commit f0e9c080 - "ALSA: compress: change the way sample rates are sent to
kernel" changed the way sample rates are sent. So now we don't need to check for
PCM_RATE_xxx in kernel
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of comma leads to the wrong channel for an SPDIF channel.
Unfortunately this wasn't caught by compiler because it's still a
valid expression.
Reported-by: Alexander Aristov <aristov.alexander@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression pdev;
@@
pci_set_power_state(pdev,
- 3
+ PCI_D3hot
)
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This motherboard seems to have a flaky jack detection - when the
front HP is not present, the jack state quickly switches on and off.
This has been reported by three people in the bug, so I doubt it's
a user error this time.
BugLink: https://bugs.launchpad.net/bugs/1248116
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
ffs() returns the bit position from 1, while the ssm2158 driver code
assumes it being 0-based. Also, the bit mask computation of the two
channel slots are incorrect; it must have worked just casually.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Try to get the device's module clock if the dt has no clocks and
system-clock-frequency properties.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a quick fix for the below two issues when building spdif as modules.
1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2)
snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create
imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe
it again due to the execution platform_driver_unregister() in probe() when
meeting -EPROBE_DEFER at Step 2.
2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok",
'rmmod snd-soc-imx-spdif' would cause kernel dump with warning:
WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90()
sysfs: can not remove 'dapm_widget', no directory
This should be caused by disordered resourse releasing of the whole link.
And trying to unregister the card and then CODEC dev can't fix this issue.
Thus this patch just provides a simple fix to these two bugs by using the
snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that
there's no need to handle the registering and unregistering of CODEC or
CODEC dai any more.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
From "ASoC: make snd_soc_dai_link more symmetrical", can we see that
the name of CPU DAI maybe omitted. If the DAI name is omitted, try to
use the component name instead.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
ssi clock which is calculated from rsnd_ssi_master_clk_start()
should have flexibility since Renesas sound has
SRC (= Sampling Rate Converter).
But current implementation is using runtime->rate directly.
This patch tidyup rsnd_ssi_master_clk_start() parameter
as preparation of future SRC support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
rsnd_scu_set_hpbif() is renamed to rsnd_scu_rate_ctrl(),
since its naming doesn't indicate the function meaning.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
INT_ENABLE is needed only Gen2.
rsnd_mod_write() do nothing on Gen1, but it is confusable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
SRC_CTRL/BUSIF_MODE are used for transfer start.
This patch adds rsnd_scu_transfer_start() and merge these
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
rsnd_scu_set_route() is needed only Gen1.
This patch renames it to rsnd_scu_set_route_if_gen1()
and it adds comment to rsnd_reg member
in order to clarify it is used for Gen1.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This driver is assuming that
RBGA is used as source clock of 44.1kHz category, and
RBGB is used as source clock of 48kHz category.
This patch clarifies the variable name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use correct register name which appears in the datasheet
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we can check rsnd_ssi_init(), not, rsnd_ssi_start()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds rsnd_adg_set_ssi_clk() to access to
AUDIO_CLK_SEL0/1/2, and removes last user of
rsnd_write/read/bset which is very low level function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When call snd_soc_register_card, it will set driver data to this
device through dev_set_drvdata, then in driver, no need to call
platform_set_drvdata again, so remove it.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Revert the SAI's endianess for fifo data to/from DMA engine.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is maybe one bug or a limitation of the hardware that the {T,R}CR2's
Synchronous Mode bits must be set as late as possible, or the SAI device
maybe hanged up, and there has not any explaination about this limitation
in the SAI Data Sheet.
And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether
for Tx or Rx stream.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Enables/Disables the corresponding data channel for tx/rx operation.
A channel must be enabled before its FIFO is accessed, and then disable
it when tx/rx is stopped or idle.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
commit 5d229ce569 ("ASoC: samsung: move plat/ headers to local directory")
moved the header files but forgot to clean the pointers to their old locaton.
Remove them now.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Jingoo Han <jg1.han@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Because we cannot make sure which one of _dai_fmt() and _dai_sysclk()
will be firstly called. So move the RCSR/TCSR and TCR1/RCR1's
initialization to _dai_probe(), and this can make sure that before any
of {T,R}CR{1~5} register to be set the RCSR/TCSR's RE/TE bit has been
cleared for the hareware limitation.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Daniel Glöckner <daniel-gl@gmx.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Vaughan device support the 352800 rate and not
the 352000
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Generally we would write code for local variable like:
static new_func()
{
struct xxx *yyy;
...
int ret;
}
But this driver only follows this pattern for some functions, not all.
Thus this patch sorts the local variable in the general way.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since using dev_err() there's no need to mention SAI any more, it will
print the full name of the driver -- fsl_sai.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We can save this ret to make the code neater.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SAi only supports two data channels on hardware level and the driver also does
register the min->1 and max->2, so no need to check channels.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use common helper function snd_pcm_format_width() to make code neater.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are two functions haven't clk_disable_unprepare() if having error.
Thus fix them.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The name of cpu DAI maybe omitted, and then strlen() will lead
kernel panic.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no need of this function and makes the code slightly shorter
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Plantronics Gamecom 780 headset has a firmware problem, and when the
FU 0x09 volume is changed, it results in either too loud or silence
except for a very narrow range. This patch provides a workaround,
ignoring the node, initialize the volume in a sane value and keep
untouched.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In this case, there are two DACs, and DAC 0x03 is mono. In order
to make headphones and front speaker use DAC 0x02, and subwoofer use
DAC 0x03, we artificially cut the connection from nodes 0x14 and 0x15
to node 0x03, so they can only use DAC 0x02.
In addition, the 5460 and 5470 differs in the sense that 5470 also
needs a headset mic patch, whereas 5460 has individual detection for
headphone and headset mic.
BugLink: https://bugs.launchpad.net/bugs/1211920
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_ACPI dependent code should include <linux/acpi.h> instead of
directly including <acpi/acpi.h>. This patch cleans up such wrong
inclusions for Thinkpad ACPI users.
Signed-off-by: Lv Zheng <lv.zheng@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can rely on mfd driver to manage the register caching via regmap. The
driver still need to cache some registers associated with DL1/2 routes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The MFD core takes care of the restore via standard regmap API, no need to
do this anymore here.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a small register cache for registers which needs special caching
to reduce pop noise:
TWL6040_REG_HSLCTL, TWL6040_REG_HSRCTL, TWL6040_REG_EARCTL, TWL6040_REG_HFLCTL
and TWL6040_REG_HFRCTL.
Switch over and use the new small cache for these registers instead of the
main reg_cache.
This is in preparation to remove the local ASoC reg_cache from the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds three main functions for DAI master mode: set_dai_fmt(),
set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock
accordingly. After appending this patch, the fsl_ssi driver on i.MX series
has the ability to derive LRCLK and BCLK from baud clock source so as to
support some audio Codecs which can only be used in slave mode.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The kernel as a number of cases of gendered language. The majority of these
refer to objects that don't have gender in English, and so I've replaced
them with "it" and "its". Some refer to people (developers or users), and
I've replaced these with the singular "they" variant. Some are simply
typos that I've fixed up.
I've left cases where gendered language was used to refer to specific
individuals, was a quote or is part of license text.
Signed-off-by: Matthew Garrett <matthew.garrett@nebula.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When a Kconfig of a codec driver doesn't match with the controller
(CONFIG_SND_HDA_INTEL), it'll result in the non-working automatic
probing. Unfortunately kbuild can't give such a restriction, but at
least, it's possible to show a warning if such a condition is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, CONFIG_SND_HDA_CODEC_* kconfigs have been booleans due to
historical reasons. The major reason was that the automatic codec
driver probing wouldn't work if user sets a codec driver as a module
while the controller driver as a built-in. And, another reason was to
avoid exporting symbols of the helper codes when all drivers are built
in.
But, this sort of "kindness" rather confuses people in the end,
especially makes the config refinement via localmodconfig unhappy.
Also, a codec module would still work if you re-bind the controller
driver via sysfs (although it's no automatic loading), so there might
be a slight use case.
That said, better to let people fallen into a pitfall than being too
smart and restrict something. Let's make things straightforward: now
all CONFIG_SND_HDA_CODEC_* become tristate, and all symbols exported
unconditionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixes here are all driver specific ones, none of which particularly
stand out but all of which are useful to users of those drivers.
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Merge tag 'asoc-v3.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
The fixes here are all driver specific ones, none of which particularly
stand out but all of which are useful to users of those drivers.
Makes the code slightly shorter.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for configuring the sample rate on the SYSCLK side of the
ASRC.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently, the driver only supports configuration of the lower sample
rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this
patch adds support for configuring the higher sample rate (FSH).
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Certain use-cases require the DRE to be disabled so expose controls for
the enables.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
Modify the SPEAr PCM driver so that it's a utility library that can be
registered on each DAI, rather than a separate struct device. This is
more in line with how many recent DT-converted platforms operate, and
avoids the need for yet another struct device.
This is also required as a pre-cursor to removing
spear_pcm_request_chan().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ep93xx_compat_request_channel() 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 ep93xx 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>
Modify the ep93xx PCM driver so that it's a utility library that can be
registered on each DAI, rather than a separate struct device. This is
more in line with how many recent DT-converted platforms operate, and
avoids the need for yet another struct device.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Function sgtl5000_enable_regulators() is somehow odd in handling the
optional external VDDD supply. The driver can only enable this supply
on SGTL5000 chip before revision 0x11, and of course when this external
VDDD is present. It currently does something like below.
1. Check if regulator_bulk_get() on VDDA, VDDIO and VDDD will fail. If
it fails, VDDD must be absent and it falls on internal LDO by calling
sgtl5000_replace_vddd_with_ldo(). Otherwise, VDDD is used. And in
either case, regulator_bulk_enable() will be called to enable
3 supplies.
2. In case that SGTL5000 revision is later than 0x11, even if external
VDDD is present, it has to roll back the 'enable' and 'get' calls
with regulator_bulk_disable() and regulator_bulk_free(), and starts
over again by calling sgtl5000_replace_vddd_with_ldo() and
regulator_bulk_enable().
Such back and forth calls sequence is complicated and unnecessary.
Also, since commit 4ddfebd (regulator: core: Provide a dummy regulator
with full constraints), regulator_bulk_get() will always succeeds
because of the dummy regulator. Thus the VDDD detection is broken.
The patch changes the flow to something like the following, which should
be more reasonable and clear, and also fix the VDDD detection breakage.
1. Check if we're running a chip before revision 0x11, on which an
external VDDD can possibly be an option.
2. If it is an early revision, call regulator_get_optional() to detect
whether an external VDDD supply is available.
3. If external VDDD is present, call sgtl5000_replace_vddd_with_ldo() to
update sgtl5000->supplies info.
4. Drop regulator_bulk_get() call in sgtl5000_replace_vddd_with_ldo(),
and call it in sgtl5000_enable_regulators() no matter it's an
external VDDD or internal LDO.
5. Call regulator_bulk_enable() to enable these 3 regulators.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This adds Freescale SAI ASoC Audio support.
This implementation is only compatible with device tree definition.
Features:
o Supports playback/capture
o Supports 16/20/24 bit PCM
o Supports 8k - 96k sample rates
o Supports master and slave mode.
Signed-off-by: Alison Wang <b18965@freescale.com>
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The kernel as a number of cases of gendered language. The majority of these
refer to objects that don't have gender in English, and so I've replaced
them with "it" and "its". Some refer to people (developers or users), and
I've replaced these with the singular "they" variant. Some are simply
typos that I've fixed up.
I've left cases where gendered language was used to refer to specific
individuals, was a quote or is part of license text.
Signed-off-by: Matthew Garrett <matthew.garrett@nebula.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the Dell machines with codec whose Subsystem Id is 0x10280640,
no external microphone can be detected when plugging a 3-ring headset.
Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem.
The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269
family.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When wm8904 work in DSP mode B, we still need to configure it to
work in DSP mode. Or else, it will work in Right Justified mode.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Some devices are getting very close to the limit whilst polling the RAM
start, this patch adds a small delay to this loop to give a longer
startup timeout.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
When the process is sleeping at the SNDRV_PCM_STATE_PAUSED
state from the wait_for_avail function, the sleep process will be woken by
timeout(10 seconds). Even if the sleep process wake up by timeout, by this
patch, the process will continue with sleep and wait for the other state.
Signed-off-by: JongHo Kim <furmuwon@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the rates declared in the CPU DAI parameters:
- SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should
not be used with SNDRV_PCM_RATE_CONTINUOUS,
- SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max,
- the device may do streaming down to 5512Hz.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Matching works completely based on the cpu of_node.
Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reported-by: Kyung-Kwee Ryu <kyung-kwee.ryu@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
use snd_dmaengine_pcm_prepare_slave_config to set slave config,
and remove the max_burst_size = 4 hard code.
select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Where possible write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Because the "ASoC: dmaengine-pcm: Provide default config" has provided
us one defualt config of DMA. When using this, the config parameter of
devm_snd_dmaengine_pcm_register() will be NULL, so here we need to have
a check before using it.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:
sound/soc/adi/axi-spdif.c:194:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 194
Generated by: coccinelle/api/devm_ioremap_resource.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:
sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195
Generated by: coccinelle/api/devm_ioremap_resource.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe()
reads it out from register, so that we can use it in
sgtl5000_enable_regulators() with no need to read register again.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM can dynamically alter the FE to BE PCM links at runtime based
on mixer/mux setting updates. Add soc_dpcm_runtime_update() calling in
get/put function for mixer/mux to support this feature.
Signed-off-by: Nenghua Cao <nhcao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the missing clk_disable_unprepare() before return from
tegra20_ac97_platform_probe() in the error handling case.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
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Merge tag 'asoc-v3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
On the Dell machines with codec whose Subsystem Id is 0x10280610,
0x10280629 or 0x1028063e, no external microphone can be detected when
plugging a 3-ring headset. If we add "model=dell-headset-multi" for
the snd-hda-intel.ko, the problem will disappear.
The codecs on these machines belong to alc_269 family.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While enabling these machines, we found we would sometimes lose an
interrupt if we change hardware volume during playback, and that
disabling msi fixed this issue. (Losing the interrupt caused underruns
and crackling audio, as the one second timeout is usually bigger than
the period size.)
The machines were all machines from HP, running AMD Hudson controller,
and Realtek ALC282 codec.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1260225
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since all Exynos platforms have been converted to dmaengine and many of
the older platforms are in the process of conversion they do not need to
use the legacy s3c-dma APIs for DMA but can instead use the standard ASoC
dmaengine helpers. This both allows them to benefit from improvements
implemented in the generic code and supports multiplatform.
This patch includes some fixes from Padma for Exynos SoCs, her testing
was on a slightly earlier version of the patch due to unrelated breakage
preventing testing.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested By: Padmavathi Venna <padma.v@samsung.com>
In preparation for using the dmaengine helpers in ASoC rather than the
dmaengine wrappers for the Samsung API wrap the configuration of dma_data.
The dmaengine code expects different data to that used by the legacy API.
Signed-off-by: Mark Brown <broonie@linaro.org>
Check the return value of dma_request_slave_channel_reason() to see if
deferred probe happens, not the variable the return value will be
assigned to later.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 5eda87b890 ("ASoC: dmaengine: support deferred probe for DMA channels")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Enhance dmaengine_pcm_request_chan_of() to support deferred probe for
DMA channels, by using the new dma_request_slave_channel_or_err() API.
This prevents snd_dmaengine_pcm_register() from succeeding without
acquiring DMA channels due to the relevant DMA controller not yet being
registered.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove original filter from fsi_dma_probe(),
and use SH-DMA suitable filter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.
However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.
Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.
Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.
Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error; all other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error. All other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Optional DT property to specify the desired parent clock for the McASP fck
clock.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
An earlier patch overlooked this when the compatible has been changed from
omap2 -> am33x.
Rename omap2_mcasp_pdata to am33xx_mcasp_pdata.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of passing __iomem address (mcasp->base + register_offset) pass
the main mcasp structure and only access the mcasp->base in the low level
IO functions.
In most cases this helps with code readability and it will make it easier
to switch over to regmap in the future.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The IP in DRA7xx is similar to the IP found in TI81xxAM3xxx/AM4xxx type of
SoCs but it is is integrated with sDMA instead of eDMA. The suitable pcm
driver for DRA7xx is the omap-pcm driver which is using dmaengine.
In the driver we can configure both dma related structures used for eDMA and
sDMA. The only thing we need to make sure that we set the correct dma_data
at startup with snd_soc_dai_set_dma_data()
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In synchronous mode both transmit and receive sections are using the TX
clocks. In setup like this the TX clocks need to be enabled when capture
is running.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The audio data to/from McASP can be sent/received via two method:
Via the data port (preferred) or via the configuration bus.
Currently the driver assumes that all data communication will be done via
the data port.
This patch adds support for selecting the configuration port as data
interface.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The FIFO registers base address is different in dm646x compared to newer
SoCs with McASP IP. Instead of using two paths (switch/case) to handle the
difference we can simply pick the correct base address beforehand and use
offsets to address the register we need to configure.
With this change the indentation depth can be reduced as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace mcasp->base use with plain base in the davinci_mcasp_set_dai_fmt()
function since it has been already used by the remaining part of the function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Rename the private struct from davinci_audio_dev to davinci_mcasp.
Change the local use of the pointer to this struct from *dev to *mcasp.
The aim is to have better readable code for the first look since having
dev->xxxx in the code when using the local private struct is a bit
surprising.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It brings no benefit to inline this function due to it's size and function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since it is a private struct strictly used by the davinci-mcasp driver it
can be moved from header file to the source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It is better for readability to have the register definitions out from the
source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
These are not used, probably leftovers from the past.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Specify the dai formats to use within the snd_soc_dai_link structures. In
this way we can remove the code dealing with the dai format configuration
from the machin driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
AM43xx have the same McASP IP as AM33xx and both platform uses eDMA. Modify
the Kconfig so it will be possible to add audio support for AM43xx based
boards later.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have several boards using the same machine driver for audio support.
All of these machines can select a generic machine driver config option to
build the needed driver while keeping the config options used within the
driver for compile time code path selection.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The help text is misleading and the prompt itself explains the purpose of
this config section.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Gen2 has 0 - 9, total 10 channels, not 9 channels.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails,
all objects allocated during registration are leaked. Fix this by adding
error-handling code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Restructure the internals of dmaengine_pcm_request_chan_of() as a loop
over all channels to be allocated. This makes it easier to add logic
that applies to all allocated channels, without having to duplicate that
logic in each of the half-duplex/full-duplex paths.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite
registers for class-d amplifier.
classd amplifier controls are only used for AIC3X_MODEL_3007.
Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route
and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007
codecs.
Testet only with AIC3X_MODEL_3007
Signed-off-by: Jan Weitzel <j.weitzel@phytec.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC driver for the AXI-SPDIF softcore. The core implements a
simple SPDIF transmitter and is used on some Analog Devices' reference designs
for various FPGA platforms. For now the driver only support the PL330 as the the
DMA controller.
The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the AXI-I2S softcore. The core implements a simple
bidirectional I2S transceiver and is used by Analog Devices in some of their
reference designs for various FPGA platforms.
The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If we update it here, the set_bias_level() of Codec driver won't be normally
called and we will then miss some essential procedures in set_bias_level() of
the Codec driver. Thus drop it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case,
"val" is never assigned to, but left uninitialized. The other case does
initialized it. Fix this by initializing val at the start of the
function, and only ever ORing into it.
Update the handling of "mask" so it works the same way for consistency.
Update tegra20_spdif.c to use the same code-style for consistency, even
though it doesn't happen to suffer from the same problem at present.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Reviewed-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes: 0f163546a7 ("ASoC: tegra: use regmap more directly")
Cc: <stable@vger.kernel.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>
AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression. In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint. This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes the hardware reports LPIB being advanced than POSBUF.
When this happens, the driver adjusts to a positive value by adding
the buffer size. Then the driver detects it as an error (greater than
the period size), and stops the LPIB delay account from this point
on.
When I took a close look at these conditions, the values shown are all
very small numbers, and it'd be better to just ignore these values
instead of discontinuing the LPIB delay correction.
In this patch, the driver checks a negative delay value and ignores if
it's a significantly small error. Currently the threshold is set to
64 frames, but could be smaller.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback mixing paths aren't initialized correctly at init
callback. Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations. The parser shouldn't escape in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch skips the default depop delay before D3 for Haswell (10 ms) and
Valleyview2 (100 ms) display codec, to reduce codec suspend time.
The analog part of display audio is implemented in the external display. Some
displays have weak pop noise while others not when suspending, no matter there
is the default delay or not.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the
current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK,
which would cause the calculation result from DSPCLK_DIV invalid since bit
DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK
while the driver won't calculate it again for the current instance. In this
circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted
due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value.
So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for
calculation and then disables it afterward.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Initially, this binding and driver only describe/support playback to
headphones and speakers, and capture from the external microphone, with
GPIO-based jack detection for the headphone jack only.
This driver is useful for the Venice2 board.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
clk_prepare_enable() may fail, so let's check its return value and propagate it
in the case of error.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will allow a marginal speed improvement when used with a bus that
supports async I/O by reducing the amount of context thrashing between
writes, allowing the bus to be more fully utilised.
Signed-off-by: Mark Brown <broonie@linaro.org>
MacBook Air 2,1 has a fairly different pin assignment from its brother
MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19,
similarly like what iMac 9,1 requires, in order to make the sound
working on it.
Reported-and-tested-by: Bruno Prémont <bonbons@linux-vserver.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change sam9x5 with wm8731 work in DSP A mode, this will fix the
left/right channel swap issue.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the SSC specifiation, it should be enabled after DMA is
enabled. So, add trigger operation to make sure the right sequence.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
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Merge tag 'asoc-v3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.
And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that there is a dmaengine driver for the jz4740 DMA core we can use the
generic dmaengine PCM driver. This allows us to remove the custom jz4740-pcm
code completely.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Even if the CONFIG_PM explicity is undefined, let's convert to the
modern PM ops.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Let the core take care of applying sample rate and sample bits constraints
instead of open-coding this in the driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we introduced symmetric_channels and symmetric_samplebits, we implement
these two features to fsl_ssi so as to drop some no-more-needed code and make
the driver neat and clean.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'.
And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce
codec suspend time.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.
Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.
It still takes a second or two before the headset mic actually starts
working, but still better than nothing.
Information update from Kailang:
The verb was ADC digital mute(bit 6 default 1).
Switch internal mic and headset mic will run alc_headset_mode_default.
The coef index 0x11 will set to 0x0041.
Because headset mode was fixed type. It doesn't need to run
alc_determine_headset_type.
So, the value still keep 0x0041. ADC was muted.
BugLink: https://bugs.launchpad.net/bugs/1256840
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The normal mode of SSI allows it to send/receive data to/from the first
slot of each period. So we can use this normal mode to trick I2S signal
by puting/getting data to/from the first slot only (the left channel)
so as to support monaural audio playback and recording.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused. Since each output has own amp,
let's use it instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too. Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio devices tend to take long time for finishing the whole
probing procedure. In this patch, the time-consuming part of the
probing procedure, the codec probe and the rest initializations, are
moved in the work, so that they can be done asynchronously in parallel
with probes of other devices.
Since we already have this mechanism in the driver code for the
firmware and i915 request_symbol() stuff, we just need to enable it
always; the resultant patch even reduces more lines, which is an
additional bonus.
Credit goes to David Henningsson, who suggested this workaround.
Reported-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static checker found a possible array overflow in atmel/abdac.c:
static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
error: buffer overflow 'dac->rates' 6 <= 6"
This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vinod Koul