Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
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Merge tag 'asoc-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.14
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add helpers for obtaining the width of a format directly from params
since this is expected to become a common operation in ASoC.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
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Merge tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas into next/boards
From Simon Horman:
Renesas ARM based SoC board updates for v3.14
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
* tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas: (75 commits)
ARM: shmobile: lager: add gpio/fixed regulator for SDHI
ARM: shmobile: bockw: remove unused RSND_SSI_CLK_FROM_ADG
ARM: shmobile: armadillo: fixup FSI address size
ARM: Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
ARM: shmobile: mackerel: Use pinconf API to configure pin pull-down
ARM: shmobile: Lager:add SPI FLASH support on QSPI
ARM: shmobile: mackerel: clk_round_rate() can return a zero to indicate an error
ARM: shmobile: Add pinctrl_register_mappings() for Koelsch
ARM: shmobile: Use ->init_late() on Lager
ARM: shmobile: Hook up SW30-SW36 on Koelsch
ARM: shmobile: koelsch: mark GPIO keys as wake-up sources
ARM: shmobile: Use ->init_late() on Koelsch
ARM: shmobile: lager: mark GPIO keys as wake-up sources
ARM: shmobile: r7s72100 Genmai Multiplatform
ARM: shmobile: r7s72100 Genmai DT reference C bits
ARM: shmobile: r7s72100 Genmai DT reference DTS bits
ARM: shmobile: Initial r8a7791 and Koelsch multiplatform support
ARM: shmobile: koelsch: set .debounce_interval
ARM: shmobile: lager: set .debounce_interval
ARM: shmobile: bockw: add pin pull-up setting for SDHI
...
Signed-off-by: Olof Johansson <olof@lixom.net>
spear_pcm_request_chan() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:
a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
pointer rather than some custom type.
b) dma_data->filter_data rather than dma_data should be passed to
snd_dmaengine_pcm_request_channel() as the filter data.
Make minor changes to the SPEAr DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When running a 32bit kernel the hda_intel driver is still reporting
a 64bit dma_mask if the HW supports it.
From sound/pci/hda/hda_intel.c:
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
else {
pci_set_dma_mask(pci, DMA_BIT_MASK(32));
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32));
}
which means when there is a call to dma_alloc_coherent from
snd_malloc_dev_pages a machine address bigger than 32bit can be returned.
This can be true in particular if running the 32bit kernel as a pv dom0
under the Xen Hypervisor or PAE on bare metal.
The problem is that when calling setup_bdle to program the BLE the
dma_addr_t returned from the dma_alloc_coherent is wrongly truncated
from snd_sgbuf_get_addr if running a 32bit kernel:
static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= PAGE_MASK;
return addr + offset % PAGE_SIZE;
}
where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr.
Without this patch the HW will fetch the 32bit truncated address,
which is not the one obtained from dma_alloc_coherent and will result
to a non working audio but can corrupt host memory at a random location.
The current patch apply to v3.13-rc3-74-g6c843f5
Signed-off-by: Stefano Panella <stefano.panella@citrix.com>
Reviewed-by: Frediano Ziglio <frediano.ziglio@citrix.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
92eba04e4b
(ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG) removed
RSND_SSI_CLK_FROM_ADG, it is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Simon Horman <horms+renesas@verge.net.au>
Add fields to struct snd_dmaengine_pcm_config to allow custom:
- DMA channel names.
This is useful when the default "tx" and "rx" channel names don't
apply, for example if a HW module supports multiple channels, each
having different DMA channel names. This is the case with the FIFOs
in Tegra's AHUB. This new facility can replace
SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.
- DMA device
This allows requesting DMA channels for a device other than the device
which is registering the "PCM" driver. This is quite unusual, but is
currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
module contains its own FIFOs which DMA writes to. However, in Tegra30,
the DMA FIFOs were split out AHUB HW module, which then routes the data
through a cross-bar, and into the DAI HW modules. However, the current
ASoC driver structure does not expose this detail, and acts as if the
FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
driver is registered with the DAI HW module, yet the DMA channels must
be looked up in the AHUB HW module's device tree node. This new config
field allows that to happen. Eventually, the Tegra drivers will be
reworked to fully expose the AHUB, and this config field can be
removed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
MICA/B Single-Ended input selection depends on mica/b config so lets
make the mixer controls for them only show for selected mic's
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch reworks the MICA an MICB config for single-ended or
differential and the selection of which MIC for the single config
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB
explicitly with ifdef, and this causes build errors on some drivers
such as:
sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type
Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_MUX() doesn't currently initialize the .mask field. This
results in the mux never affecting HW, since no bits are ever set or
cleared. Fix SND_SOC_DAPM_MUX() to use SND_SOC_DAPM_INIT_REG_VAL() to
set up the reg, shift, on_val, and off_val fields like almost all other
SND_SOC_xxx() macros. It looks like this was a "typo" in the fixed
commit linked below.
This makes the speakers on the Toshiba AC100 (PAZ00) laptop work again.
Fixes: de9ba98b6d ("ASoC: dapm: Make widget power register settings more flexible")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: <stable@vger.kernel.org> # v3.12+
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.
In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.
Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.
Thus, this patch applys symmetry for channels and sample bits.
Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The drain and drain_notify callback were blocked by low level driver
until the draining was complete. Due to this being invoked with big
fat mutex held, others ops like reading timestamp, calling pause, drop
were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would
be required to be invoked by low level driver when drain or partial
drain has been completed by the DSP. Thus we make the drain and
partial_drain callback as non blocking and driver returns immediately
after notifying DSP. The waiting is done while releasing the lock so
that other ops can go ahead.
[ The commit 917f4b5cba was wrongly applied from the preliminary
patch. This commit corrects to the final version.
Sorry for inconvenience! -- tiwai ]
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
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Merge tag 'asoc-v3.13-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.13
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
The size of the register cache array is actually 6 instead of 7,
as it caches up to AK4114_REG_INT1_MASK. This resulted in unexpected
access out of array range, although most of them aren't so serious
(just reading one more byte on the stack at snd_ak4114_create()).
Also, the check of cache size was wrongly done by checking with
sizeof() instead of ARRAY_SIZE(). Fixed this together.
(And yes, hardcoded numbers are bad, but I keep the coding style as is
for making it clear what this patch actually does.)
Spotted by coverity among several CIDs, e.g. 711621.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds platform data support for a reset GPIO.
Also uses reset_gpio to toggle reset of the CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
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Merge tag 'asoc-v3.13' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.13
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
The drain and drain_notify callback were blocked by low level driver untill the
draining was complete. Due to this being invoked with big fat mutex held, others
ops like reading timestamp, calling pause, drop were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would be required
to be invoked by low level driver when drain or partial drain has been completed
by the DSP. Thus we make the drain and partial_drain callback as non blocking
and driver returns immediately after notifying DSP.
The waiting is done while relasing the lock so that other ops can go ahead.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that we can't use gen_pool_*() functions on archs
without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since
linux/genalloc.h doesn't provide dummy functions for all. We'd be
able to fix linux/genalloc.h size, but I take an easier path for
now...
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have more than just simple TX and RX DMA channels, for example
modern Samsung I2S IPs support a secondary transmit DMA stream which is
mixed into the primary stream during playback. Allow such devices to
specify the names of the channels to be requested in their dma_data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Now it's quite common that an SoC contains its on-chip internal RAM.
By using this RAM space for DMA buffer during audio playback/record,
we can shutdown the voltage for external RAM to save power.
So add new DEV type with iram malloc()/free() and accordingly modify
current default mmap() for the iram circumstance.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current SSI needs RSND_SSI_DEPENDENT flag to
decide dependent/independent mode.
And SCU needs RSND_SCU_USE_HPBIF flag
to decide HPBIF is enable/disable.
But these 2 means same things.
This patch adds new rsnd_scu_hpbif_is_enable()
function, and merges above methods.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow DMA data to be set at probe time for devices that can do that,
avoiding the need to do it every time we start a stream and supporting
non-DT dmaengine users using the helpers.
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently each platform making use the the generic dmaengine PCM driver still
needs to provide a custom snd_pcm_hardware struct which specifies the
capabilities of the DMA controller, e.g. the maximum period size that can be
supported. This patch adds code which uses the newly introduced
dma_get_slave_caps() API to query this information from the dmaengine driver.
The new code path will only be taken if the 'pcm_hardware' field of the
snd_dmaengine_pcm_config struct is NULL.
The patch also introduces a new 'fifo_size' field to the
snd_dmaengine_dai_dma_data struct which is used to initialize the
snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name()
callback on each component drivers.
But required behavior on almost all these drivers is
just returns its indexed driver's name.
This patch adds this feature as default behavior.
.of_xlate_dai_name() can overwrite it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for RST GPIO and Charge Pump Freq in platform data
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a comment to the trigger function in snd_soc_dai_ops struct about
possible command sequences.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rcar is using rsnd_is_gen1/gen2() to checking its
IP generation, but it needs data mask.
This patch fixes it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for virtual DAPM mixer controls. They are similar to
virtual DAPM enums. There is no hardware register backing the control, so
changing the control's value wont have any direct effect on the hardware. But it
still influences the DAPM graph by causing the path it sits on to be connected
or disconnected. This in turn can cause power changes for some of the widgets on
the DAPM graph, which will then modify the hardware state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC sound driver requires CPU/CODEC drivers for probing,
and each CPU/CODEC has some DAI on it.
Then, "dai name matching" have been used to identify
CPU-CODEC DAI pair on ASoC.
But, the "dai port number matching" is now required from DeviceTree.
The solution of this issue is to replace
the dai port number into dai name.
Now, CPU/CODEC are based on struct snd_soc_component,
and it can care above as common issue.
This patch adds .of_xlate_dai_name callback interface
on struct snd_soc_component_driver,
and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name.
Then, #sound-dai-cells which enables DAI specifier is required
on CPU/CODEC device tree properties.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Codec includes component by this patch,
and component moved to upside of codec
to avoid extra declaration.
Codec dai will be registered via component
by this patch.
Current component register function
is used for cpu, and it is using
dai/dais functions properly to keep
existing cpu dai name.
And now, it will be used from codec also.
But codec driver had been used dais function only
even though it was single dai.
This patch adds new flag which can selects
dai/dais function on component register
function to keep existing codec dai name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some codec drivers when running in slave mode require that BCLK to sample rate ratio
is explicitly set by the machine driver as it may not be exactly rate * frame size.
Extend the DAI API by adding :-
int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify error handling and remove repetitive (and rarely executed) code
for unregistration by providing a devm_snd_soc_register() card.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Since with the wider use of devres many drivers are now only calling
snd_soc_unregister_component() in their remove functions providing a
managed version will save a reasonable amount of code.
Signed-off-by: Mark Brown <broonie@linaro.org>
The only cache type left is the flat cache and new other cache types won't be
added since new drivers are supposed to use regmap directly for IO and caching.
This patch removes the snd_soc_cache_ops indirection that was added to support
multiple cache types and modifies the code to always use the flat cache
directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The reg_size field is calculated in snd_soc_register_codec() and then used
exactly once in snd_soc_flat_cache_init(). Since it is calculated based on other
fields from the codec struct just move the calculation to
snd_soc_flat_cache_init() and remove the 'reg_size' field from the codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
reg_def_copy was introduced in commit 3335ddca ("ASoC: soc-cache: Use
reg_def_copy instead of reg_cache_default") to keep a copy of the register
defaults around in case the register defaults where placed in the __devinitdata
section. With the __devinitdata section gone we effectivly keep the same data
around twice. This patch removes reg_def_copy and uses reg_cache_default
directly instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of snd_soc_bulk_write_raw() are left and new drivers are going to use
regmap directly for this, so the function can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of reg_access_defaults are left and new drivers are going to use regmap
for this, so support for it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Pull media updates from Mauro Carvalho Chehab:
"This series contains:
- Exynos s5p-mfc driver got support for VP8 encoder
- Some SoC drivers gained support for asynchronous registration
(needed for DT)
- The RC subsystem gained support for RC activity LED;
- New drivers added: a video decoder(adv7842), a video encoder
(adv7511), a new GSPCA driver (stk1135) and support for Renesas
R-Car (vsp1)
- the first SDR kernel driver: mirics msi3101. Due to some troubles
with the driver, and because the API is still under discussion, it
will be merged at staging for 3.12. Need to rework on it
- usual new boards additions, fixes, cleanups and driver
improvements"
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (242 commits)
[media] cx88: Fix regression: CX88_AUDIO_WM8775 can't be 0
[media] exynos4-is: Fix entity unregistration on error path
[media] exynos-gsc: Register v4l2 device
[media] exynos4-is: Fix fimc-lite bayer formats
[media] em28xx: fix assignment of the eeprom data
[media] hdpvr: fix iteration over uninitialized lists in hdpvr_probe()
[media] usbtv: Throw corrupted frames away
[media] usbtv: Fix deinterlacing
[media] v4l2: added missing mutex.h include to v4l2-ctrls.h
[media] DocBook: upgrade media_api DocBook version to 4.2
[media] ml86v7667: fix compile warning: 'ret' set but not used
[media] s5p-g2d: Fix registration failure
[media] media: coda: Fix DT driver data pointer for i.MX27
[media] s5p-mfc: Fix input/output format reporting
[media] v4l: vsp1: Fix mutex double lock at streamon time
[media] v4l: vsp1: Add support for RT clock
[media] v4l: vsp1: Initialize media device bus_info field
[media] davinci: vpif_capture: fix error return code in vpif_probe()
[media] davinci: vpif_display: fix error return code in vpif_probe()
[media] MAINTAINERS: add entries for adv7511 and adv7842
...
Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.
The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysfs_registered field was added to the snd_soc_codec struct in commit
f0fba2ad ("ASoC: multi-component - ASoC Multi-Component Support"), but has never
been used.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAPM context struct has its own field where it stores the pointer to the
DAPM debugfs entry. The debugfs_dapm field in the snd_soc_platform and
snd_soc_codec structs are completely unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The control_type field was used by the core to track which raw IO methods to
use, but when switching to regmap this was no longer necessary and so the last
user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map
API for ASoC generic physical I/O"). The field is now completely unused and can
be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_dapm_new_widgets() works on the ASoC card as a whole not on a specific
DAPM context. The DAPM context that is passed as the parameter is only used to
look up the pointer to the card. This patch updates the signature of
snd_soc_dapm_new_widgets() to take the card directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
This patch adds generic ac97 reset functions using pincontrol and gpio
parsed from devicetree.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not
defined. Add a wrapper macro around printk_limit() that is conditionally defined by
CONFIG_SND_DEBUG.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move include/sound/tea575x-tuner.h to include/media/tea575x.h and update files that include it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
Use snd_dmaengine_dai_dma_data for passing the dma parameters from
clients to the pxa pcm lib. This does no functional change, it's just an
intermedia step to migrate the pxa bits over to dmaengine.
The calculation of dcmd is a transition hack which will be removed again
in a later patch. It's just there to make the transition more readable.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The
only difference is that snd_soc_info_enum_double() is also able to handle stereo
controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext()
for the SOC_ENUM_EXT control's info callback allows us to remove
snd_soc_info_enum_ext().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info
callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The
snd_soc_info_volsw_ext() function has been unused ever since then, so remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds BUSIF support for R-Car sound DMAEngine transfer.
The sound data will be transferred via FIFO which can cover blank time
which will happen when DMA channel is switching.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DMAEngine transfer on SSI.
But, it transfers sound data from memory to SSI directly
without using HPBIF at this time.
It will be updated soon
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is using struct rsnd_dai_platform_info
so that indicate sound DAI information (playback/capture SSI ID).
But, SSI settings were also required separately.
Thus, platform settings was very un-understandable.
This patch adds dai_id to SSI
settings, and removed rsnd_dai_platform_info.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some devices have the problem that if a internal audio signal source is disabled
the output of the source becomes undefined or goes to a undesired state (E.g.
DAC output goes to ground instead of VMID). In this case it is necessary, in
order to avoid unwanted clicks and pops, to disable any mixer input the signal
feeds into or to active a mute control along the path to the output. Often it is
still desirable to expose the same mixer input control to userspace, so cerain
paths can sill be disabled manually. This means we can not use conventional DAPM
to manage the mixer input control. This patch implements a method for letting
DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable
the control if the path on which the control sits becomes inactive. Userspace
will then only see a cached copy of the controls state. Once DAPM powers the
path up again it will sync the userspace setting with the hardware and give
control back to userspace.
To implement this a new widget type is introduced. One widget of this type will
be created for each DAPM kcontrol which has the auto-disable feature enabled.
For each path that is controlled by the kcontrol the widget will be connected to
the source of that path. The new widget type behaves like a supply widget,
which means it will power up if one of its sinks are powered up and will only
power down if all of its sinks are powered down. In order to only have the mixer
input enabled when the source signal is valid the new widget type will be
disabled before all other widget types and only be enabled after all other
widget types.
E.g. consider the following simplified example. A DAC is connected to a mixer
and the mixer has a control to enable or disable the signal from the DAC.
+-------+
+-----+ | |
| DAC |-----[Ctrl]-| Mixer |
+-----+ : | |
| : +-------+
| :
+-------------+
| Ctrl widget |
+-------------+
If the control has the auto-disable feature enabled we'll create a widget for
the control. This widget is connected to the DAC as it is the source for the
mixer input. If the DAC powers up the control widget powers up and if the DAC
powers down the control widget is powered down. As long as the control widget
is powered down the hardware input control is kept disabled and if it is enabled
userspace can freely change the control's state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code is limited to only setting or clearing a single bit in a
register to power a widget up or down. This patch extends the DAPM code to be
more flexible in that regard and allow widgets to use arbitrary values to be
used to put a widget in either on or off state.
Since the snd_soc_dapm_widget struct already contains a on_val and off_val field
no additional fields need to be added and in fact the invert field can even be
removed. Also the generated code is slightly smaller.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we store for each path which control (if any at all) is associated
with that control. But we are only ever interested in the reverse relationship,
i.e. we want to know all the paths a certain control is associated with. This is
currently implemented by always iterating over all paths. This patch updates the
code to keep a list for each control which contains all the paths that are
associated with that control. This improves the run time of e.g.
soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n
being the number of paths for the card) to O(1).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The 'value' field is really per control and not per widget. Currently it is only
used for virtual MUXes, which only have one control per widget. So in that case
there is not so much of a difference between whether it is stored per widget or
per control. Moving the 'value' field from the widget to the control will allow
us to use it also for cases where we have more than one control per widget. E.g.
for mixers with multiple input controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places.
This patch puts them into a common helper function. Having this encapsulated in
a helper function will also make it more easier to eventually change the data
layout of the kcontrol's private data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The update field of a DAPM context is only assigned while the card's dapm_mutex
is locked, the field is also cleared again while the mutex is stil locked. So
there will only ever be one DAPM context at a time with a non-NULL update field.
So it is safe to move the update field from the DAPM context struct to the card
struct. Doing so will allow further cleanups in this area.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is useful for drivers who want to grab a pointer to
snd_kcontrol outside of the kcontrol callbacks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
As 1st protype, this patch adds SSI feature on this driver.
But, it is PIO sound playback support only at this point.
The DMA transfer, and capture feature will be supported in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds ADG feature which controls sound clock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds SCU feature on this driver.
But, it defines SCU style only, does nothing at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
The main difference between Gen1 and Gen2 are
1) register offset, 2) data path
In order to control Gen1/Gen2 by same method,
this patch adds gen.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuits are different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2).
(Actually, there are many difference in Generation1 chips)
Basically, for the future, Renesas R-Car series will use
Gen2 style sound circuit, but driver should care Gen1 also.
The main differences between Gen1 and Gen2 peripheral
are 1) register offset, 2) data path.
This patch adds basic (core) feature for R-Car
series sound driver as prototype
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move HW initialization to separate function to allow using the code without
the v4l parts. This is needed for use in the bttv driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
In order to avoid race conditions the assignment of dapm->update should happen
while card->dapm_mutex is being held. To allow CODEC drivers to run a register
update when using snd_soc_dapm_mux_update_power() or
snd_soc_dapm_mixer_update_power() add a update parameter to these two functions.
The update parameter will be assigned to dapm->update while card->dapm_mutex is
locked.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when updating a control that is shared between multiple widgets the
whole power-up/power-down sequence is being run once for each widget. The
control register is updated during the first run, which means the CODEC internal
routing is also updated for all widgets during this first run. The input and
output paths for each widgets are only updated though during the respective run
for that widget. This leads to a slight inconsistency between the CODEC's
internal state and ASoC's state, which causes non optimal behavior in regard to
click and pop avoidance.
E.g. consider the following setup where two MUXs share the same control.
+------+
A1 ------| |
| MUX1 |----- C1
B1 ------| |
+------+
|
control ---+
|
+------+
A2 ------| |
| MUX2 |----- C2
B2 ------| |
+------+
If the control is updated to switch the MUXs from input A to input B with the
current code the power-up/power-down sequence will look like this:
Run soc_dapm_mux_update_power for MUX1
Power-down A1
Update MUXing
Power-up B1
Run soc_dapm_mux_update_power for MUX2
Power-down A2
(Update MUXing)
Power-up B2
Note that the second 'Update Muxing' is a no-op, since the register was already
updated.
While the preferred order for avoiding pops and clicks should be:
Run soc_dapm_mux_update_power for control
Power-down A1
Power-down A2
Update MUXing
Power-up B1
Power-up B2
This patch changes the behavior to the later by running the updates for all
widgets that the control is attached to at the same time.
The new code is also a bit simpler since callers of
soc_dapm_{mux,muxer}_update_power don't have to loop over each widget anymore
and neither do we need to keep track for which of the kcontrol's widgets the
current update is.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
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Merge tag 'asoc-v3.11-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.11
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
Currently we can only have a single platform built in with AC'97 support
due to the use of a global variable to provide the bus operations. Fix
this by making that variable a pointer and having the bus drivers set the
operations prior to registering.
This is not a particularly good or nice approach but it avoids blocking
multiplatform and a real fix involves fixing the fairly deep problems
with AC'97 support - we should be converting it to a real bus.
Acked-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a new helper function, snd_ctl_sync_vmaster(), which updates
the slave put callbacks forcibly as well as calling the hook. This
will be used in the upcoming patch in HD-audio codec driver for
toggling the mute in vmaster slaves.
Along with the new function, the old snd_ctl_sync_vmaster_hook() is
replaced as a macro calling with the argument hook_only=true.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
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Merge tag 'asoc-v3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.11
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
* for-linus: (635 commits)
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
ALSA: hda - Fix pin configurations for MacBook Air 4,2
ALSA: usb-audio: work around Android accessory firmware bug
ALSA: hda - Headset mic support for three more machines
Linux 3.10-rc6
smp.h: Use local_irq_{save,restore}() in !SMP version of on_each_cpu().
powerpc: Fix missing/delayed calls to irq_work
powerpc: Fix emulation of illegal instructions on PowerNV platform
powerpc: Fix stack overflow crash in resume_kernel when ftracing
snd_pcm_link(): fix a leak...
use can_lookup() instead of direct checks of ->i_op->lookup
move exit_task_namespaces() outside of exit_notify()
fput: task_work_add() can fail if the caller has passed exit_task_work()
xfs: don't shutdown log recovery on validation errors
xfs: ensure btree root split sets blkno correctly
xfs: fix implicit padding in directory and attr CRC formats
xfs: don't emit v5 superblock warnings on write
mei: me: clear interrupts on the resume path
mei: nfc: fix nfc device freeing
mei: init: Flush scheduled work before resetting the device
...
Since commit 85762e71 ("ASoC: dapm: Implement mixer control sharing") the
long_name field of the snd_soc_dapm_path struct is unused. All of the name
handling now happens entirely in dapm_create_or_share_mixmux_kcontrol(). So we
can remove the long_name field from the snd_soc_dapm_path struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the ALC5640 codec driver.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.
Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.
Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently ALSA supports up to 32 card instances when the dynamic minor
is used. While 32 cards are usually big enough for normal use cases,
there are sometimes weird requirements with more card support.
Actually, this limitation, 32, comes from the index option, where you
can pass the bit mask to assign the card. Other than that, we can
actually give more cards up to the minor number limits (currently 256,
which can be extended more, too).
This patch adds a new Kconfig to specify the max card numbers, and
changes a few places to accept more than 32 cards.
The only incompatibility with high card numbers would be the handling
of index option. The index option can be still used to pass the
bitmask for card assignments, but this works only up to 32 slots.
More than 32, no bitmask style option is available but only a single
slot can be specified via index option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ops field of the snd_pcm_substream struct is never modified inside the ALSA
core. Making it const allows drivers to declare their snd_pcm_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices may benefit from being able to start some parts of the widget
power up/down sequence earlier on in the sequence than the point at which
the final power state is committed. Support these by providing events which
are called before any power state changes are done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
Userspace is not meant to have to handle all strange dB ranges,
so add a specification comment.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
Fix typo in printk and comments within various drivers.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some platforms which are half-duplex share the same DMA channel between the
playback and capture stream. Add support for this to the generic dmaengine PCM
driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
Use the generic dmaengine PCM driver instead of a custom implemention. There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platforms which don't use devicetree yet or have to optionally
support a non-devicetree way to request the DMA channel. The patch adds the
compat_request_channel and compat_filter_fn callbacks to the
snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it
will be used to request the DMA channel. If not dma_request_channel with
compat_filter_fn as the filter function will be used to request the channel.
The patch also exports the snd_dmaengine_pcm_request_chan() function, since
compat platforms will want to use it to request their DMA channel.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a generic dmaengine PCM driver. It builds on top of the
dmaengine PCM library and adds the missing pieces like DMA channel management,
buffer management and channel configuration. It will be able to replace the
majority of the existing platform specific dmaengine based PCM drivers.
Devicetree is used to map the DMA channels to the PCM device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and
snd_soc_unregister_platform with the difference that they won't allocate and
free the snd_soc_platform structure.
Also add snd_soc_lookup_platform which looks up a platform by the device it has
been registered for.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the dmaengine PCM library to allow the DMA channel to be requested
before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA
channel instead of a filter function and filter parameter as its parameters.
snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows
a dmaengine based PCM driver to request its channels before the substream is
opened.
The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan()
and snd_dmaengine_pcm_close_release_chan(), which have the same signature and
behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new
variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are
updated to use snd_dmaengine_pcm_open_request_chan() and
snd_dmaengine_pcm_close_release_chan().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
This patch adds a common DMA data struct which can be used by DAI drivers to
communicate their DMA configuration requirements to the DMA pcm driver. Having
a common data structure for this allows us to implement common functions on top
of them, which can be used by multiple platforms.
This patch also introduces a new function to initialize certain fields of a
dma_slave_config struct from the common DAI DMA data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's compr_ops structure. Making it
const allows ASoC platform drivers to declare their snd_compr_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's ops structure. Making it const
allows ASoC platform drivers to declare their snd_pcm_ops struct as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does no not modify the driver of a platform. Making it const
allows ASoC platform drivers to declare the snd_soc_platform_driver struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions were initially added to be able to support some oddball dma
drivers, but all users have been updated to deal with the situation without the
help of snd_dmaengine_pcm_{set,get}_data, so these two functions can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Source files shouldn't have the executable bit set.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds .name member on snd_soc_component_driver.
But this patch doesn't care about whether cmpnt_drv was NULL,
and/or its name was NULL in snd_soc_register_component()
at this point.
Because, it is easy to switch over to
snd_soc_register_component() from snd_soc_register_dais()
if it doesn't care cmpnt_drv was NULL.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.
As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.
This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG
is set leads to frequent bugs, since other similar macros in the kernel
have different behavior. Let's make snd_BUG_ON() act like those macros
so it will stop being accidentally misused.
Signed-off-by: Christine Spang <christine.spang@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a driver for TI's TA5086 6-channel PWM processor.
This chip has a very unusual register layout, specifically because the
registers are of unequal size, and multi-byte registers require bulk
writes to take effect. Regmap does not support these kind of mappings.
Currently, the driver does not touch any of the registers >= 0x20, so
it doesn't matter, because the register map is mapped to an 8-bit array.
In case more features will be added in the future that require access
to higher registers, the entire regmap H/W I/O routines have to be
open-coded.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 497098be ("ASoC: dapm: Remove bodges for no-widget CODECs") removed the
last user of the n_widgets field. Currently it is incremented for each widget
added, but the value is never used, so we can remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This field was added in commit 2e72f8e ("ASoC: New enum type: value_enum"), but
has never been used since. Considering that the soc_enum struct is usually
shared between all instances of a CODEC, it also doesn't make much sense to have
a pointer to DAPM specific data in it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tegra only supports, and always enables, device tree. Remove all runtime
checks for DT support from the driver.
This allows removal of the hard-coded Harmony ASoC mapping table, since
Harmony only boots with DT now.
All board-specific configuration now comes from device tree, so there is
no need to have a platform_data structure. Rework the driver to parse the
device tree directly into struct tegra_wm8903.
Also some slight re-ordering of probe() so that the code more closely
resembles other drivers for easier comparison. Inparticular, the GPIO DT
parsing and initial programming are moved together for each GPIO.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This branch contains changes for OMAP that came in late during the release
staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs (Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and revision
detection, as well as removal of some dead code (CONFIG_32K_TIMER_HZ)
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Merge tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC late OMAP changes from Olof Johansson:
"This branch contains changes for OMAP that came in late during the
release staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs
(Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and
revision detection, as well as removal of some dead code
(CONFIG_32K_TIMER_HZ)"
* tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (34 commits)
ARM: omap2: include linux/errno.h in hwmod_reset
ARM: OMAP2+: fix some omap_device_build() calls that aren't compiled by default
ARM: OMAP4: hwmod data: Enable AESS hwmod device
ARM: OMAP4: hwmod data: Update AESS data with memory bank area
ARM: OMAP4+: AESS: enable internal auto-gating during initial setup
ASoC: TI AESS: add autogating-enable function, callable from architecture code
ARM: OMAP2+: hwmod: add enable_preprogram hook
ARM: OMAP4: clock data: Add missing clkdm association for dpll_usb
ARM: OMAP2+: PM: Fix the dt return condition in pm_late_init()
ARM: OMAP2: am33xx-hwmod: Fix "register offset NULL check" bug
ARM: OMAP2+: AM33xx: hwmod: add missing HWMOD_NO_IDLEST flags
ARM: OMAP: AM33xx hwmod: Add parent-child relationship for PWM subsystem
ARM: OMAP: AM33xx hwmod: Corrects PWM subsystem HWMOD entries
ARM: DTS: AM33XX: Add nodes for OCMC RAM and WKUP-M3
ARM: OMAP2+: AM33XX: Update the hardreset API
ARM: OMAP2+: AM33XX: hwmod: Update the WKUP-M3 hwmod with reset status bit
ARM: OMAP2+: AM33XX: hwmod: Fixup cpgmac0 hwmod entry
ARM: OMAP2+: AM33XX: hwmod: Update TPTC0 hwmod with the right flags
ARM: OMAP2+: AM33XX: hwmod: Register OCMC RAM hwmod
ARM: OMAP2+: AM33XX: CM/PRM: Use __ASSEMBLER__ macros in header files
...
Currently if a path loops back on itself we correctly skip over it to
avoid going into an infinite loop but this causes us to ignore the need
to power up the path as we don't count the loop for the purposes of
counting inputs and outputs. This means that internal loopbacks within a
device that have powered devices on them won't be powered up.
Fix this by treating any path that is currently in the process of being
recursed as having a single input or output so that it is counted for
the purposes of power decisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
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Merge tag 'asoc-3.9-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.9
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
This patch adds support for the Dialog DA7213 audio codec.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
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Merge tag 'asoc-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.9
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
Add a basic header file for the TI AESS IP block, located in the OMAP4
Audio Back-End subsystem.
Currently, this header file only contains a function to enable the
AESS internal clock auto-gating. This will be used by a subsequent
patch to ensure that the AESS won't block the entire chip
low-power-idle mode. We wish to be able to place the AESS into idle
even when no AESS driver has been compiled in.
Signed-off-by: Paul Walmsley <paul@pwsan.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Péter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: Jerry Wong <jerry.wong@maximintegrated.com>
Tested-by: Matthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: Ralph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some audio drivers are calling snd_dma_continuous_data(GFP_KERNEL)
which makes "sparse" give a warning:
$ make C=2 M=sound/usb modules
...
sound/usb/6fire/pcm.c:625:25: warning: cast from restricted gfp_t
sound/usb/caiaq/audio.c:845:41: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:997:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:1001:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:774:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:778:54: warning: cast from restricted gfp_t
Add __force to the cast to silence the warning.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds snd_soc_of_parse_daifmt() and supports below style on DT.
[prefix]format = "i2c";
[prefix]clock-gating = "continuous";
[prefix]bitclock-inversion;
[prefix]bitclock-master;
[prefix]frame-master;
Each driver can use specific [prefix]
(ex simple-card,cpu,dai,format = xxx;)
This sample will be
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT |
SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core does not modify these fields, so they can be made const. This allows
drivers to declare their op tables as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current simple-card driver calls asoc_simple_card_dai_init()
if platform had a asoc_simple_card_dai_init pointer.
And, this initialization function works only
when platform has an applicable initial value for each dai settings.
And basically, almost all sound card requires certain initialization.
This means that almost all platform has initialization settings,
and driver do nothing if it doesn't have settings.
And additionally, current simple-card supports sysclk settings but it was
only for codec. In order to abolish deviation between cpu and codec,
and in order to simplify processing,
this patch adds asoc_simple_dai, and removed pointless
struct asoc_simple_dai_init_info which was trigger of
calling asoc_simple_card_dai_init().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All MXS users have been converted to device tree and the board files have been
removed.
No need to keep platform data in the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Empty files can get deleted by the patch program, so remove empty Kbuild
files and their links from the parent Kbuilds.
Signed-off-by: David Howells <dhowells@redhat.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
FSI driver's flag usage was changed/removed by
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
ab6f6d8521
(ASoC: fsi: add master clock control functions)
And unused flags had been removed on FSI driver,
but the definition had been kept to avoid compile error.
It is possible to cleanup sh_fsi.h now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ab6f6d8521
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.
One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.
This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
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Merge tag 'asoc-3.8p1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
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Merge tag 'asoc-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.8
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Yet again like previous two commits, drop the old hwdep user-space
firmware code from vx driver (snd-vxpocket and snd-vx222).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CS4271 has a feature to sync its analog mute flags, so one mute
circuitry can be used for both channels.
Give users access to this feature with a new DT property and a flag in
the platform data.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Pull media updates from Mauro Carvalho Chehab:
"The first part of the media updates for Kernel 3.7.
This series contain:
- A major tree renaming patch series: now, drivers are organized
internally by their used bus, instead of by V4L2 and/or DVB API,
providing a cleaner driver location for hybrid drivers that
implement both APIs, and allowing to cleanup the Kconfig items and
make them more intuitive for the end user;
- Media Kernel developers are typically very lazy with their duties
of keeping the MAINTAINERS entries for their drivers updated. As
now the tree is more organized, we're doing an effort to add/update
those entries for the drivers that aren't currently orphan;
- Several DVB USB drivers got moved to a new DVB USB v2 core; the new
core fixes several bugs (as the existing one that got bitroted).
Now, suspend/resume finally started to work fine (at least with
some devices - we should expect more work with regards to it);
- added multistream support for DVB-T2, and unified the API for
DVB-S2 and ISDB-S. Backward binary support is preserved;
- as usual, a few new drivers, some V4L2 core improvements and lots
of drivers improvements and fixes.
There are some points to notice on this series:
1) you should expect a trivial merge conflict on your tree, with the
removal of Documentation/feature-removal-schedule.txt: this series
would be adding two additional entries there. I opted to not
rebase it due to this recent change;
2) With regards to the PCTV 520e udev-related breakage, I opted to
fix it in a way that the patches can be backported to 3.5 even
without your firmware fix patch. This way, Greg doesn't need to
rush backporting your patch (as there are still the firmware cache
and firmware path customization issues to be addressed there).
I'll send later a patch (likely after the end of the merge window)
reverting the rest of the DRX-K async firmware request, fully
restoring its original behaviour to allow media drivers to
initialize everything serialized as before for 3.7 and upper.
3) I'm planning to work on this weekend to test the DMABUF patches
for V4L2. The patches are on my queue for several Kernel cycles,
but, up to now, there is/was no way to test the series locally.
I have some concerns about this particular changeset with regards
to security issues, and with regards to the replacement of the old
VIDIOC_OVERLAY ioctl's that is broken on modern systems, due to
GPU drivers change. The Overlay API allows direct PCI2PCI
transfers from a media capture card into the GPU framebuffer, but
its API is crappy. Also, the only existing X11 driver that
implements it requires a XV extension that is not available
anymore on modern drivers. The DMABUF can do the same thing, but
with it is promising to be a properly-designed API. If I can
successfully test this series and be happy with it, I should be
asking you to pull them next week."
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (717 commits)
em28xx: regression fix: use DRX-K sync firmware requests on em28xx
drxk: allow loading firmware synchrousnously
em28xx: Make all em28xx extensions to be initialized asynchronously
[media] tda18271: properly report read errors in tda18271_get_id
[media] tda18271: delay IR & RF calibration until init() if delay_cal is set
[media] MAINTAINERS: add Michael Krufky as tda827x maintainer
[media] MAINTAINERS: add Michael Krufky as tda8290 maintainer
[media] MAINTAINERS: add Michael Krufky as cxusb maintainer
[media] MAINTAINERS: add Michael Krufky as lg2160 maintainer
[media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer
[media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer
[media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer
[media] MAINTAINERS: add Michael Krufky as tda18271 maintainer
[media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap
[media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend()
[media] exynos-gsc: Add missing static storage class specifiers
[media] exynos-gsc: Remove <linux/version.h> header file inclusion
[media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs()
[media] s5p-tv: Fix potential NULL pointer dereference error
[media] s5k6aa: Fix possible NULL pointer dereference
...
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Additional updates for v3.7
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
Convert #include "..." to #include <path/...> in kernel system headers.
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This patch adds support for Dialog semiconductor's DA9055 audio codec.
This has been tested on DA9055 EVB with Samsung SMDK6410 board.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <david.chen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow regulators managed via DAPM to make use of the bypass support that
has recently been added to the regulator API by setting a flag
SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will
be put into bypass mode before being disabled, allowing the regulator to
fall into bypass mode if it can't be disabled due to other users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing struct snd_dma_buffer pointer instead, so that they work no
matter whether real SG buffer is used or not.
This is a preliminary work for the HD-audio DSP loader code.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.
It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for tuning AM (on devices with the necessary additional
hardware components), and advertise the available bands using the new
VIDIOC_ENUM_FREQ_BANDS ioctl.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
For following the standard, define more channel map positions and
shuffle the items a bit:
- As both PulseAudio and gstreamer define MONO channel position
explicitly, we should follow that, too. The mono streams point to
this channel position unless they are explicitly assigned to certain
channel positions.
- Top-front-* and Top-rear-* positions are added, carried from
PulseAudio's definitions.
- Move NA and MONO definitions at the top of table right after
UNKNOWN, since these are more abstract in comparison with other
practical positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be used to enable additional control of the regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
There is already a set of channel position definitions in alsa-lib
mixer.h, and it'd be more practical to keep the same order for the
PCM channel map, too. The value is shifted with 1 to keep zero for
UNKNOWN.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC650 has a channel swap option between surround and CLFE channels,
so we need to tweak the channel maps dynamically depending on the
register bit.
Now struct snd_ac97 can contain chmap pointers for playback and
capture. The driver may store these and let ac97 driver changing the
channel mapping dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the basic data types for the standard channel
mapping API handling.
- The definitions of the channel positions and the new TLV types are
added in sound/asound.h and sound/tlv.h, so that they can be
referred from user-space.
- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
control elements representing the channel maps for each PCM
(sub)stream.
- Some standard pre-defined channel maps are provided for
convenience.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since bypass paths aren't part of DAPM streams and we may not have any
DAPM streams there may not be anything that triggers a DAPM sync for
them. Mark all input and output widgets as dirty and then sync to do so
at the end of suspend and resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove
current PGA control handling").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the Tegra+WM8903 ASoC platform data header out of
arch/arm/mach-tegra, as a pre-requisite of single zImage.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
