Add missing pm to current machine drivers so that all of them would
correctly do suspend/resume.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When using the legacy filter function channel requests we currently pass
the audio specific struct snd_dmaengine_dai_dma_data which isn't likely to
be helpful for actual filtering. Since there's already a field in the
structure called filter_data clearly intended for use here convert the
driver to use that.
All existing users of plain filter functions have been converted to use
an explicit compat function to override this behaviour except i.MX which
is working around this issue in its filter function and is updated to
just use filter_data directly here.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
When HP laptops with mute and mic-record LEDs go to runtime suspend,
these LEDs are turned on forcibly no matter whether GPIO pis are on or
off. This strange behavior seems triggered by resetting the HD-audio
bus link at azx_rutime_suspend(). So, just add a new hda_bus flag to
avoid the link reset at runtime suspend and set it for these HP
machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's just another variant of ALC269 & co.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Emitting an OOM message isn't necessary after input_allocate_device
as there's a generic OOM and a dump_stack already done.
Signed-off-by: Joe Perches <joe@perches.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>
When a machine goes to S3/S4 after power-save is enabled, the runtime
PM refcount might be incorrectly decreased because the power-down
triggered soon after resume assumes that the controller was already
powered up, and issues the pm_notify down.
This patch fixes the incorrect pm_notify call simply by checking the
current value properly.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.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>
Add device tree support to tpa6130a2 driver and document the
bindings.
Signed-off-by: Sebastian Reichel <sre@debian.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Device tree support for Davinci Machine driver
When the board boots with device tree, the driver will receive card,
codec, dai interface details (like the card name, DAPM routing map,
phandle for the audio components described in the dts file, codec mclk
speed). The card will be set up based on this information. Since the
routing is provided via DT we can mark the card fully routed so core
can take care of disconnecting the unused pins.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Darren Etheridge <detheridge@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
AM33xx uses same McASP IP as the Davinci Platform. This patch updates
Kconfig and makefile to enable build for McASP, PCM & Codec drivers.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Darren Etheridge <detheridge@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Setting a field in a static struct to NULL has no effect so don't bother
(and don't generate false positives for grep).
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Currently the ep93xx DMA code is one of the few users relying on the fact
that the compat code uses the dma_data as the filter data for non-DT
channel requests. Since the rest of the core expects this to be a struct
snd_dmaengine_dai_data this isn't terribly helpful this will be changed to
use the already existing filter data so avoid breaking ep93xx by open
coding the current behaviour.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
The serial-dir array gives this information so there is no need to have the
num-serializer property in DT description.
Just ignore the property in the driver the DTS files can be updated
separately without regression.
Update the documentation at the same time for davinci-mcasp
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Change the model omap2-mcasp-audio in compatible property to
am33xx-mcasp-audio as omap2 does not have mcasp.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Extract DMA channels directly from DT as they can not be found from
platform resources anymore. This is a work-around until davinci audio
driver is updated to use dmaengine.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a separate register location for data port registers to
mcasp DT bindings. On am33xx SoCs the McASP registers are mapped
trough L4 interconnect, but data port registers are also mapped trough
L3 bus to a different memory location.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Darren Etheridge <detheridge@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds S/PDIF input/output for mvebu DT boards.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
While reporting the jack status snd_soc_jack_report() invokes snd_soc_dapm_sync()
always. This should be required when we have pins associated with jack and
reporting enables or disables these.
So add a check for this case
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
... which was introduced by the previous commit a4e9a38b, causing
build errors without CONFIG_PROC_FS.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds codec ID (0x80862882) and module alias for
Valleyview2 display codec.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion of the si476x to regmap removed locking of the core during
register updates, allowing things like power state changes for the MFD to
happen during a register update. Avoid this by taking the core lock in the
DAI operations (which are the only things that do register updates) as we
used to do in the open coded register I/O functions.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Andrey Smirnov <andrew.smirnov@gmail.com>
At the moment, this driver supports only playback, while FFADO supports
(only) full-duplex devices. So, prevent conflicts by not claiming
devices that would be better handled by FFADO.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
While most DICE devices keep TCAT's default category ID of 0x04, Weiss
devices identify themselves with 0x00.
Reported-by: Rolf Anderegg <rolf.anderegg@weiss.ch>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Doing accesses without quadlet alignment is a bad idea because the
firmware's byte-swapping would garble the data; clarify this in the
documentation.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Instead of relying of some control panel application to configure some
fixed sample rate, allow applications to set it automatically.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
In preparation for sample rate selection support, read the stream
parameters that might change when running at different sample rates.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
In preparation for sample rate selection support, ensure that the driver
knows about the device's clock capabilities.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Avoid a lock inversion between dice->mutex and pcm->open_mutex.
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Since commit f2b3614cef (Don't check DMA time-out too shortly), we
need no longer to restrict the period length to less than 10 s.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When a bus reset happens, the enable register is automatically cleared,
so we do not need to clear it manually when stopping the stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add a flag to snd_fw_transaction() to allow it to abort when a bus reset
happens. This removes most of the duplicated error handling loops that
were required around calls to the low-level fw_run_transaction().
Also add a flag to suppress error messages; errors are expected when we
attempt to clean up after the device was unplugged.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the AMDTP streaming code to handle the non-standard stream format
that DICE devices use at sample rates greater than 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
DICE devices do not have a unique specifier ID in their unit directory
(it's always the same as the device vendor's ID), so rely on just the
version ID for driver loading, and use a heuristic in the probe callback
to detect actual DICE devices.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Introduce the helper function amdtp_out_stream_running(). This makes
many checks in amdtp.c clearer and frees the device drivers from having
to track this with a separate variable.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When aborting a PCM stream, the xrun is signaled only if the stream is
running. When disconnecting a PCM stream, calling snd_card_disconnect()
too early would change the stream into a non-running state and thus
prevent the xrun from being noticed by user space.
To prevent this, move the snd_card_disconnect() call after the xrun.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Allow AMDTP output streams to use blocking mode.
Use it for DICE devices, because the old DICE-II chip will in some cases
not be able to lock to non-blocking streams (erratum E7).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Instead of forcing a constant 44.1 kHz, read the current sample rate
from the device when opening the PCM device.
Actually changing the sample rate requires some separate controller
application.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
After a bus reset, do not stop the stream completely to avoid having to
reconfigure the device when restarting the stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
This patch adds support for device tree for the CS42L73 CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When there is an external clock, always use this one.
This prevents the two Dove audio devices to use the same DCO clock
at different rates.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound has clock pin for each SSI, and sometimes,
these pins are shared with paired SSI.
It may sometimes become "SSI-A clock pin is master" and
"SSI-B clock pin is slave", but "SSI-A/B clock pins are shared".
SSI-B needs SSI-A clock in this case.
Current R-Car sound driver is using RSND_SSI_xxx flag
to control this kind of shared pin behavior.
But, this information, especially clock master setting,
can be got from ASoC set_fmt settings.
This patch removes rsnd_ssi_mode_init() and extend rsnd_ssi_mode_set()
to controlling pin settings via .set_fmt.
This patch doesn't removes RSND_SSI_CLK_FROM_ADG flag at this point
to avoid conflict branch merging between ASoC <-> SH-ARM.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
Current simple-card returns error if DAI doesn't
support .set_fmt callback.
But the error is -ENOTSUPP (= not supported),
and it is not error.
This patch avoids such case
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A number of small, fairly unexciting, fixes for drivers - nothing stand
out.
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Merge tag 'asoc-v3.12-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.12
A number of small, fairly unexciting, fixes for drivers - nothing stand
out.
This patch adds some default settings for the generic dmaengine PCM driver for
the case that no config has been supplied. The following defaults are used:
* Use snd_dmaengine_pcm_prepare_slave_config for preparing the DMA slave
config.
* 512kB for the prealloc buffer size. This value has been chosen based on
'feels about right' and is not backed up by any scientific facts. We
may need to come up with something smarter in the future but it should
work fine for now.
With this infrastructure in place we can finally write DAI drivers which are
independent of the DMA controller they are connected to. This is e.g. useful if
the DAI IP core is reused across different SoCs, but the SoCs uses different DMA
controllers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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>
There is no need for the CODEC to go to standby on resume since the core will
power it up as needed and in any case it is an idle_bias_off CODEC so would
normally sit with bias off while idle.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Stephen Warren <swarren@nvidia.com>
If we have control over the LDO then disable it during suspend; the device
is already being put into reset so will be non-functional over suspend
anyway and this will save a small amount of power.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Stephen Warren <swarren@nvidia.com>
The core support for ep93xx (currently only the DMA driver) does not
depend on the architecture at all and everything else has more strict
dependencies so enable compile test builds for improved build coverage.
Signed-off-by: Mark Brown <broonie@linaro.org>
This is more idiomatic and is required for robust operation since we must
ensure that the clocking configuration is valid as rapidly as possible.
Signed-off-by: Mark Brown <broonie@linaro.org>
Partially restructures _snd_emu10k1_audigy_init_efx() and
_snd_emu10k1_init_efx() functions.
Be noted that the cast is demanded to use '__user'. So, in these cases,
avoid patches based on the coccinelle 'drop_kmalloc_cast' semantic patch.
Signed-off-by: Geyslan G. Bem <geyslan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cleanup to namespace the defines for the cs42l73 driver
Signed-off-by: Brian Austin <brian.austin@cirrus.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>
After convertion to snd_soc_register_card, platform driver should
reference snd_soc_pm_ops callbacks to properly suspend/resume sound
hardware. This was missed during conversion of PXA sound devices.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
After recent changes to codec/DAI initialization order changes, codec
driver (wm9712 in my case) tries to access codec prior to
pxa2xx_ac97_hw_probe() being called (because DAIs are probed after all
codecs are probed). Move hw-related probe/remove/suspend/resume
functions to pxa2xx-ac97 driver level, instead of DAI level.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
PXA25x also shows some problems when using interrupts during reset
handling. Thus do not use interrupts on all pxa kinds (to detect codec
ready state). Instead use a common mdelay-loop on all platforms to
detect codecs becoming ready.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As a start point for further development, this is an incomplete driver
for DICE devices:
- only playback (so no clock source except the bus clock)
- only 44.1 kHz
- no MIDI
- recovery after bus reset is slow
- hwdep device is created, but not actually implemented
Contains compilation fixes by Stefan Richter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Since the lock is used primarily in patch_hdmi.c, it's better to move
it in the local struct instead of exporting in hda_eld. The only
functions requiring the lock in hda_eld.c are proc accessors. So in
this patch, the proc entry and its creation/deletion/accessors are
moved into patch_hdmi.c, together with the mutex lock to pin_spec
struct.
The former proc info functions are exported so that they can be called
from patch_hdmi.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some per_pin fields and ELD contents might be changed dynamically in
multiple ways where the concurrent accesses are still opened in the
current code. This patch fixes such possible races by using eld->lock
in appropriate places.
Reported-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The zero-length connection list happens so often on Haswell HDMI, and
it results in warning messages like
ALSA: hda_codec: invalid CONNECT_LIST verb 5[1]:0
at each time the codec resumes from the power-save, which is fairly
annoying.
Since this is no real error, make it shown only in the verbose debug
mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fixup which should be called before codec being freed will come
to use in the next patch.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
'of_match_ptr' is defined in linux/of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
In GCC the sizeof(hdsp_version) is 8 because there is a 2 byte hole at
the end of the struct after ->firmware_rev.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pcm_usb_stream plugin requires the mremap explicitly for the read
buffer, as it expands itself once after reading the required size.
But the commit [314e51b9: mm: kill vma flag VM_RESERVED and
mm->reserved_vm counter] converted blindly to a combination of
VM_DONTEXPAND | VM_DONTDUMP like other normal drivers, and this
resulted in the failure of mremap().
For fixing this regression, we need to remove VM_DONTEXPAND for the
read-buffer mmap.
Reported-and-tested-by: James Miller <jamesstewartmiller@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Trigger commands may be passed multiple times. To avoid errors with
clk_enable/disable, store the saif state and return if saif is already
running/stopped.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the datasheet, the max_register is register 23.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the datasheet, the max_register is 13h.
ARRAY_SIZE(pcm1681_reg_defaults) + 1 is 18 which is wrong.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the datasheet, the max_register is 13h.
ARRAY_SIZE(pcm1681_reg_defaults) + 1 is 18 which is wrong.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Tegra124 introduces some small changes to the layout of some registers.
Modify the affected drivers to program those registers appropriately
based on which SoC they're running on.
Tegra124 also introduced some new modules on the AHUB configlink register
bus. These will require new entries in configlink_clocks[] in the AHUB
driver. However, supporting that change likely relies on switching Tegra
to the common reset framework, so I'll defer that change for now.
Based-on-work-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Based-on-work-by: Songhee Baek <sbaek@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When converting this driver to devm_ioremap_resource, the removal of this now
unneeded function has been forgotten.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit ab78029 (drivers/pinctrl: grab default handles from device core),
we can rely on device core for setting the default pins.
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Acked-by: Linus Walleij <linus.walleij@linaro.org> (personally at LCE13)
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When audmux_clk is used and clk_prepare_enable function succeed,
the memory alloc'd to buf variable is leaked
Signed-off-by: Felipe Pena <felipensp@gmail.com>
Reviewed-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The create_bind_cap_vol_ctl does not create any control indicating
that an inverted dmic is present. Therefore, create multiple
capture volumes in this scenario, so we always have some indication
that the internal mic is inverted.
This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13
(both are based on the CX20590 codec), but the fix is generic and
could be needed for other codecs/machines too.
Thanks to Szymon Acedański for the pointer and a draft patch.
BugLink: https://bugs.launchpad.net/bugs/1239392
BugLink: https://bugs.launchpad.net/bugs/1227491
Reported-by: Szymon Acedański <accek@mimuw.edu.pl>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
checking id in rsnd_dai_get() is good idea
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Un-implemented platform callback is not error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add more infrastructure (i.e. routes, muxes, switches) to the mc13783 DAPM.
Signed-off-by: Steffen Trumtrar <s.trumtrar@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add more controls to the alsa mixer infrastructure.
Signed-off-by: Steffen Trumtrar <s.trumtrar@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using the headset mic model will cause the headset mic to be labeled
"headset mic" instead of just "mic".
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The external mic showed up with a precense detect of "always present",
essentially disabling the internal mic. Therefore turn off presence
detection for this pin.
Note: The external mic seems not yet working, but an internal mic is
certainly better than no mic at all.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1227093
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to avoid trying to use an external clock or supply for an
on-chip supply prior to it being enabled move the clock and regulator
supply events to a separate step in DAPM sequencing from normal supply
events.
This should have minimal practical impact since these widgets are sorted
using SND_SOC_NOPM which is a negative value and hence sorted
separately to any real register writes, though it may be relevant if
supplies have event callbacks only.
Signed-off-by: Mark Brown <broonie@linaro.org>
Within a DAPM sequence we normally don't care about when exactly a register
write has completed so long as they happen in the order we requested. This
means that we can issue most of the writes we do asynchronously which
should maximise the ability of the underlying frameworks to keep the
hardware busy, providing a small performance improvement on some systems.
We currently ensure that all writes are completed both when changing to a
different device and when calling into the regulator and clock frameworks.
This should ensure that the previous ordering is maintained.
We also ensure that writes are completed prior to calling into widget
event functions since some event functions implement delays. This
should be improved in future so that widgets can disable this sync in
order to add extra writes.
Signed-off-by: Mark Brown <broonie@linaro.org>
In the case where we have both line out and more than stereo speakers,
the speaker DACs will end up in extra_out_nid.
In fact, AFAIU, speakers are the only ones that can end up in extra_out_nid,
and if we have several of those, they should be surround outputs
rather than copy front.
BugLink: https://bugs.launchpad.net/bugs/1236965
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS N56VZ needs a fixup for the bass speaker pin, which was already
provided via model=asus-mode4.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently hdmi_setup_audio_infoframe() reprograms the HDA channel
mapping only when the infoframe is not up-to-date or the non-PCM flag
has changed.
However, when just the channel map has been changed, the infoframe may
still be up-to-date and non-PCM flag may not have changed, so the new
channel map is not actually programmed into the HDA codec.
Notably, this failing case is also always triggered when the device is
already in a prepared state and a new channel map is configured while
changing only the channel positions (for example, plain
"speaker-test -c2 -m FR,FL").
Fix that by always programming the channel map in
hdmi_setup_audio_infoframe(). Tested on Intel HDMI.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The two paths were not connected in the DAPM route causing the associated
routes to be non working and the following warnings printed in the logs:
tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths
tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for dev speed USB_SPEED_WIRELESS in
snd_usb_parse_datainterval which allows the usb sound core to create
ISO urbs with the correct number and size of buffers.
Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By using the new virtual DAPM mixer controls it is possible to remove the
twl4030 specific implementation of virtual controls.
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>
Allow channel map debugging for both automatic and manual channel maps,
and print CA always when updating infoframe.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the available channel maps TLV only contains channel maps that
are limited to the traditional 7.1 speakers.
Since the other HDMI channel mapping functions have been fixed to
properly handle all CEA-861-E specified speakers, allow them to be
listed.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some speakers and slots the CEA slot <-> speaker assignment depends
on the used CEA Channel Allocation value.
Therefore the from_cea_slot() and to_cea_slot() helpers currently only
work correctly for the regular 7.1 speakers.
Fix them to work with all speakers, taking the re-ordered CA index as
input and adapting use sites accordingly.
This change allows manual channel mapping to actually work for all CEA
allocated speakers. Additionally, this fixes incorrect channel map
reporting in automatic channel mapping mode when an affected speaker
position is used (e.g. 6.1 map which contains an RC speaker).
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdmi_manual_setup_channel_mapping() and hdmi_std_setup_channel_mapping
try to assign ALSA channels to HDMI channel slots and disable (i.e.
silence) other slots.
However, they try to disable a slot by using AC_VERB_SET_CHAN_SLOT with
parameter ((alsa_ch << 8) | 0xf), while the correct parameter is
((0xf << 8) | hdmi_slot), i.e. the slot should be unassigned, not the
ALSA channel.
Fix that by actually disabling the unused slots.
Note that this bug did not cause any (reported) issues because slots
incorrectly having audio are normally ignored by a receiver if the CEA
channel allocation used does not map that slot to any speaker.
Additionally, the converter channel count configuration limits the
number of actually active channels in any case.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By using the new virtual DAPM mixer controls it is possible to remove the
twl6040 specific implementation of virtual controls.
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>
Currently the converter channel count is set to the number of actual
input channels. The audio infoframe channel count field is set
similarly.
However, sometimes the used channel map does not map all input channels
to outputs. Notably, 3 channel modes (e.g. 2.1) require a dummy input
channel so there are 4 input channels. According to the HDA
specification, converter channel count should be programmed according to
the number of _active_ channels.
On Intel HDMI codecs (but not on NVIDIA), setting the converter channel
to a higher value than there are actually mapped channels to HDMI slots
will cause no audio to be output at all.
Note that the effects of this issue are currently partially masked by
other bugs that prevent the driver from actually unmapping channels in
certain cases. For example, if a 4 channel stream is first created and
prepared, it gets a FL,FR,RL,RR mapping (ALSA->HDMI slot mapping 0->0,
1->1, 2->4, 3->5). If one thereafter assigns a FR,FL,FC mapping to it,
the driver will remap 2->3 but fail to unmap 2->4 and 3->5, so there are
still 4 active channels and the issue will not trigger in this case.
These bugs will be fixed separately.
Fix the channel counts in the converter channel count field and in the
audio infoframe channel count field to match the actual number of active
channels.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdmi_std_setup_channel_mapping() selects a Channel Allocation according
to the sink reported speaker mask, preferring the ALSA standard layouts.
If the channel allocation is not one of the ALSA standard layouts, the
ALSA channels are mapped directly to HDMI channels in order. However,
the function does not take into account that there a holes in the HDMI
channel map.
Additionally, the function tries to disable a slot by using
AC_VERB_SET_CHAN_SLOT with parameter ((alsa_ch << 8) | 0xf), while the
correct parameter is ((0xf << 8) | hdmi_slot), i.e. the slot should be
unassigned, not the ALSA channel.
Fix both of the issues for non-ALSA-default layouts.
Tested on Intel HDMI with a speaker mask of FL | FR | FC | RC, which
causes CA 0x06 to be selected for 4-channel audio, which causes
incorrect output (sound destined to RC goes to FC and FC goes nowhere)
without the patch.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
hdmi_setup_fake_chmap() is supposed to set the reported channel map when
the channel map is not specified by the user.
However, the function indexes channel_allocations[] with a wrong value
and extracts the wrong nibble from hdmi_channel_mapping[], causing wrong
channel maps to be shown.
Fix those issues.
Tested on Intel HDMI to correctly generate various channel maps, for
example 3,4,14,15,7,8,5,6 (instead of incorrect 3,4,8,7,5,6,14,0) for
standard 7.1 channel audio. (Note that the side and rear channels are
reported as RL/RR and RLC/RRC, respectively, as per the CEA-861
standard, instead of the more traditional SL/SR and RL/RR.)
Note that this only fixes the layouts that only contain traditional 7.1
speakers (2.0, 2.1, 4.0, 5.1, 7.1, etc.). E.g. the rear center of 6.1
is still being shown wrongly due to an issue with from_cea_slot()
which will be fixed in a later patch.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VMID widgets behave very similar to signal generator widgets. Both are always
considered to be powered up. This means that we need to ignore the VMID widgets
in the same way as signal generator widgets when calculating the DAPM context's
target bias level. Otherwise the presence of a VMID widget, regardless whether
it is on an active path or not, will cause the DAPM context to be powered up.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
On imx31 with mc13783 codec the FIQ is not necessary and not enabled
as DMA transfer is available.
Change the probe() function to fail only if both FIQ and DMA are not
available.
Signed-off-by: Philippe Rétornaz <philippe.retornaz@epfl.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
As Clemens Ladisch kindly explained:
"Please note that there are two methods to identify alternate settings:
the number, which is the value in bAlternateSetting, and the index,
which is the index in the descriptor array. There might be some wording
in the USB spec that these two values must be the same, but in reality,
[insert standard rant about firmware writers], bAlternateSetting
must be treated as a random ID value."
This patch changes the name to express the correct usage semantics.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of snd_usb_endpoint_deactivate() is not used,
make the function have no return value.
Update the documentation to reflect what the function is actually
doing.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If an endpoint in use, its associated URBs should not be
deactivated.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only call site for deactivate_endpoints() at snd_usb_hw_free().
The return value is not checked there, as it is irrelevant if it
fails on hw_free.
This patch moves the deactivation of the endpoints directly into
snd_usb_hw_free().
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the format is not actually used in sync_ep_set_params(),
there is no need to pass it down.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On this machine, DAC on node 0x03 seems to give mono output.
Also, it needs additional patches for headset mic support.
It supports CTIA style headsets only.
Alsa-info available at the bug link below.
Cc: stable@kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1236228
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b0e0a4d ("ASoC: omap: Enable COMPILE_TEST build for DT platforms")
added two incorrect CONFIG_ARCH_ARM dependencies making impossible to select
audio support for Nokia RX-51. Fix this by using correct CONFIG_ARM.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 42810d (ASoC: imx-mc13783: Add audmux settings for mx27pdk) broke
the sound on mx31moboard. Restore back the audmux setting on such boards.
Signed-off-by: Philippe Rétornaz <philippe.retornaz@epfl.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
irq_of_parse_and_map() returns 0 on error, not NO_IRQ.
Fix the following xtensa:allmodconfig build error.
sound/soc/fsl/fsl_ssi.c:705:26: error: 'NO_IRQ' undeclared (first use in this function)
make[4]: *** [sound/soc/fsl/fsl_ssi.o] Error 1
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Cc: Grant Likely <grant.likely@linaro.org>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the system returns from suspend, it looses its configuration. Most
of it is restored by running a normal audio stream startup, but the DAI
format is left unset as that's configured on the audio device creation.
Hence, it suffices here to care for the registers which are touched by
davinci_mcasp_set_dai_fmt() and restore them when the system is resumed.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When going to suspend, shut down all channels and re-do the init
procedure at resume time.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We'll need to call code to initialize and reset the codec again at
resume time, so factor it out first.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We need to access the charge_period and start_mid_z values from other
places later, so move them to the private struct.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The frame check in i_usX2Y_urb_complete() and
i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as
described in this LAU thread:
http://linuxaudio.org/mailarchive/lau/2013/5/20/200177
This patch removes the check code entirely.
Cc: fzu@wemgehoertderstaat.de
Reported-by: Dr Nicholas J Bailey <nicholas.bailey@glasgow.ac.uk>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
writel() supposes the first argument of type unsigned int. This fix the
warning:
sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open':
sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly
truncated to unsigned type [-Woverflow]
Signed-off-by: Vladimir Murzin <murzin.v@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit e8f00c1b01 (Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm)
fixed a conflict, but missed to add the .remove function back,which causes the
following build warning:
sound/soc/fsl/imx-sgtl5000.c:185:12: warning: 'imx_sgtl5000_remove' defined but not used [-Wunused-function]
Fix the warning by adding the .remove function back.
Reported-by: Olof Johansson <olof@lixom.net>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
dev_ prints are already prefixed by ": " before format string so there is no
need for extra spaces.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute
control. When a machine is booted after Windows 8, the GPIO pin is
turned off and it results in the silent output.
This patch adds the manual fixup of GPIO bit 0 for this model.
Reported-by: Christopher <DIDI2002@web.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds LED support for the Native Instruments Maschine
Controller. It adds ALSA controls for dimming the LEDs of all
buttons and the backlight of the two displays.
Signed-off-by: Hannes Gräuler <hgraeule@uos.de>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet:
http://www.ti.com/lit/ds/slas253a/slas253a.pdf
The weird thing is that the chip is physically marked 320AD91.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing too exciting here, all driver specific except for the fix from
Liam for DPCM systems which have both front and back end DAIs which is
not yet used by anything in mainline.
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Merge tag 'asoc-v3.12-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.12
Nothing too exciting here, all driver specific except for the fix from
Liam for DPCM systems which have both front and back end DAIs which is
not yet used by anything in mainline.
Reset the device on probe to make sure that the register settings match the
register cache defaults.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
More thorough testing showed that these verbs were necessary to
improve quality of the internal mic. Patch originally from Realtek.
BugLink: https://bugs.launchpad.net/bugs/1231931
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
