.
It doesn't include miniDSP support for aic3254.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is run after the DAPM widgets and routes are added, allowing setup
of things like jacks using the routes. The main card probe() is run before
anything else so can't be used for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
These will be added after all devices are registered and allow most DAI
init functions in machine drivers to be replaced by simple data.
Regular controls are not supported as the registration function still
works in terms of CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This means that rather than adding the board specific DAPM widgets to a
random CODEC DAPM context they can be added to the card itself which is
a bit cleaner. Previously there only was one DAPM context and it was
tied to the single supported CODEC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM9081 IRQ output can be either active high or active low and can
support either CMOS or open drain modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the core code where sparse complains. In most cases, this means
just adding annotations to confirm that we indeed want to do the dirty
things we're doing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* There is no hysteresis enable field in the current datasheet.
* Mic detection threshold field is only 2 bits wide.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds soc-jack support for adding voltage zones and for
detecting jack type
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move Chip Select control out of the CODEC code for CS4271.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide driver data for cards within the card structure. To simplify the
implementation of the PM operations we don't use the struct device driver
data as this is used by the core to retrieve the card in callbacks from
the device model and PM core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allows drivers to distinguish which subsequence is being notified when
they get called back.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Could just as well live in sysfs but sysfs doesn't have the simple
value export helpers debugfs does.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow hookup of cards registered directly with the core to the PM
operations by exporting the device power management operations to
modules, also exporting the default PM operations since it is
expected that most cards will end up using exactly the same setup.
Note that the callbacks require that the driver data for the card be
the snd_soc_card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As requested by Takashi and Jaroslav, these arrays should not be in the
header file.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current format selection of FSI-codecs depended on platform information for FSI,
and chip default settings for codecs. It is not understandable/formal method.
This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt.
But FSI can use I2S/PCM and SPDIF format today.
It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF.
So, this patch change FSI platform information to have DAI/SPDIF mode.
If platform selects DAI mode (default),
FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt,
and if it is SPDIF mode, FSI become SPDIF format.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have zero users for PGA controls and the core support for them was
removed a while ago so no point in cut'n'pasting them into new macros,
even if it's too much hassle to update the existing ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We generally refer to registers as unsigned ints (including in the
underlying CODEC driver operation).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add snd_soc_dai_ops :: set_fmt to FSI driver and
select master/slave clock mode by snd_soc_dai_set_fmt on
fsi-xxx.c instead of platform infomation code.
This patch remove fsi_is_master function which is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for CS4271 codec to ASoC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, update platform_data GPIO handling to have an explicit "don't
touch this pin" option.
Add #defines for the GPIO pin functions.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is primarily needed to avoid writing back to the cache
whenever we are syncing the cache with the hardware. This gives a
performance benefit especially for large register maps.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
With larger devices there may be many widgets of the same type in series
in an audio path. Allow drivers to specify an additional level of ordering
within each widget type by adding a subsequence number to widgets and then
splitting operations on widgets so that widgets of the same type but
different sequence numbers are processed separately. A typical example
would be a supply widget which requires that another widget be enabled
to provide power or clocking.
SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided
allowing this to be used with PGAs and supplies as these are the most
commonly affected widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The machine driver can't register the card directly and need to do this thru
soc-audio device creation
This patch allows the register and unregister card to be directly called by
machine drivers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the soc_probe initializes the card hence it does the card list
initialzation. But if machines directly register the card they would need to
do these steps, so putting them as inline would save lot of code
This patch adds an inline to do list initialzation
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <harsha.priya@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that all calls to readable_register()/volatile_register() go via
the snd_soc_codec function pointers.
If the default register access table has been given but no functions
for handling readable()/volatile() registers, use the default ones provided
by soc-cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For common scenarios, device drivers can provide a table of all the
registers that are at least either readable/writable/volatile. The idea
is that if a register lookup fails, all of its read/write/vol members
will be zero and will be treated as default. This also reduces the
size of the register access array.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simplify the use of reg_size, by calculating it once and storing it in
the codec structure for later reference. The value of reg_size is
reg_cache_size * reg_word_size.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Everything else is using snd_soc_ so we should use it here too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.
Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new type is a virtual version of snd_soc_dapm_mux. It is used
when a backing register value is not necessary for deciding which
input path to connect. A simple virtual enumeration control e.g.
SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used
to choose which path to connect.
The snd_soc_dapm_virt_mux type ensures that during the initial
path setup, the first (which is also the default) input path will
be connected.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Attempt to minimise audible effects from mixer and mux updates by
implementing the actual register changes between powering down widgets
that have become unused and powering up widgets that are newly used.
This means that we're making the change with the minimum set of widgets
powered, that the input path is connected when we're powering up widgets
(so things like DC offset correction can run with their signal active)
and that we bring things down to cold before switching away. Since
hardware tends to be designed for the power on/off case more than for
dynamic reconfiguration this should minimise pops and clicks during
reconfiguration while active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.
This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.
DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no users of these and it's not clear what they would do given
the mono flow modelling which DAPM does. If need arises we can add them
again.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.
Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes some legacy structure definitions which are not using
in current ASoC drivers.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.
Remove redundant newline in source code.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch allows machine drivers to override the compression type
provided by the codec driver.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary. This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The snd_soc_codec_conf struct now holds codec specific configuration
information.
A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.
In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead. In all other cases the one
set in the snd_soc_codec_conf struct takes effect.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the base value of compress_type starts at 1 so that
we know whether the machine driver has provided a compress_type
for overriding the codec supplied one.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with. The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.
Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Neither drivers nor the core should be fiddling with the actual ops
structure at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes possible to register auxiliary dailess codecs in a machine
driver. Term dailess is used here for amplifiers and codecs without DAI or
DAI being unused.
Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs
are probed after initializing the DAI links. There are no major differences
between DAI link codecs and dailess codecs in ASoC core point of view. DAPM
handles them equally and sysfs and debugfs directories for dailess codecs
are similar except the pmdown_time node is not created.
Only suspend and resume functions are modified to traverse all probed codecs
instead of DAI link codecs.
Example below shows a dailess codec registration.
struct snd_soc_aux_dev foo_aux_dev[] = {
{
.name = "Amp",
.codec_name = "codec.2",
.init = foo_init2,
},
};
static struct snd_soc_card card = {
...
.aux_dev = foo_aux_dev,
.num_aux_devs = ARRAY_SIZE(foo_aux_dev),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and
/dev/snd/timer the usual static minors, and export specific
module aliases to generate udev module on-demand loading
instructions:
$ cat /lib/modules/2.6.33.4-smp/modules.devname
# Device nodes to trigger on-demand module loading.
microcode cpu/microcode c10:184
fuse fuse c10:229
ppp_generic ppp c108:0
tun net/tun c10:200
uinput uinput c10:223
dm_mod mapper/control c10:236
snd_timer snd/timer c116:33
snd_seq snd/seq c116:1
The last two lines instruct udev to create device nodes, even
when the modules are not loaded at that time.
As soon as userspace accesses any of these nodes, the in-kernel
module-loader will load the module, and the device can be used.
The header file minor calculation needed to be simplified to
make __stringify() (supports only two indirections) in
the MODULE_ALIAS macro work.
This is part of systemd's effort to get rid of unconditional
module load instructions and needless init scripts.
Cc: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need anymore to include soc.h in soc-dapm.h and soc-dai.h as
drivers are converted to include only soc.h.
Thanks to Lars-Peter Clausen <lars@metafoo.de> for pointing out the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows to disable period interrupts which are
not needed when the application relies on a system timer
to wake-up and refill the ring buffer. The behavior of
the driver is left unchanged, and interrupts are only
disabled if the application requests this configuration.
The behavior in case of underruns is slightly different,
instead of being detected during the period interrupts the
underruns are detected when the application calls
snd_pcm_update_avail, which in turns forces a refresh of the
hw pointer and shows the buffer is empty.
More specifically this patch makes a lot of sense when
PulseAudio relies on timer-based scheduling to access audio
devices such as HDAudio or Intel SST. Disabling interrupts
removes two unwanted wake-ups due to period elapsed events
in low-power playback modes. It also simplifies PulseAudio
voice modules used for speech calls.
To quote Lennart "This patch looks very interesting and
desirable. This is something have long been waiting for."
Support for this in hardware drivers is optional.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.
This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.
Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.
static const struct snd_soc_dapm_route map0[] = {
{"Spk", NULL, "MONO"},
};
static const struct snd_soc_dapm_route map1[] = {
{"Vibra", NULL, "foo MONO"},
};
static struct snd_soc_prefix_map codec_prefix[] = {
{
.dev_name = "codec.2",
.name_prefix = "foo",
},
};
static struct snd_soc_card card = {
...
.prefix_map = codec_prefix,
.num_prefixes = ARRAY_SIZE(codec_prefix),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for rbtree compression when storing the
register cache. It does this by not adding any uninitialized registers
(those whose value is 0). If any of those registers is written
with a nonzero value they get added into the rbtree.
Consider a sample device with a large sparse register map. The
register indices are between [0, 0x31ff]. An array of 12800 registers
is thus created each of which is 2 bytes. This results in a 25kB
region. This array normally lives outside soc-core, normally in the
driver itself. The original soc-core code would kmemdup this region
resulting in 50kB total memory. When using the rbtree compression
technique and __devinitconst on the original array the figures are
as follows. For this typical device, you might have 100 initialized
registers, that is registers that are nonzero by default. We build
an rbtree with 100 nodes, each of which is 24 bytes. This results
in ~2kB of memory. Assuming that the target arch can freeup the
memory used by the initial __devinitconst array, we end up using
about ~2kB bytes of actual memory. The memory footprint will increase
as uninitialized registers get written and thus new nodes created in
the rbtree. In practice, most of those registers are never changed.
If the target arch can't freeup the __devinitconst array, we end up
using a total of ~27kB. The difference between the rbtree and the LZO
caching techniques, is that if using the LZO technique the size of
the cache will increase slower as more uninitialized registers get
changed.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for LZO compression when storing the register
cache. The initial register defaults cache is marked as __devinitconst
and the only change required for a driver to use LZO compression is
to set the compress_type member in codec->driver to SND_SOC_LZO_COMPRESSION.
For a typical device whose register map would normally occupy 25kB or 50kB
by using the LZO compression technique, one can get down to ~5-7kB. There
might be a performance penalty associated with each individual read/write
due to decompressing/compressing the underlying cache, however that should not
be noticeable. These memory benefits depend on whether the target architecture
can get rid of the memory occupied by the original register defaults cache
which is marked as __devinitconst. Nevertheless there will be some memory
gain even if the target architecture can't get rid of the original register
map, this should be around ~30-32kB instead of 50kB.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces the new caching API and migrates the
old caching interface into the new one. The flat register caching
technique does not use compression at all and it is equivalent to
the old caching technique. One can still access codec->reg_cache
directly but this is not advised as that will not be portable
across different caching strategies.
None of the existing drivers need to be changed to adapt to this
caching technique. There should be no noticeable overhead associated
with using the new caching API.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There will be need to have sound card specific debugfs entries. This patch
introduces a new debugfs/asoc/{card->name}/ directory but does not add yet
any entries there.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Facilitating adding trace type stuff. For a first pass add some dev_dbg()
statements into them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
This patch removes the old CONFIG_SYSFS_DEPRECATED_V2 config option,
but it keeps the logic around to handle block devices in the old manner
as some people like to run new kernel versions on old (pre 2007/2008)
distros.
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Cc: Jens Axboe <axboe@kernel.dk>
Cc: Stephen Hemminger <shemminger@vyatta.com>
Cc: "Eric W. Biederman" <ebiederm@xmission.com>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: "James E.J. Bottomley" <James.Bottomley@suse.de>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: Randy Dunlap <randy.dunlap@oracle.com>
Cc: Tejun Heo <tj@kernel.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit f6765502f8 and adds
the missing include file.
Signed-off-by: Peter Hsiang <Peter.Hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CCR is defined in emu10k1, but SuperH is defined too.
If user use this driver with SuperH, it becomes a double definition.
Signed-off-by: Nobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>
Cc: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we compile the ASoC code with PM disabled, we hit stuff like:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_suspend_check':
sound/soc/soc-dapm.c:440: warning: unused variable 'codec'
So tweak the stub macro to avoid these issues.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With generic AC97 ASoC glue driver (codec/ac97.c), we get following warning when
the device is registered (slightly stripped the backtrace):
kobject (c5a863e8): tried to init an initialized object, something is seriously
wrong.
[<c00254fc>] (unwind_backtrace+0x0/0xec)
[<c014fad0>] (kobject_init+0x38/0x70)
[<c0171e94>] (device_initialize+0x20/0x70)
[<c017267c>] (device_register+0xc/0x18)
[<bf20db70>] (snd_soc_instantiate_cards+0x924/0xacc [snd_soc_core])
[<bf20e0d0>] (snd_soc_register_platform+0x16c/0x198 [snd_soc_core])
[<c0175304>] (platform_drv_probe+0x18/0x1c)
[<c0174454>] (driver_probe_device+0xb0/0x16c)
[<c017456c>] (__driver_attach+0x5c/0x7c)
[<c0173cec>] (bus_for_each_dev+0x48/0x78)
[<c0173600>] (bus_add_driver+0x98/0x214)
[<c0174834>] (driver_register+0xa4/0x130)
[<c001f410>] (do_one_initcall+0xd0/0x1a4)
[<c0062ddc>] (sys_init_module+0x12b0/0x1454)
This happens because the generic AC97 glue driver creates its codec->ac97 via
calling snd_ac97_mixer(). snd_ac97_mixer() provides own version of
snd_device.register which handles the device registration when
snd_card_register() is called.
To avoid registering the AC97 device twice, we add a new flag to the
snd_soc_codec: ac97_created which tells whether the AC97 device was created by
SoC subsystem.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than block the workqueue by sleeping to do the debounce use delayed
work to implement the debounce time. This should also means that we extend
the debounce time on each new bounce, potentially allowing shorter debounce
times for clean insertions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8962 features five GPIOs, add support for controlling their output
state via gpiolib.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add the widget for MICBIAS power control and allow configuration of the
microphone bias setup via the platform data for the WM8962. When
microphone status signals are brought out to GPIO this should be
sufficient to enable microphone detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide an initial hookup for interrupts on the WM8962. Currently we simply
report error status via log messages if an IRQ is provided for the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some devices have more flexible microphone detection and can detect
a wider range of buttons.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Swapping the bias level enumeration is only meant to help debugging. It is
easier if number 0 means bias off and bigger number means bigger bias level.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.
It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.
More information: Kernel bugzilla bug#16300
[A copmile warning fixed by tiwai]
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fairly simple conflicts, the most serious ones are the i.MX ones which I
suspect now need another rename.
Conflicts:
arch/arm/mach-mx2/clock_imx27.c
arch/arm/mach-mx2/devices.c
arch/arm/mach-omap2/board-rx51-peripherals.c
arch/arm/mach-omap2/board-zoom2.c
sound/soc/fsl/mpc5200_dma.c
sound/soc/fsl/mpc5200_dma.h
sound/soc/fsl/mpc8610_hpcd.c
sound/soc/pxa/spitz.c
unifdef-y and header-y has same semantic.
So there is no need to have both.
Drop the unifdef-y variant and sort all lines again
Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request(). This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.
Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence number
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.
This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems codecs need to configure some registers before and after
powering down some of their part. As a convenience add a macro for that.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.
I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.
Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.
Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation. I did this because request more
accurately represents what it actually does.
Also, I added a string based ABI for users wanting to use a string
interface. So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface. (someone asked me for it and I don't think
it hurts anything.)
This patch updates some documentation input I got from Randy.
Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.
Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.
Also, a new PnP id is added for the card I acquired (the change
was tested on this card).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6f3991152f.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)
If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:
snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We actually pass an array of 7 chars not 5.
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.
Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices provide support for detection of a small number of
buttons on their jacks. One common implementation provides a single
button, implemented by shorting the microphone to ground and detected
along with microphone presence detection by detecting varying current
draws on the microphone bias signal.
Provide support for up to three buttons via the jack interface. These
default to reporting BTN_n but an API is provided to allow these to
be remapped to other keys by the machine driver where it knows what
the keys are. More keys can be added with ease if required.
This is only intended to support simple accessory button designs. If
the interface is limiting then either creating a child device for the
accessory or accessing the input device in the jack directly is
recommended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8904 allows microphone detection signals to be brought out as
alternate functions of the GPIO signals which can be detected using
interrupt inputs on the CPU. Allow this to be configured using
platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide platform data allowing the configuration of the GPIO pins
on the WM8904 to be selected, allowing alternate functions to be
enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support use of the WM8903 IRQ for reporting of microphone presence
and short detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently used to detect completion of the write sequencer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide support for WM8903 microphone presence and short detection
using the GPIOs to route out a logic signal suitable for handling
using snd_soc_jack_add_gpios() on the processor GPIOs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow users to pass in a default configuration for the GPIOs of
the WM8903 as platform data. This allows configuration of the pin
muxing of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow pins to be forced on regardless of their power state. This is
intended for use with microphone bias supplies which need to be
enabled in order to support microphone detection - in systems without
appropriate hardware leaving the microphone unbiased when not in use
saves power.
The force done at power check time in order to avoid disrupting other
power detection logic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Platform data option for the codec to keep the BCLK clock
continuously running in FIFO modes (codec master).
OMAP3 McBSP when in slave mode needs continuous BCLK running
on the serial bus in order to operate correctly.
Since in FIFO mode the DAC33 can also shut down the BCLK clock
and enable it only when it is needed, let the platforms decide
if the CPU side needs the BCLK running or not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The flag is no longer used in the code so it just wastes a bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.
Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Avoids machine files having to peer into sound/soc which is a bit
rude and icky.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr
update functions" commit:
"It is possible for the status/delay ioctls to be called when the sound
card's pointer register alreay shows a position at the beginning of the
new period, but immediately before the interrupt is actually executed.
(This happens regularly on a SMP machine with mplayer.) When that
happens, the code thinks that the position must be at least one period
ahead of the current position and drops an entire buffer of data."
Return back the hw_ptr_interrupt variable. The last interrupt pointer
is always computed from the latest hw_ptr instead of tracking it
separately (in this case all hw_ptr checks and modifications might
influence also hw_ptr_interrupt and it is difficult to keep it
consistent).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Several shortcuts for popular uses of some of SOC_ENUM_* and
SOC_VALUE_ENUM_* macros.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many macros from include/sound/soc-dapm.h take an array and a number of
elements in it as arguments, whereas most users use static arrays and use
"x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of
those macros, calling ARRAY_SIZE() internally.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O"
commit. New sleeping queue is introduced to separate user space and kernel
space wake_ups. runtime->nowake is renamed to twake (transfer wake).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd
need non-cached behavior more or less, even for the intermediate ring-
buffers.
Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback
that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new function to look up a quirk entry with the given PCI SSID
instead of a pci device pointer. This can be used when the searched ID
is overridden for debugging or such a purpose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
tpa6140a2 uses different names for the regulators.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As noted by pl bossart <bossart.nospam@gmail.com>, the PCM I/O routines
(snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls
until all samples are not processed.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them.
The main change is hw_ptr_interrupt variable removal to simplify code
logic. This variable can be computed directly from hw_ptr.
Ensure that updated hw_ptr is not lower than previous one (it was possible
with old code in some obscure situations when interrupt was delayed or
the lowlevel driver returns wrong ring buffer position value).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
In some debug cases, it might be usefull to see previous ring buffer
positions to determine position problems from the lowlevel drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is one of Sound Blaster Pro compatible chipsets which is supported
by Linux OSS driver and was missing native supoort for ALSA.
The Jazz16 audio codec is Crystal CS4216 which is capable
of playback and recording up to 48 kHz stereo.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8955 is a low power, high quality stereo DAC with integrated
headphone and loudspeaker amplifiers, designed to reduce external
component requirements in portable digital audio applications. This is
an initial driver implementing support for the majority of the
functionality in the device, currently OUT3 is not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
There are now five copies of the code to allocate a PCM buffer using
vmalloc(). Add a sixth in the core so that the others can be removed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8904 is a high performance ultra-low power stereo CODEC
optimised for portable audio applications, with features including
a class W amplifier, FLL with free running mode, Mobile ReTune and
ground referenced headphone and line outputs.
Support for some features, most particularly the digital microphone
interface, is not yet present.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.
Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.
Also, delete one misnamed cs4231 register define.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Record the pid of the task that opened a RawMIDI substream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of storing the PID number, take a reference to the task's pid
structure. This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.
Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).
Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.
Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.
An additional firmware initialization code has been moved from the OSS
driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DAI format definition for PDM interfaces.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* complete support for ak4113
* based on code for ak4114 and ak4117
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* complete support for ak4620
* codec regs listed in proc for all codecs/chips
* adding total regs for each codec
* fixing nb. of steps in input attenuation controls
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* the previous version had a typo - values of AK4114_OPS10-12 were
identical with AK4114_OPS00-02
* Since no cards actually use this feature, the bug was not identified earlier
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.
Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.
To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.
A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps CODECs with sparse register maps work better with the
register cache display interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for
double controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double
controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using SG-buffers with dma_alloc_coherent() is often very inefficient
on non-coherent architectures because a tracking record could be
allocated in addition for each dma_alloc_coherent() call.
Instead, simply disable SG-buffers but just allocate normal continuous
buffers on non-supported (currently all but x86) architectures.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides. For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.
At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses. However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.
This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended. At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/pcm-jiffies-check:
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: pcm - Fix update of runtime->hw_ptr_interrupt
ALSA: pcm - Fix a typo in hw_ptr update check
ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value
ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
ALSA: PCM midlevel: introduce mask for xrun_debug() macro
ALSA: PCM midlevel: improve fifo_size handling
* topic/asoc: (135 commits)
ASoC: Apostrophe patrol
ASoC: codec tlv320aic23 fix bogus divide by 0 message
ASoC: fix NULL pointer dereference in soc_suspend()
ASoC: Fix build error in twl4030.c
ASoC: SSM2602: assign last substream to the master when shutting down
ASoC: Blackfin: document how anomaly 05000250 is handled
ASoC: Blackfin: set the transfer size according the ac97_frame size
ASoC: SSM2602: remove unsupported sample rates
ASoC: TWL4030: Check the interface format for 4 channel mode
ASoC: TWL4030: Use reg_cache in twl4030_init_chip
ASoC: Initialise dev for the dummy S/PDIF DAI
ASoC: Add dummy S/PDIF codec support
ASoC: correct print specifiers for unsigneds
ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
ASoC: Switch FSL SSI DAI over to symmetric_rates
ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
ASoC: Fabric bindings for STAC9766 on the Efika
ASoC: Support for AC97 on Phytec pmc030 base board.
ASoC: AC97 driver for mpc5200
ASoC: Main rewite of the mpc5200 audio DMA code
...
They are now only accessed within dapm_power_widgets() so can be local
to that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce snd_card_set_id() function to allow lowlevel drivers to set
default identification name for card slot. The function checks also
for identification name collisions and tries to create unique name.
Also, the snd_card_create() function is simplified, because this new
function is used. As bonus, proper name collision checks are evaluated
at the card create time.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fifo_size assignment to hw->ioctl callback to allow lowlevel
drivers overwrite the default behaviour.
fifo_size is in frames not bytes as specified in asound.h and alsa-lib's
documentation, but most hardware have fixed byte based FIFOs. Introduce
internal SNDRV_PCM_INFO_FIFO_IN_FRAMES.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Should be no impact on the generated code but it helps the compiler
print clearer messages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added runtime->delay field to adjust the delayed samples for snd_pcm_delay().
Typically a hardware FIFO length is stored in this field, so that the
extra delay between hwptr and applptr can be computed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAI structure has two pointers to the codec, one in the body of the
DAI and one in a union for a parent pointer. Drop the parent pointer
version.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The defines for TDM and synchronous clocks are not used - they are
mostly a legacy of the automatic clocking configuration. TDM will
require configuration of the number of timeslots and which ones to use
so can't be fit into the DAI format and synchronous mode is handled by
symmetric_rates (and needs to be done by constraints rather than when
the DAI format is being configured).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AC97 wire format is completely fixed so CODECs don't have any choice
about the formats they accept but controllers accept a variety of data
formats and render them down onto the bus. Have a shared define so all
the CODEC drivers will interoperate with any of our controller drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a macro for double controls with special callback functions.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently there are two possible platform datas for the PXA AC97 driver:
one supported by the generic AC97 driver only which provides callbacks
to allow board-specific configuration at stream startup and teardown,
and another for pxa2xx-ac97-lib which allows configuration of the reset
GPIO for PXA2xx CPUs.
Obviously this won't actually work when using the generic AC97 driver
since the drivers will attempt to parse the platform data in both
formats. Fix this by merging the two structures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Cc: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Added private_data and private_free fields to struct snd_jack so that
the caller can assign the data. It'll be helpful for avoiding the
double-free of the jack instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants.
This patch adds more strict buffer position checks based on jiffies when
internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong
positions.
As a side effect, the hw_ptr interrupt update routine might do slightly better
job when many interrupts are lost.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Impact: cleanup
The earlier patch 'make most exported headers use strict integer
types' accidentally includes <linux/types.h> both from the common and
from the kernel-only parts.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
This takes care of all files that have only a small number
of non-strict integer type uses.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Mauro Carvalho Chehab <mchehab@redhat.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Cc: YOSHIFUJI Hideaki <yoshfuji@linux-ipv6.org>
Cc: netdev@vger.kernel.org
Cc: linux-ppp@vger.kernel.org
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
A number of standard posix types are used in exported headers, which
is not allowed if __STRICT_KERNEL_NAMES is defined. In order to
get rid of the non-__STRICT_KERNEL_NAMES part and to make sane headers
the default, we have to change them all to safe types.
There are also still some leftovers in reiserfs_fs.h, elfcore.h
and coda.h, but these files have not compiled in user space for
a long time.
This leaves out the various integer types ({u_,u,}int{8,16,32,64}_t),
which we take care of separately.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Cc: YOSHIFUJI Hideaki <yoshfuji@linux-ipv6.org>
Cc: netdev@vger.kernel.org
Cc: linux-ppp@vger.kernel.org
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.
This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use define instead of enum for ioctl definitions since strace can't
parse ioctls defined via enum properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls. The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks. OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.
The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
cs4232 and cs4236 driver merge to solve PnP BIOS detection.
Also, the patch adds recognition if the chip is cs4236b+
or earlier part. This unifies drivers for both cs4232
and cs4236+ chips. It allows to use the PnP BIOS
detection for the cs4236+ chips. Previously, only
the snd-cs4232 could be detected by the PnP BIOS.
The cs4232+ cards reports two separate PnP BIOS ids.
The patch adds search for the second id to find out
resources assigned to a control port.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the AC97 controller found on Atmel
AVR32 devices.
Tested on ATSTK1006 + ATSTK1000 with a development board with a AC97
codec.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the Audio Bistream DAC found on Atmel
AVR32 devices. The ABDAC is an Atmel IP which might show up on AT91
devices in the future, hence making a generic driver which can be
utilized by AT91 arch if needed.
Datasheet describing the ABDAC peripheral is available in the AT32AP7000
datasheet, http://www.atmel.com/dyn/products/datasheets.asp?family_id=682
Tested on ATSTK1006 + ATSTK1000 with a class D amplifier stage.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: cleanup
snd_pcm_new takes a char *id argument, although it is not modifying
the string. it can therefore be declared as const char *id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix the following 'make headers_check' warning:
usr/include/sound/hdsp.h:33: found __[us]{8,16,32,64} type without #include <linux/types.h>
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Introduced snd_card_create() function as a replacement of snd_card_new().
The new function returns a negative error code so that the probe callback
can return the proper error code, while snd_card_new() can give only NULL
check.
The old snd_card_new() is still provided as an inline function but with
__deprecated attribute. It'll be removed soon later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.
This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.
[Edited to raise priority of error log message and document parameters.
-- broonie]
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a jack reporting interface to ASoC. This wraps the ALSA
core jack detection functionality and provides integration with DAPM to
automatically update the power state of pins based on the jack state.
Since embedded platforms can have multiple detecton methods used for a
single jack (eg, separate microphone and headphone detection) the report
function allows specification of which bits are being updated on a given
report.
The expected usage is that machine drivers will create jack objects and
then configure jack detection methods to update that jack.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge the recently introduced soc_value_enum structure to the soc_enum.
The value based enums are still handled separately from the normal enum types,
but with the merge some of the newly introduced functions can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows you to define the mixer paths as having the same name as the
paths they represent.
This is required to support codecs such as the wm9705 neatly without extra
controls in the alsa mixer.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Add support for reporting new jack types SND_JACK_VIDEOOUT and
SND_JACK_AVOUT (a combination of LINEOUT and VIDEOOUT) to the jack
reporting API.
Also add the corresponding SW_VIDEOOUT_INSERT switch to the input system
header.
Signed-off-by: Jani Nikula <ext-jani.1.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces a new enum type.
In this enum type each enumerated items referred with a value.
This new enum type can handle enums encoded in bitfield, or any other
weird ways. twl4030 codec has several mux selection register, where the
input/output mux is coded in a bitfield. With the normal enum type this type
of mux can not be handled in a clean way.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a lookup table rather than explicit code to map input subsystem jack
types into ASoC ones, implemented as suggested by Takashi Iwai.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce a struct v4l2_file_operations for v4l2 drivers.
Remove the unnecessary inode argument.
Move compat32 handling (and llseek) into the v4l2-dev core: this is now
handled in the v4l2 core and no longer in the drivers themselves.
Note that this changeset reverts an earlier patch that changed the return
type of__video_ioctl2 from int to long. This change will be reinstated
later in a much improved version.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.
Thanks to Guillem Jover for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 factors most of the contents of soc.h out into separate headers,
including soc-dai.h for the DAI. Factor the existing DAI API out into
this file in order to prepare for backporting of the ASoC v2 DAI API.
Also backport some of Liam's improvements to the documentation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change coding style to be more acceptable by checkpatch.pl.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change snd_BUG_ON() to evaluate the given condition, at least, in syntax
for avoiding compile warnings such as unused variables. The compiler
should optimize out the condition evaluation in the real code, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/pcxhr/pcxhr_core.c: In function 'pcxhr_set_pipe_cmd_params':
sound/pci/pcxhr/pcxhr_core.c:700: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:706: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:710: warning: statement with no effect
Due to
try to fix this, and be more conventional about the empty stubs.
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch introduces support for reporting SW_LINEOUT_INSERT detection events
via the jack abstraction layer.
Also adds a SND_JACK_LINEOUT define to the input system header.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: Dmitry Torokhov <dtor@mail.ru>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was marked as deprecated in 2.6.27 and all users except for
playpaq_wm8510 fixed in that release.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (313 commits)
V4L/DVB (9186): Added support for Prof 7300 DVB-S/S2 cards
V4L/DVB (9185): S2API: Ensure we have a reasonable ROLLOFF default
V4L/DVB (9184): cx24116: Change the default SNR units back to percentage by default.
V4L/DVB (9183): S2API: Return error of the caller provides 0 commands.
V4L/DVB (9182): S2API: Added support for DTV_HIERARCHY
V4L/DVB (9181): S2API: Add support fot DTV_GUARD_INTERVAL and DTV_TRANSMISSION_MODE
V4L/DVB (9180): S2API: Added support for DTV_CODE_RATE_HP/LP
V4L/DVB (9179): S2API: frontend.h cleanup
V4L/DVB (9178): cx24116: Add module parameter to return SNR as ESNO.
V4L/DVB (9177): S2API: Change _8PSK / _16APSK to PSK_8 and APSK_16
V4L/DVB (9176): Add support for DvbWorld USB cards with STV0288 demodulator.
V4L/DVB (9175): Remove NULL pointer in stb6000 driver.
V4L/DVB (9174): Allow custom inittab for ST STV0288 demodulator.
V4L/DVB (9173): S2API: Remove the hardcoded command limit during validation
V4L/DVB (9172): S2API: Bugfix related to DVB-S / DVB-S2 tuning for the legacy API.
V4L/DVB (9171): S2API: Stop an OOPS if illegal commands are dumped in S2API.
V4L/DVB (9170): cx24116: Sanity checking to data input via S2API to the cx24116 demod.
V4L/DVB (9169): uvcvideo: Support two new Bison Electronics webcams.
V4L/DVB (9168): Add support for MSI TV@nywhere Plus remote
V4L/DVB: v4l2-dev: remove duplicated #include
...
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
ALSA: ASoC codec: remove unused #include <version.h>
ALSA: ASoC: update email address for Liam Girdwood
ALSA: hda: corrected invalid mixer values
ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
ALSA: ASoC: Add destination and source port for DMA on OMAP1
ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
ALSA: ASoC: Fix build of GTA01 audio driver
ALSA: ASoC: Add widgets before setting endpoints on GTA01
ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
ALSA: ASoC: Make TLV320AIC26 user-visible
ALSA: ASoC - clean up Kconfig for TLV320AIC2
ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
ALSA: ASoC: Implement WM8510 bias level control
ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
ALSA: ASoC: Add WM8510 SPI support
ALSA: ASoC: Add WM8753 SPI support
...
Add a new API call snd_soc_dapm_nc_pin() which allows machine drivers to
mark pins as being permanently disabled. At present this is identical
to snd_soc_dapm_disable_pin() except in terms of improving the internal
documentation of machine drivers that use it. The intention is that in
future it will be extended to provide additional features such as hiding
controls that are only relevant to paths using the disconnected pin.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the video_exclusive_open/release functionality into the
driver itself.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Increase the card components[] (and thus snd_card_info.components[],
too) array size from 80 to 128 chars so that more strings can be
stored. The 80 chars aren't enough for more than 2 HD-audio codecs,
and this hits an ugly snd_BUG() as reported by Wu Fegguang for HP
2230s.
The control protocol number is increased to 2.0.6 as well, in case
it matters.
Reported-by: Wu Fengguang <wfg@linux.intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We have some arithmetic operations against snd_pcm_hw_param_t, thus
bitwise isn't correct for it. Better to remove the flag to shut up
sparse warnings.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC and non-ASoC drivers for PCM DMA on PXA share lots of common code.
Move it to pxa2xx-lib.
[Fixed some checkpatch warnings -- broonie]
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC and non-ASoC drivers for ACLINK on PXA share lot's of common code.
Move all common code into separate module snd-pxa2xx-lib.
[Fixed handing of SND_AC97_CODEC in Kconfig and some checkpatch warnings
-- broonie]
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Empty files remained likely due to wrong patching.
Remove them now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- more register naming work
- finally figured out that weird CR register stuff
(and did I mention that I hate _really_ undecipherable open-coded values?)
- fix handling of IRQ sharing in interrupt handler
(hopefully properly, otherwise I'd be grateful to hear your
pedantic comments ;)
- add handy SPECS_PAGE references wherever useful
- comments, cleanup
- add me as module author
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in other places, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/core/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Introduced snd_BUG_ON() macro as a replacement of snd_assert() macro.
snd_assert() is pretty ugly as it has the control flow in its argument.
OTOH, snd_BUG_ON() behaves like a normal conditional, thus it's much
easier to read the flow.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss detection code and kill the ad1848 library.
The library is fully assimilated into the new wss library.
This required reworking of the AD1848 family code
so the code is changed to correctly detect chips from
the AD1848 and CS4231 families.
I have tested it on following cards:
Gallant SC-6600 (codec: AD1848, driver: snd-sc6600)
SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape)
SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy)
Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236)
Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330)
Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232)
Sound playback and recording works on all these cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss pcm code and kill the ad1848 pcm code.
The AD1848 chip is much slower than CS4231 chips
so the waiting loop was increased 100x (10x is not
enough).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss mixer code and kill the ad1848 mixer code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use CS4231P instead of AD1848P (kill the AD1848P).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss macros instead of ad1848 ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use wss constants for mode.
Move ad1848 hardware constants to the wss.h.
Move mixer tlv macros into the ad1848_lib.c from the ad1848.h.
Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem
to be needed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_wss is superset of the snd_ad1848 so kill
the latter and replace it with the snd_wss.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename functions and structures from the former
cs4321_lib to names more corresponding with the
new name: wss_lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename file include/sound/cs4231.h
into include/sound/wss.h
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices.
Additionally the device field of struct snd_pcm had to be changed to int,
as its "unsignednity" caused a lot of problems when comparing it to
potentially negative signed values. (-1 is 0xffffffff or even more then ;-)
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new US122L usb-audio driver. This driver works together with a
dedicated alsa-lib plugin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit fb3d6f2b77bdec75d45aa9d4464287ed87927866.
New, updated patch with same subject replaces this commit.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Updated IEC958 consumer status channel definitions according
to the third edition of IEC60958-3 spec.
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices. The same limit was present in OSS emulation code. It has
been fixed as well.
Additionally the device field of struct snd_pcm is now int, instead of
unsigned int, as there is no obvious reason for keeping it unsigned.
This caused a lot of problems with comparing this value with other
(almost always signed) variables. There is just one more place where
device number is unsigned - in struct snd_pcm_info, which should be
also sorted out in future.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASOC: convert use of uint to unsigned int
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The OpenFirmware API headers don't build on all platforms so ensure
that they are not included unless they are being used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Simple utility layer for creating ASoC machine instances based on data
in the OpenFirmware device tree. OF aware platform drivers and codec
drivers register themselves with this framework and the framework
automatically instantiates a machine driver. At the moment, the driver
is not very capable and it is expected to be extended as more features
are needed for specifying the configuration in the device tree.
This is most likely temporary glue code to work around limitations in
the ASoC v1 framework. When v2 is merged, most of this driver will
need to be reworked.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most of the ASoC controls refer to the maximum value that can be set for
a control as mask but there is no actual requirement for all bits to be
set at the highest possible value making the name mask misleading.
Change the code to use max instead.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert bitfields in ASoC into full int width. This is a
simple mechanical conversion. Two places in the DAPM code
were fixed to properly use mask.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some codecs have unusual features in their register maps such as very
large registers representing arrays of coefficients. Support these
codecs in the register cache sysfs file by allowing them to provide a
function register_display() overriding the default output for register
contents.
Also ensure that we don't overflow PAGE_SIZE while writing out the
register dump.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently very few systems provide information about jack status to user
space, even though many have hardware facilities to do detection. Those
systems that do use an input device with the existing SW_HEADPHONE_INSERT
switch type to do so, often independently of ALSA.
This patch introduces a standard method for representing jacks to user
space into ALSA. It allows drivers to register jacks for a sound card with
the input subsystem, binding the input device to the card to help user
space associate the input devices with their sound cards. The created
input devices are named in the form "card longname jack" where jack is
provided by the driver when allocating a jack. By default the parent for
the input device is the sound card but this can be overridden by the
card driver.
The existing user space API with SW_HEADPHONE_INSERT is preserved.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to
be exported to support building them as modules and prototyped to avoid
sparse warnings and potential build issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new ALSA driver for the audio device found inside
most of the SGI O2 workstation. The hardware uses a SGI custom chip,
which feeds a AD codec chip.
Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
A bunch of things in alsa depend on CONFIG_KMOD,
use CONFIG_MODULES instead where the dependency
is needed at all.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.
This merger removes duplication in both DAI structures and simplifies
the API for other users.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This generic register modifier widget is for updating multiple codec
register bits at once when the widget changes its power state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit 36b34d2437104f323e09d7c6af6451d3c0b9c0cd.
From: Al Viro <viro@ZenIV.linux.org.uk>
WIW, *all* this stuff is not bitwise at all. For crying out loud, half
of these types are routinely used as array indices and loop variables...
If anything, we want a different set of allowed operations - subtraction
between elements of type (yielding integer), addition/subtraction of
integer types not bigger than ours (yielding our type), comparisons,
assignments (=, +=, -=, passing to function as argument, return from
function, initializers) and second/third arguments in ?:. With 0 *not*
being allowed as a constant of such type.
It's not bitwise; we may use the same infrastructure in sparse, but it
should be a separate class of types (__attribute__((affine))).
dma_addr_t is another candidate for the same treatment, but there we'll
need helpers for conversions to hw-acceptable form (dma_to_le32(), etc.)
and gradual conversion of drivers.
ALSA ones and pm mess are absolutely straightforward cases, though.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fully half of all alsa sparse warnings are from snd_pcm_hw_param_t degrading
to integer type, this goes a long way towards eliminating them.
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On OpenMoko soc-audio resume is taking 700ms of the whole resume time of
1.3s, dominated by writes to the codec over I2C. This patch shunts the
resume guts into a workqueue which then is done asynchronously.
The "card" is locked using the ALSA power state APIs as suggested by
Mark Brown.
[Added fix for race with resume to suspend and fixed a couple of nits
from checkpatch -- broonie.]
Signed-off-by: Andy Green <andy@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd_ctl_elem_read() and snd_ctl_elem_write() are no longer used by
any other drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mike Montour <mail@mmontour.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds support for WSS compatible Opti93x
codec to the cs4231-lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Tested-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On Audigy2 Platinum, the Analog/Digital mixer switch is inverted.
https://bugzilla.novell.com/show_bug.cgi?id=396204
The patch adds a simple workaround.
There might be another device requiring a similar fix, too (or fix for
audigy2 generically), but right now I fix only the known broken one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE to
represent its meaning more better. This config isn't provided only
for the detection but for more verbose debug prints in general.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes CVS keywords that weren't updated for a long time
from comments.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.
Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.
Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The CPU and codec DAI operations differ only in the presence of the
digital mute operation for the codec so they may as well be the same
type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC codecs and machine drivers that use DAPM routes all cut'n'paste a
loop iterating over a null terminated array of routes. Factor out this
into a bulk registration function, improving the error reporting for
most users, and deprecate the old API to help out of tree users pick up
the changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most SoC drivers cut'n'paste a loop iterating over an array to register
their DAPM controls. Provide a function they can call instead.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This adds a hook to read the power state of a DAPM widget, I use this
in the gta02 driver to expose certain DAPM widgets in the mixer for
ease of audio routing.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I suspect that snd_ctl_boolean_mono should have been
snd_ctl_boolean_mono_info instead. This fixes the build for magician.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_minor_info_oss_* is an function returning int _or_ comment,
depending on config parameters. That is truly evil, fix it.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two sentences seem to be spliced into one in comment, fix that and fix
english. Also fix codingstyle.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* removing the hack with NON_AKM ak4xxx type
* support for card-specific flags in ak4114_stats
* definition of the flags for corresponding cards
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added definition for byte 4 of SPDIF channel status, according to
second edition of IEC 60958-3 (consumer) spec.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the codes for virtual master controls to sound core part so that
not only hda-intel drivers can use it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of 8 channel sound for codecs that are known to work.
So far, only ALC850 is marked as a 8ch-support codec.
This fix is a modified version of the patch on ALSA BTS#2097 by
Martin Ellis:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper ifdef's to the patch loading code moved from the old instr
layer so that opl3 driver can be compiled without the sequencer support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This patch improves E-Mu 1616(M) cardbus support. It adds definitions of the
new Microdock and 1010 cardbus registers (thanks again for descriptions
James) and improves mixer for this card. Now you can use S/PDIF and ADAT on
Mirodock and also use headpohone output on host cardbus card as another
independent output.
Signed-off-by: Ctirad Fertr <c.fertr@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is improvement of the early support of the FM-only cards where the
fm801 chip represents the PCI to tuner bridge.
The tuner initialization isn't included the mute on as well as mute support
via V4L request. Proposed patch should fix this at least for 64-PCR model.
Signed-off-by: Andy Shevchenko <andy@smile.org.ua>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Change semantics for SNDRV_PCM_TSTAMP_MMAP. Doing timestamping only in
the interrupt handler might cause that hw_ptr is not related to actual
timestamp. With this change, grab timestamp at every hw_ptr update to
have always valid timestamp + ring buffer position pair.
With this change, SNDRV_PCM_TSTAMP_MMAP was renamed to
SNDRV_PCM_TSTAMP_ENABLE. It's no regression (I think).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This header file exists only for some hacks to adapt alsa-driver
tree. It's useless for building in the kernel. Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it. This should be really killed in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The 'tick' in PCM is set (again) via sw_params. And, nobody uses
this feature at all except for a command line option of aplay.
(This is literally 'nobody', as I checked alsa-lib API calls in all
programs in major distros.)
Above all, if we need finer wake-ups for the position update, it's
basically an issue that the driver should solve, not tuned by each
application.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The xfer_align sw_params parameter has never been used in a sane manner,
and no one understands what this does exactly. The current
implementation looks also buggy because it allows write of shorter size
than xfer_align. So, if you do partial writes, the write isn't actually
aligned at all.
Removing this parameter will make some pcm_lib_* code more readable
(and less buggy).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch removes the indirect control access to the control elements.
The indirect access has never been used and is even broken on 32bit
ioctl wrapper. Let's clean it up.
The pointers still remain in snd_ctl_elem_* structs just to make sure
that the struct size won't change. Once after checking the size
consistency, we can get rid of them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
info_oss: move prototype of snd_card_info_read_oss to info.h
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We need an accurate and continuous (monotonic) time sources to do
accurate synchronization among more timing sources. This patch allows
to enable monotonic timestamps for ALSA PCM devices and enables monotonic
timestamps for ALSA timer devices.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* support for switching rate in STAC9460 - using set_rate_val of the akm
infrastructure
* listing all STAC9460 registers in proc
* disabling mpu401 device for Prodigy192 - otherwise the currently
flawed mpu401 code hangs kernel when opening the midi device
* removing old unused commented-out code
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
remove dead config symbols from sound code
Signed-off-by: Jiri Olsa <olsajiri@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Updated the forgotten SNDRV_HWDEP_IFACE_LAST to point the really last member.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Remove sequencer instrument layer from the tree.
This mechanism hasn't been used much with the actual devices. The only
reasonable user was OPL3 loader, and now it was rewritten to use hwdep
instead. So, let's remove the rest of rotten codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the exclusive access lock in hwdep instead of the own one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the hwdep device for loading OPL2/3 patch data instead of the
messy sequencer instrument layer.
Due to this change, the sbiload program should be updated, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
open and close operations are called only from pcm layer
and mutexed there with pcm->open_mutex.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Since the last patch made the ENTER_UART command optional, the
enter_uart option and its corresponding flag have become superfluous.
The uart_enter option remains for backward compatibility but just prints
a warning when used.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
These patches enable some YMF743 controls (Tone/3D/IEC958) that won't
be detected with the current version of ALSA.
The first one contains only cosmetic changes to share a few
YMF753-specific symbols with YMF743.
Signed-off-by: Keita Maehara <maehara@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The volatile prefix is just useless there. Let's kill them, and then
gcc will be happier, too.
sound/acore/pcm.c:867: warning: passing argument 1 of ‘__constant_c_and_count_memset’ discards qualifiers from pointer target type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed cast messes in pcm.h.
include/sound/pcm.h: In function ‘hw_param_interval_c’:
include/sound/pcm.h:800: warning: passing argument 1 of ‘hw_param_interval’ discards qualifiers from pointer target type
Simply redefine the inline functions again for const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Consistent variable naming is a good thing, but let's be a little less
sneaky about enforcing it... ;-/
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch splits the cs4231.h file into two parts:
- cs4231-regs.h which contain register constants and macros
- cs4231.h which includes the above and contain rest of the definitions
This will allow to share register definitions between x86 ISA cs4231
and SPARC cs4231.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add a snd_pcm_rate_to_rate_bit() function to factor out common code used
by several drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Merge the rates[] arrays from pcm_misc.c and pcm_native.c because they
are both the same.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a hwdep interface for each codec (enabled per kconfig).
This interface can be used for reading/writing HD-audio verbs
and other purposes as future extensions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix codes to follow more to the standard kernel coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The mode change / recalibration doesn't work always with opl3sa2 devices,
e.g. the first time it's played back. The patch fixes the problem.
Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clean up codes using the new common snd_ctl_boolean_*_info() callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added helper functions for frequenty used callbacks:
snd_ctl_boolean_mono_info() and snd_ctl_boolean_stereo_info()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Notebook.
Description: The .device=0x0008 chips have new, but different EMU32 in/out
channels. Driver updated to make use of these EMU32 channels.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed PM resume of cs46xx devices. It now restores properly the DSP
image and kick-off the DSP.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* adding 8 more 32-bit capture channels (total of 16) for emu1010 cards
* adding some code comments and card details description
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add support for Cyrix/NatSemi Geode SC5530 (VSA1).
The driver is snd-cs5530.
Signed-off-by Ash Willis <ashwillis@programmer.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the support of mute for front channels of M-Audio
Revolution 7.1 (the DAC AK4381 features a mute bit).
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes a bug whereby AC97 bus device data was being clobbered
when AC97 codecs using the generic ac97_codec.c driver were being
registered. Codecs that didn't use the generic driver were unaffected
(e.g. WM9712, WM9713).
Changes:-
o Add new AC97 codec class for custom (or need bus dev registration)
AC97 codecs.
o Only register/deregister this custom codec device with the AC97 bus.
The generic AC97 driver already does this for generic codec devices.
This may be related to bug #3038 :-
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* correct register for 'IEC958 Non-PCM Bitstream', 'IEC958 DTS Bitstream'
to use AK4114_REG_RCS0
* correct check for control name: if (strstr(kctl->id.name, 'Playback'))
* correct check: if (!chip->init) in snd_ak4114_external_rate
* added PCM control 'IEC958 PPL Lock Status'
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed a typo in AK4114_DIF2 bit definition. This may fix some
problems for Audiophile 192 and Juli boards.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added MPU401_INFO_UART_ONLY bitflag to avoid issueing UART_ENTER command
at opening streams. Some devices support only UART mode and give errors
to UART_ENTER.
A new module option, uart_enter, is added to snd-mpu401 driver.
For UART-only devices, set uart_enter=0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a new macro snd_pcm_group_for_each_entry() just for code cleanup.
Old macros, snd_pcm_group_for_each() and snd_pcm_group_substream_entry(),
are removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Many struct file_operations in the kernel can be "const". Marking them const
moves these to the .rodata section, which avoids false sharing with potential
dirty data. In addition it'll catch accidental writes at compile time to
these shared resources.
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The fix for sysfs breakage with CONFIG_SYSFS_DEPRECATED was flown
away by the conflicted merge of the ALSA git tree. The patch below
fixes it again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch updates the API's to include the new DAI configuration and
clocking architecture.
Changes:-
o Removed DAI automatic matching and capabilities structure (struct
snd_soc_dai_mode) and macros.
o Added DAI operations for codec and CPU interfaces.
o Removed config_sysclk() function and struct snd_soc_clock_info. No
longer needed as clocking is now configured manually in the machine
drivers. Also removed other clocking data from structures.
o Updated version to 0.13
o Added shift to SOC_SINGLE_EXT kcontrol macro.
Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Mark TLV data as 'const'
Signed-of-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Make data passed to ak4xxx_create 'const'.
Signed-of-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix ABI for older ld10k1. When no EMU10K1_PVERSION ioctl is issued,
the driver accepts ioctls with the old struct size without TLV information.
Also, changed the struct field to make the conversion easier from the
old to the new structs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Previously, ac97_codec.c was coded to support AD1986 and AD1986A
CODECs using code written for the AD1985 CODEC. This allowed the
LINE_OUT and HEADPHONE jacks to function properly, however register
differences between the CODECs prevented line and microphone inputs
from functioning.
Specifically, this patch fixes issues with the following mixer
controls: 'V_REFOUT', 'Spread Front to Surround and Center/LFE',
'Exchange Front/Surround', 'Surround Jack Mode', and 'Channel Mode'.
This patch removes the undocumented AD1888 control
'High Pass Filter Enable' and adds the new control
'Exchange Mic/Line In'.
Signed-off-by: Randy Cushman <rcushman_linux@earthlink.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix the changes realted to delayed_work in soc/codecs/wm8750.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use global workqueue for simplicity instead of own workqueue.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
From: Andrew Morton <akpm@osdl.org>
I converted the workqueues to per-device while I was there. It seems
strange to create a new kernel thread (on each CPU!) and to then only
have a single global work to ever be queued upon it.
Plus without this, I'd have to use the _NAR stuff, gawd help me.
Does that workqueue really need to be per-cpu?
Does that workqueue really need to exist? Why not use keventd?
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Implement functionallity in order to fixe ALSA bug#2058.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Following patch will make the driver to use the 44.1kHz SRC automatically
if the pcm source is 44.1kHz signed 16bit stereo.
The SRC is available in YMF754 only.
Signed-off-by: Teru KAMOGASHIRA <teru@sodan.ecc.u-tokyo.ac.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a helper function snd_pci_quirk_lookup()
to look up PCI SSID quirk list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Removed obsolete typedefs.h. It existes only for backward compatibility,
and now all codes should be free from such typedefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Now that everyone uses snd_ctl_new1() and noone is using snd_ctl_new()
anymore, we can make it static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Instead of using a somewhat algorithmic approach of initializing the
YSS225's registers, just use a simple series of port/value pairs.
This makes it easier to later replace or entirely remove the register
data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the CSP programs using request_firmware(), if possible, instead of
using the built-in firmware blobs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Don't enable power-saving mode on drivers that don't support
it. The supporting drivers set AC97_SCAP_POWER_SAVE to scaps
at creation of ac97 instance.
Currently enable on the following drivers: intel8x0, intel8x0m,
atiixp, atiixp-modem, via82xx and via82xx-modem.
Also, a bit clean up of power-saving stuff:
- Don't create an own workq
- Remove superfluous ifdefs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the DSP and controller microcode using request_firmware(), if
possible, instead of using the built-in firmware.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the ASoC and DAPM headers.
Features:-
o Defines Digital Audio Interface (DAI) API
o Defines Codec, Platform and Machine API
o Defines Dynamic Audio Power Management API
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility. This allows
to control the analog loopback of the card.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility. Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The recent change for a new sysfs tree with card* object breaks the
/sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled.
The device in each entry doesn't point the correct device object:
/sys/class/sound
...
|-- pcmC0D0c
| |-- dev
| |-- device -> ../../../class/sound/card0
| |-- pcm_class
| |-- power
| | `-- wakeup
| |-- subsystem -> ../../../class/sound
| `-- uevent
Also, this change breaks some drivers (like sound/arm/*) referring
card->dev directly to obtain the device object for memory handling.
This patch reverts the semantics of card->dev to the former version,
which points to a real device object. The card* object is stored in a
new card->card_dev field, instead. The device parent is chosen either
card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to
keep the tree compatibility.
Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled. The
reason of card* object is a root of all beloing devices, and it makes
little sense if each sound device points to the real device object
directly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Monty Montgomery <xiphmont@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Fix the type of PCI revision to char from int and avoid invalid
assignment with pointer cast.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed the race among multiple threads accessing the OSS PCM
instance concurrently by simply introducing a mutex for protecting
a setup of the PCM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes incorrect assignment of swap_rear,
which was broken since patch 'ymfpci - make rear channel swap optional'
It removes module_param rear_swap.
Signed-off-by: Glen Masgai <mimosius@gmx.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Conflicts:
drivers/ata/libata-scsi.c
include/linux/libata.h
Futher merge of Linus's head and compilation fixups.
Signed-Off-By: David Howells <dhowells@redhat.com>
Conflicts:
drivers/infiniband/core/iwcm.c
drivers/net/chelsio/cxgb2.c
drivers/net/wireless/bcm43xx/bcm43xx_main.c
drivers/net/wireless/prism54/islpci_eth.c
drivers/usb/core/hub.h
drivers/usb/input/hid-core.c
net/core/netpoll.c
Fix up merge failures with Linus's head and fix new compilation failures.
Signed-Off-By: David Howells <dhowells@redhat.com>
Converts from using struct "class_device" to "struct device" making
everything show up properly in /sys/devices/ with symlinks from the
/sys/class directory.
It also makes the struct sound_card to show up as a "real" device
where all the different sound class devices are placed as childs
and different card attribute files can hang off of. /sys/class/sound is
still a flat directory, but the symlink targets of all devices belonging
to the same card, point the the /sys/devices tree below the new card
device object.
Thanks to Kay for the updates to this patch.
Signed-off-by: Kay Sievers <kay.sievers@novell.com>
Acked-by: Jaroslav Kysela <perex@suse.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Alsa used to kmalloc one file->f_op per file per disconnecting snd_card.
This led to oopses sometimes when file->f_op was freed before __fput()
finished.
Patch adds a virtual device for disconnect: VDD.
VDD consists of:
LIST_HEAD(shutdown_files)
protected by DEFINE_SPINLOCK(shutdown_mutex)
static struct file_operations snd_shutdown_f_ops
and functions assigned to it
Additions to struct snd_monitor_file
to specify if instance is hidden by VDD or not.
A VDD's instance is
created in snd_card_disconnect() under the card->files_lock.
cleaned up in snd_card_file_remove() under the card->files_lock.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Maintain a per-CPU global "struct pt_regs *" variable which can be used instead
of passing regs around manually through all ~1800 interrupt handlers in the
Linux kernel.
The regs pointer is used in few places, but it potentially costs both stack
space and code to pass it around. On the FRV arch, removing the regs parameter
from all the genirq function results in a 20% speed up of the IRQ exit path
(ie: from leaving timer_interrupt() to leaving do_IRQ()).
Where appropriate, an arch may override the generic storage facility and do
something different with the variable. On FRV, for instance, the address is
maintained in GR28 at all times inside the kernel as part of general exception
handling.
Having looked over the code, it appears that the parameter may be handed down
through up to twenty or so layers of functions. Consider a USB character
device attached to a USB hub, attached to a USB controller that posts its
interrupts through a cascaded auxiliary interrupt controller. A character
device driver may want to pass regs to the sysrq handler through the input
layer which adds another few layers of parameter passing.
I've build this code with allyesconfig for x86_64 and i386. I've runtested the
main part of the code on FRV and i386, though I can't test most of the drivers.
I've also done partial conversion for powerpc and MIPS - these at least compile
with minimal configurations.
This will affect all archs. Mostly the changes should be relatively easy.
Take do_IRQ(), store the regs pointer at the beginning, saving the old one:
struct pt_regs *old_regs = set_irq_regs(regs);
And put the old one back at the end:
set_irq_regs(old_regs);
Don't pass regs through to generic_handle_irq() or __do_IRQ().
In timer_interrupt(), this sort of change will be necessary:
- update_process_times(user_mode(regs));
- profile_tick(CPU_PROFILING, regs);
+ update_process_times(user_mode(get_irq_regs()));
+ profile_tick(CPU_PROFILING);
I'd like to move update_process_times()'s use of get_irq_regs() into itself,
except that i386, alone of the archs, uses something other than user_mode().
Some notes on the interrupt handling in the drivers:
(*) input_dev() is now gone entirely. The regs pointer is no longer stored in
the input_dev struct.
(*) finish_unlinks() in drivers/usb/host/ohci-q.c needs checking. It does
something different depending on whether it's been supplied with a regs
pointer or not.
(*) Various IRQ handler function pointers have been moved to type
irq_handler_t.
Signed-Off-By: David Howells <dhowells@redhat.com>
(cherry picked from 1b16e7ac850969f38b375e511e3fa2f474a33867 commit)
Add maximum latency tracking to the ALSA subsystem for PCM playback. In
ALSA, the playback application controls the buffer size and thus indirectly
the period of latency that it can deal with. This patch uses 75% of the
total available latency as threshold to announce to the latency subsystem;
While 75% is a crude heuristic it's a quite reasonable one; the remaining
25% can be used for all driver processing for the next samples which is
also proportional to the size of the buffer.
With ogg123 a latency setting of about 4msec was seen (at 44Khz), while
with the "play" command a much longer maximum tolerable latency was seen.
Other, more multimedia oriented players as well as games, will have a lot
smaller buffers to allow better synchronization and those will actually get
into the latency domains where there is impact on the power management
rules.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Remove IPGA volume controls and merge the IPGA range to ADC volume
controls. These two volumes are not really independent but connected
simply in different ranges 0-0x7f and 0x80-max. It doesn't make sense
to provide two controls.
Since both 0x7f and 0x80 specify 0dB, a hack is needed for IPGA range
to skip 0x80 (increment one) for such controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a new attribute, pcm_class, to each PCM sysfs entry.
It's useful to detect what kind of PCM stream is, for example, HAL
can check whether it's a modem or not.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the definition of TLV dB range compound. It contains one or
more dB-range or linear-volume TLV entries with min/max ranges.
Used for volume controls with non-linear curves.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
- Clean up the code in AK4xxx-ADDA i2c code.
- Fix capture gain controls for AK5365
- Changed the static table for DAC/ADC mixer labels to use
structs
- Implemented TLV entries for each AK codec
The volumes in AK4524, AK4528 and AK5365 are corrected with
a table to be suitable for dB conversion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the definition of linear volume TLV type.
Some DSP chips and codecs (e.g. AK codec) use linear volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the dB scale information to vxpocket and vx222 drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add support for the AK5365 ADC.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The flag to avoid 32bit-incompatible mmap for control/status records
should be outside the pcm substream instance since a substream can be
shared among multiple opens. Now it's flagged in pcm_file list that
is directly assigned to file->private_data.
Also, removed snd_pcm_add_file() and remove_file() functions and
substream.files field that are not really used in the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remove unused tlv_rw field from struct snd_kcontrol. The callback is
set in tlv.c field, instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added CONFIG_SND_AC97_POWER_SAVE kernel config to enable the support
of aggressive AC97 power-saving mode. In this mode, the AC97
powerdown register bits are dynamically controlled at each open/close
of PCM streams.
The mode is activated via power_save option for snd-ac97-codec
driver. As default it's off. It can be turned on/off on the fly
via sysfs, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Orignally proposed by Sam Revitch <sam.revitch@gmail.com>.
Unregister device files at disconnection to avoid the futher accesses.
Also, the dev_unregister callback is removed and replaced with the
combination of disconnect + free.
A new function snd_card_free_when_closed() is introduced, which is
used in USB disconnect callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
- Add the linked list to each proc entry to enable a single-shot
disconnection (unregister)
- Deprecate snd_info_unregister(), use snd_info_free_entry()
- Removed NULL checks of snd_info_free_entry()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch implements a TLV mechanism to transfer an additional information
like dB scale to the user space. The types might be extended in future.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed 'section mismatch' errors in ALSA PCI drivers:
- removed invalid __devinitdata from pci id tables
- fix/remove __devinit of functions called in suspend/resume
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Drop the snd_minor structure's name field that was just a helper for
devfs device deregistration.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix spaces, fold lines to fit 80 columns in ak4xxx-adda driver codes.
Split a long reset function to each codec routine just for better
readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds stereo controls to revo cards by making the ak4xxx
driver mixers configurable from the card driver.
Signed-off-by: Jani Alinikula <janialinikula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds two mixer controls. The V_REFOUT enable is a
documented register that couples the microphone input lines
to the V_REFOUT DC source. The High Pass Filter enable in the
AC97_AD_TEST2 (0x5c) is an undocumented register provided by
Miller Puckette via Analog Devices that enables the AD codec
to apply a high pass filter to the input.
Signed-off-by: Jaya Kumar <jayakumar.alsa@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Suppress 'irq handler mismatch' messages at auto-probing of irqs
in ALSA ISA drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
In file included from sound/i2c/other/tea575x-tuner.c:30:
include/sound/tea575x-tuner.h:36: error: field 'vd' has incomplete type
include/sound/tea575x-tuner.h:37: error: field 'fops' has incomplete type
sound/i2c/other/tea575x-tuner.c:89: warning: 'struct inode' declared inside parameter list
sound/i2c/other/tea575x-tuner.c:89: warning: its scope is only this definition or declaration, which is probably not what you want
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_ioctl':
sound/i2c/other/tea575x-tuner.c:91: warning: implicit declaration of function 'video_devdata'
sound/i2c/other/tea575x-tuner.c:91: warning: initialization makes pointer from integer without a cast
sound/i2c/other/tea575x-tuner.c:92: warning: implicit declaration of function 'video_get_drvdata'
sound/i2c/other/tea575x-tuner.c:92: warning: initialization makes pointer from integer without a cast
sound/i2c/other/tea575x-tuner.c:96: warning: implicit declaration of function '_IOR'
sound/i2c/other/tea575x-tuner.c:96: error: syntax error before 'struct'
sound/i2c/other/tea575x-tuner.c:99: error: 'v' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:99: error: (Each undeclared identifier is reported only once
sound/i2c/other/tea575x-tuner.c:99: error: for each function it appears in.)
sound/i2c/other/tea575x-tuner.c:108: warning: implicit declaration of function 'copy_to_user'
sound/i2c/other/tea575x-tuner.c:112: warning: implicit declaration of function '_IOWR'
sound/i2c/other/tea575x-tuner.c:112: error: syntax error before 'struct'
sound/i2c/other/tea575x-tuner.c:115: warning: implicit declaration of function 'copy_from_user'
sound/i2c/other/tea575x-tuner.c: At top level:
sound/i2c/other/tea575x-tuner.c:129: error: syntax error before 'case'
sound/i2c/other/tea575x-tuner.c:146: warning: type defaults to 'int' in declaration of 'snd_tea575x_set_freq'
sound/i2c/other/tea575x-tuner.c:146: warning: parameter names (without types) in function declaration
sound/i2c/other/tea575x-tuner.c:146: error: conflicting types for 'snd_tea575x_set_freq'
sound/i2c/other/tea575x-tuner.c:62: error: previous definition of 'snd_tea575x_set_freq' was here
sound/i2c/other/tea575x-tuner.c:146: warning: data definition has no type or storage class
sound/i2c/other/tea575x-tuner.c:147: error: syntax error before 'return'
sound/i2c/other/tea575x-tuner.c:151: error: syntax error before '&' token
sound/i2c/other/tea575x-tuner.c:152: error: syntax error before '.' token
sound/i2c/other/tea575x-tuner.c:152: warning: type defaults to 'int' in declaration of 'strcpy'
sound/i2c/other/tea575x-tuner.c:152: warning: function declaration isn't a prototype
sound/i2c/other/tea575x-tuner.c:152: error: conflicting types for 'strcpy'
sound/i2c/other/tea575x-tuner.c:152: warning: data definition has no type or storage class
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_init':
sound/i2c/other/tea575x-tuner.c:194: warning: implicit declaration of function 'video_set_drvdata'
sound/i2c/other/tea575x-tuner.c:197: error: 'video_exclusive_open' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:198: error: 'video_exclusive_release' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:200: warning: implicit declaration of function 'video_register_device'
sound/i2c/other/tea575x-tuner.c:200: error: 'VFL_TYPE_RADIO' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_exit':
sound/i2c/other/tea575x-tuner.c:215: warning: implicit declaration of function 'video_unregister_device'
distcc[7333] ERROR: compile sound/i2c/other/tea575x-tuner.c on x/32 failed
make[1]: *** [sound/i2c/other/tea575x-tuner.o] Error 1
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
This patch by Rodolfo Giometti disables the AC97 AUX and VIDEO controls
on the WM9705 when the touchscreen is selected as the AUX and VIDEO
lines are shared with the touch controller.
Changes:-
o Added AC97_HAS_NO_AUX flag
o Test for AC97_HAS_NO_AUX flag in snd_ac97_mixer_build()
o Sets AC97_HAS_NO_VIDEO and AC97_HAS_NO_AUX in patch_wolfson05() when
WM9705 touch driver is selected.
Signed-off-by: Rodolfo Giometti <giometti@linux.it>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the 5th argument of snd_mpu401_uart_new() to bit flags
instead of a boolean. The argument takes bits that consist of
MPU401_INFO_XXX flags.
The callers that used the value 1 there are replaced with
MPU401_INFO_INTEGRATED.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed rwlock around snd_iprintf() in sound core part.
Replaced with mutex.
Also, make mutex and flags static variables with addition of
snd_card_locked() function (just for sound.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a get_port_info callback to the snd_rawmidi_global_ops structure to
allow the USB MIDI driver to supply information flags for the sequencer
ports created by seq_midi.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add four new information flags SNDRV_SEQ_PORT_TYPE_HARDWARE, _SOFTWARE,
_SYNTHESIZER, _PORT for sequencer ports. This makes it easier for apps
like Rosegarden to make policy decisions based on the port type.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Move mmap_count to snd_pcm_substream instead of runtime struct
so that multiplly opened substreams via O_APPEND can be handled
correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added O_APPEND flag support to PCM to enable shared substreams
among multiple processes. This mechanism is used by dmix and
dsnoop plugins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move EXPORT_SYMBOL()s to places adjacent to functions/variables.
Also move OSS-specific hw_params helper functions to pcm_oss.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds the Kbuild files listing the files which are to be installed by
the 'headers_install' make target, in generic directories.
Signed-off-by: David Woodhouse <dwmw2@infradead.org>
Fixed Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n.
Add ifdef to struct fields for optimization and better compile
checks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: PCM Midlevel
This patch makes the needlessly global snd_pcm_format_name() static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix possible race of referring the setup hook from the running PCM
- Fix memory leak in an error path of proc write
- Clean up the setup hook parser
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Clean up initialization and destruction of substream instance
Now snd_pcm_open_substream() alone does most initialization jobs.
Add pcm_release callback for cleaning up at snd_pcm_release_substream()
- Tidy up PCM oss code
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark the f_ops members of inodes as const, as well as fix the
ripple-through this causes by places that copy this f_ops and then "do
stuff" with it.
Signed-off-by: Arjan van de Ven <arjan@infradead.org>
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
OPL3_HW_OPL3_PC98 #define isn't used anywhere; previously in
sound/drivers/opl3/opl3_lib.c and sound/isa/cs423x/pc98.c, the latter of which
went away with the rest of PC98 subarch.
Signed-off-by: Arthur Othieno <apgo@patchbomb.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Modules: AC97 Codec
Add the pointer to a static volume resolution table to ac97 template,
so that the drivers can define the volume resolution, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: AC97 Codec
Added the support of static resolution table support for codecs
that the driver cannot probe the volume resolution properly.
The table pointer should be set in each codec patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: YMFPCI driver
Added rear_swap module option / kernel parameter to configure the rear
channel swapping. Default value is enable to make the AC3 passthrough
working, but analog only users might revert the previous behaviour.
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Semaphore to mutex conversion.
The conversion was generated via scripts, and the result was validated
automatically via a script as well.
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Semaphore to mutex conversion.
The conversion was generated via scripts, and the result was validated
automatically via a script as well.
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Semaphore to mutex conversion.
The conversion was generated via scripts, and the result was validated
automatically via a script as well.
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Semaphore to mutex conversion.
The conversion was generated via scripts, and the result was validated
automatically via a script as well.
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: ALSA Core,PCM Midlevel,ALSA<-OSS emulation,USB generic driver
1) The verbose procfs code for the PCM midlevel and usb audio
can be removed now (more patches will follow).
CONFIG_SND_VERBOSE_PROCFS
2) The PCM OSS plugin system can be also compiled optionaly.
CONFIG_SND_PCM_OSS_PLUGINS
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix typos in comments to remove kernel-doc warnings.
Signed-off-by: Martin Waitz <tali@admingilde.org>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Since version 4.1 the gcc is warning about ignored attributes. This patch is
using the equivalent attribute on the struct instead of on each of the
structure or union members.
GCC Manual:
"Specifying Attributes of Types
packed
This attribute, attached to struct or union type definition, specifies
that
each member of the structure or union is placed to minimize the memory
required. When attached to an enum definition, it indicates that the
smallest integral type should be used.
Specifying this attribute for struct and union types is equivalent to
specifying the packed attribute on each of the structure or union
members."
Signed-off-by: Jan Blunck <jblunck@suse.de>
Cc: Dave Jones <davej@codemonkey.org.uk>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Modules: ALSA Core
Revert the nested-device patch to keep the compatibility with the
current HAL configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: EMU10K1/EMU10K2 driver
Description:
Part way to fix ALSA bug#927
Add support for the SPI interface on the CA0108 chip.
This is used to control the registers on the DAC.
Headphone output tested.
Other outputs and Capture not tested yet.
Note: The red LED does not come on, but sound is still OK.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Modules: ALSA sequencer
Reduce the maximum possible number of global clients to 16 to make
more numbers available for card clients, and allow dynamically allocated
card client numbers to share the same range as application client
numbers to make sure that all 32 cards can be used at the same time.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
All users of snd_seq_create_kernel_client() have to set the client name
anyway, so we can just pass the name as parameter. This relieves us
from having to muck around with a struct snd_seq_client_info in these
cases.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The fields of struct snd_seq_client_callback either aren't used or are
always set to the same value, so we can get rid of it altogether.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: AC97 Codec,ATIIXP driver,Intel8x0 driver
This patch adds a new quirk for ac97 hardware that combines the existing
AC97_TUNE_MUTE_LED and AC97_TUNE_HP_ONLY quirks. This is needed for several
current HP laptops. Additionally, it adds the HP nx6125 to the
AC97_TUNE_MUTE_LED list.
Fixed for the latest version of ALSA by Takashi Iwai <tiwai@suse.de>.
Signed-off-by: Matthew Garrett <mjg59@srcf.ucam.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: EMU10K1/EMU10K2 driver
Distorted sound now comes from the Audio Out socket. Still more work to do.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Modules: ALSA Core,Memalloc module,ALSA sequencer
With dynamic minor numbers, we can increase the number of sound cards.
This requires that the sequencer client numbers of some kernel drivers
are allocated dynamically, too.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: ALSA Core,ALSA Minor Numbers
Add an option to allocate device file minor numbers dynamically.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Instead of storing the pointers to the device-specific structures in an
array, put them into the struct snd_minor, and look them up dynamically.
This makes the device type modules independent of the minor number
encoding.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: ALSA Core
Store the snd_minor structure pointers in one array instead of using a
separate list for each card. This simplifies the mapping from device
files to minor struct by removing the need to know about the encoding
of the card number in the minor number.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Instead of a comment string, store the device type in the snd_minor
structure. This makes snd_minor more flexible, and has the nice side
effect that we don't need anymore to create a separate snd_minor
template for registering a device but can pass the file_operations
directly to snd_register_device().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: ALSA Core,Control Midlevel,/oss/Makefile
Remove the centralized PM control in the sound core.
Each driver is responsible to get callbacks from bus/driver now.
SND_GENERIC_DRIVER is removed together with this action.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: ALSA Core
Backward-compatibility typedefs are stored in the new header, typedefs.h,
for out-of-tree drivers. This will be removed in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: AC97 Codec
Remove the definition of ac97_enum struct from the public ac97_codec.h.
It's used only in the module.
The location of struct ac97_pcm is moved closer to its accessor
to improve readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: EMU10K1/EMU10K2 driver
It appears that either the Audigy DMA engine or the Linux kernel cannot
handle 32 bit DMA with this device. Problem manifests as noise when
using more than 2GB of RAM, possibly only on 64 bit machines.
The OSS driver actually uses a 29 bit DMA mask for both devices, this
seems like overkill for now.
Signed-off-by: Lee Revell <rlrevell@joe-job.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: HWDEP Midlevel,PCM Midlevel,RawMidi Midlevel,ALSA Core
Replace usage of CONFIG_SND_MAJOR with snd_major, where appropriate.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: ALSA Core,ALSA Minor Numbers
Remove the unused and undefined symbols SNDRV_DEVICE_TYPE_{MIXER,
PCM_PLOOP,PCM_CLOOP}, and introduce a new symbol SNDRV_MINOR_GLOBAL
for non-card-specific devices like the sequencer or the timer.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modules: RTC timer driver,Timer Midlevel
Add a module pointer to the timer structure and use it for refcounting
instead of the card's module pointer to prevent the global timer
modules (rtctimer and hpetimer) from being removed while in use.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
- Remove vmalloc wrapper
- Add release_and_free_resource() to remove kfree_nocheck() from each driver
and simplify the code
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: ALSA Core,ALSA<-OSS emulation
Remove a global function snd_task_name(), and move it local
to snd-pcm-oss module.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: Documentation,PCM Midlevel,Timer Midlevel,ALSA Core
Use the standard getnstimeofday() function instead of ALSA's own one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove snd_runtime_check() macro.
This macro worsens the readability of codes. They should be either
normal if() or removable asserts.
Also, the assert displays stack-dump, instead of only the last caller
pointer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch cleans last ac97 audio/modem codec interception in
initialization procedures (ac97_mixer_new()) and removes obsolete
SHARED_TYPE 'locking' which prevents from AMC codecs to function
correctly.
Signed-off-by: Sasha Khapyorsky <sashak@smlink.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds the magic IO wakeup code for the CardBus version of the
Creative Labs Audigy 2 to the snd-emu10k1 driver.
Without the magic IO enable sequence, reading from the IO region of the
card will fail spectacularly, and the machine will hang.
My next task will be getting the driver to actually play sound without
distortion.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
[ This is a work-in-progress, but since it avoids a total lockup
if the emu10k module is loaded on a machine with the cardbus
card inserted, we're better off with it than without it, even
if sound quality is bad right now ]
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
- added typedef unsigned int __nocast gfp_t;
- replaced __nocast uses for gfp flags with gfp_t - it gives exactly
the same warnings as far as sparse is concerned, doesn't change
generated code (from gcc point of view we replaced unsigned int with
typedef) and documents what's going on far better.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
EMU10K1/EMU10K2 driver
Fixed the error at loading SBLive Game board (and possible other models).
The PCI SSIDs of this board conflicts with SB Live 5.1 Platinum, which has
no AC97 chip.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AC97 Codec
Don't use dev.platform_data to store a reference to the containing
ac97_t structure. Such assignment is redundent since we can deduce the
ac97_t structure location from the contained device structure. This
sets platform_data free for other purposes.
Signed-off-by: Nicolas Pitre <nico@cam.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA Core
A new function snd_card_set_generic_dev() is introduced to add the
'generic device' support for devices without proper bus on sysfs.
It's a last resort, and should be removed in future when they have
a proper bus, instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes a macro definition so that kernel-doc can understand it.
Signed-off-by: Martin Waitz <tali@admingilde.org>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
- Removed kernel version dependency from tea575x-tuner.h
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
This patch introduces a memory-leak tracking version of kzalloc for ALSA.
Signed-off-by: Pekka Enberg <penberg@cs.helsinki.fi>
Cc: Jaroslav Kysela <perex@suse.cz>
Signed-off-by: Jiri Slaby <xslaby@fi.muni.cz>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
PCM Midlevel,ALSA<-OSS emulation,USB USX2Y
This patch removes open_flag from struct _snd_pcm_substream.
All of its uses are substituted by querying struct _snd_pcm_substream's
member ffile instead.
Signed-off-by: Karsten Wiese <annabellesgarden@yahoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
YMFPCI driver
Implements mixer controls for the volume of each playback substream of
the main PCM device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Documentation,AD1816A driver
Added clockfreq module option for the card with a different clock frequency
than 33kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AC97 Codec,PCI drivers
I've made the review changes and as requested I've pasted the RFC by
Nicolas below:-
'I would like to know what people think of the following patch. It
allows for a codec on an AC97 bus to be shared with other drivers which
are completely unrelated to audio. It registers a new bus type, and
whenever a codec instance is created then a device for it is also
registered with the driver model using that bus type. This allows, for
example, to use the extra features of the UCB1400 like the touchscreen
interface and the additional GPIOs and ADCs available on that chip for
battery monitoring. I have a working UCB1400 touchscreen driver here
that simply registers with the driver model happily working alongside
with audio features using this.'
Changes over RFC:-
o Now matches codec name within codec group.
o Added ac97_dev_release() to stop kernel complaining about no release
method for device.
o Added 'config SND_AC97_BUS' to sound/pci/Kconfig and moved 'config
SND_AC97_CODEC' out with the PCI=n statement.
o module is now called snd-ac97-bus
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Nicolas Pitre <nico@cam.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Documentation,CS46xx driver,EMU10K1/EMU10K2 driver,AD1848 driver
SB16/AWE driver,CMIPCI driver,ENS1370/1+ driver,RME32 driver
RME96 driver,ICE1712 driver,ICE1724 driver,KORG1212 driver
RME HDSP driver,RME9652 driver
This patch changes .iface to SNDRV_CTL_ELEM_IFACE_MIXER whre _PCM or
_HWDEP was used in controls that are not associated with a specific PCM
(sub)stream or hwdep device, and changes some controls that got
inconsitent .iface values due to copy+paste errors. Furthermore, it
makes sure that all control that do use _PCM or _HWDEP use the correct
number in the .device field.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
AC97 Codec
o Enhanced current WM97xx support to provide additional controls and
use the kcontrol suffix naming convention.
o Added AC97_HAS_NO_MIC, AC97_HAS_NO_TONE and AC97_HAS_NO_STD_PCM.
o Cleaned up WM97xx related comments.
o Removed some wm9713 double mono controls and replaced with stereo
controls.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fixed kconfig dependencies on ISA_DMA_API for parts of sound/* that rely
on it.
Signed-off-by: Al Viro <viro@parcelfarce.linux.theplanet.co.uk>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Charles Keepax