The pcm3008_codec struct is not used outside of davinci-sffsdr.c, so make it
static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sn95031_codec struct is not used outside of sn95031.c, so make it static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mop500_dai_links struct is not used outside of mop500.c, so make it static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mop500_ab8500_startup(), the mop500_ab8500_shutdown() and the
mop500_ab8500_hw_params() function are not used outside of mop500_ab8500, so
make them static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The bf5xx_tdm_pcm_ops struct is only used in bf5xx-tdm-pcm.c, so make it static.
Cc: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mmp_sspa_dai struct is only used in mmp-sspa.c, so make it static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mmp_pcm_ops and mmp_soc_platform struct as well as the mmp_pcm_new()
function are only used in mmp-pcm.c, so make them static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The spdif_soc_dai_probe function is only used in spdif_out.c, so make it static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The spdif_in_dai struct is not used outside of spdif_in.c, so make it static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The kirkwood_dma_ops struct is not used outside of kirkwood-dma.c, so make it
static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ep93xx_i2s_dma_data struct is not used outside of ep93xx-i2s.c, so make it
static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
kasprintf calculates the size of the result string, allocates a buffer large
enough to hold the string and then performs the format string operation. There
are a couple of places in ASoC where these three steps are done by hand and
where kasprintf can be used instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We use the same code to initialize the runtime pcm based on the
snd_soc_pcm_stream struct on both the playback and capture path. Factor this
code into a helper function to make things a bit more tidy.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_set_runtime_hwparams() is the only PCM related function that lives in
soc-core.c. All other PCM related functions live in soc-pcm.c, so move
snd_soc_set_runtime_hwparams() over as well for a bit more consistency.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are two return paths which don't kfree(name).
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that all of the Tegra device trees have been updated to represent
the required audio clocks, remove the compatibility code from the Tegra
ASoC utility code, and always use clk_get() rather than clk_get_sys().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting ep39xx to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit d4702b189c ("sound: Fix make allmodconfig on MIPS") added a
(negative) dependency on ISA_DMA_SUPPORT_BROKEN. Since that Kconfig
symbol doesn't exist, this dependency will always evaluate to true.
Apparently GENERIC_ISA_DMA_SUPPORT_BROKEN was meant to be used here.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to
use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and
SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma
params code can be removed now.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Later WM8994 devices support an enhanced accuracy FLL divisor mode called
EFS which allows more precise selection of fractional source to output
ratios. Support this on relevant devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation to switching the jz4740 clk driver to the common clk framework
update the clk enable/disable calls to clk_prepare_enable/clk_disable_unprepare.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the AB8500 user manual AD to Slot register multiplexer
accept values from 0 to 15 where:
0 to 7 corresponds to AD_OUTx slots
8 to 11 corresponds to zero output
12 to 15 sets the output in tristate mode
Update enum_ad_to_slot_map array to reflect this definition.
This also allows alsamixer to properly display the default
configuration, as all controls are set to tristate (=12) at reset.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop ab8500 clock gating widgets from mop500_ab8500_ctrls, as these bits
are already controlled by ab8500 codec driver and should not be exposed
to the user.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update mop500_ab8500_machine_init to register mop500_ab8500_ctrls as
card control instead of codec control, as it only contains
SOC_DAPM_PIN_SWITCH definitions.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop dev_set_drvdata as this is handled in the core and use
devm_request_and_ioremap instead of devm_ioremap to properly register the
address range used
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting spear to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add devicetree support for this dummy audio soc driver.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add devicetree support for this dummy audio soc driver.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Transceiver usually means receiver + transmitter. This codec can do only
transmit. Update driver accordingly.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After a 'reboot' command in Linux or after pressing the system's reset button
the sgtl5000 driver fails to probe:
sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000
sgtl5000 0-000a: ASoC: failed to probe CODEC -19
imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19
imx-sgtl5000 sound.12: snd_soc_register_card failed (-19)
sgtl5000 codec does not have a reset line, nor a reset command in software, so
after a system reset the codec does not contain the default register values
from sgtl5000_reg_defaults[] anymore, as these are only valid after a
power-on-reset cycle.
Fix this issue by explicitly reading all the reset register values from
sgtl5000_reg_defaults[] and writing them back into sgtl5000 to ensure a sane
state.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The usual place for reading chip ID is inside i2c_probe, so move it there and
also convert it to regmap.
sgtl5000_enable_regulators() needs to read the chip revision, so keep the
revision check there.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It makes no sense to hardcode the I2C bus into the codec_name field.
cpu_dai_name and platform_name are also overwritten later in
mxs_sgtl5000_probe_dt().
So remove the three fields, as mxs platform is dt-only platform.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Variable 'ret' is not needed here, so just remove it.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The list below the comment relates to sgtl5000 registers addresses, so change
the comment to improve the description.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit ab78029 (drivers/pinctrl: grab default handles from device core),
we can rely on device core for handling pinctrl.
So remove devm_pinctrl_get_select_default() from the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit ab78029 (drivers/pinctrl: grab default handles from device core),
we can rely on device core for handling pinctrl.
So remove devm_pinctrl_get_select_default() from the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It makes no sense to hardcode the I2C bus into the codec_name field.
cpu_dai_name and platform_name are also overwritten later in
mxs_sgtl5000_probe_dt().
So remove the three fields, as mxs platform is dt-only platform.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Variable 'ret' is not needed here, so just remove it.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return on success instead of entering the error path.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices may benefit from being able to start some parts of the widget
power up/down sequence earlier on in the sequence than the point at which
the final power state is committed. Support these by providing events which
are called before any power state changes are done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Since commit ab78029 (drivers/pinctrl: grab default handles from device
core) we can rely on device core for handling pinctrl so remove
devm_pinctrl_get_select_default() from the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Bo Shen <voice.shen@atmel.com>
Add initial support for runtime tuning for the ADSP cores. This
is achieved by exposing the coefficient configuration blocks as
ALSA binary controls.
The current code assumes that no controls on the DSP are volatile.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
This reverts commit affdb62b81.
The commit introduced a regression with AD codecs where the stream is
always clean up. Since the patch is just a minor optimization and
reverting the commit fixes the issue, let's just revert it.
Reported-and-tested-by: Michael Burian <michael.burian@sbg.at>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When set dmic_samplephase and dmic_clk_rate bits for dmic_cfg,
current code checks pdata->dmic_data_sel which is wrong.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
AD slots definitions for ab8500 codec were erroneously swapped between
even and odd channels. Fix this by swapping the definitions to be
coherent with the channel number.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a revised patch based on Mengdong Lin's fix patch, which is a
supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for
Haswell to enable all pin and convertor widgets].
Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets
when the HD-A controller changes state from D3 to D0. So when the
controller resumes after a system or runtime suspend, these widgets
are disabled and programming these widgets to D0 will cause H/W error
and codec will not respond.
In addition, we found out that some BIOS disables the pins at S3
although it shows up at boot. This confuses the driver utterly, and
the hardware falls into the fatal communication error like the above.
So in this patch, we apply intel_haswell_enable_all_pins() not only as
a fixup to a certain device (with 8086:2010) but to all Haswell
machines. The codec driver basically assumes that all pins are
exposed, so it's anyway better to see them from the beginning. Even
if all pins and converters are shown by this call, there should be no
regression in practice: the pin default configurations are still kept,
thus the disabled pins are handled as disabled by the driver
properly.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return -ENOMEM in the memory malloc of 'out' and 'img_swap' error
handling case instead of 0, as done elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The interrupt handler azx_interrupt will call azx_update_rirb,
which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event
will dereference chip->bus pointer.
The problem is we alloc chip->bus in azx_codec_create
which will be called after we enable IRQ and enable unsolicited
event in azx_probe.
This will cause Oops due dereference NULL pointer. I meet it, good luck:)
[Rearranged the NULL check before the tracepoint and added another
NULL check of bus->workq -- tiwai]
Signed-off-by: Wang YanQing <udknight@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Cc: Vivien Chappelier <vivien.chappelier@linux-mips.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Cc: Thomas Bogendoerfer <tsbogend@alpha.fanken.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The older Conexant codecs have up to two EAPDs and these are supposed
to be rather statically turned on. The new generic parser code
assumes the dynamic on/off per path usage, thus it resulted in the
silent output on some machines.
This patch fixes the problem by simply assuming the static EAPD on for
such old Conexant codecs as we did until 3.8 kernel.
Reported-and-tested-by: Christopher K. <c.krooss@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
The compile of soundcard.c is broken on MIPS when allmodconfig is used
because of the missing MAX_DMA_CHANNELS definition. As a simple
workaround, just add a Kconfig dependency.
Reported-by: Andrew Morton <akpm@linux-foundation.org>
Cc: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audio driver mistakenly allows 64 bit addresses to be created for
the audio driver on Nvidia GPUs. Unfortunately, the hardware normally
only supports up to 40 bits of DMA. This can cause system panics as
well as misdirected data when the address is > 40 bits as the upper
part the address is truncated.
Signed-off-by: Mike Travis <travis@sgi.com>
Reviewed-by: Mike Habeck <habeck@sgi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 0998d06310 (device-core: Ensure drvdata = NULL when no
driver is bound) removes the need to set driver data field to
NULL.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more bug fixes, the DAPM clock fix is actually a driver specific
one since currently there's only one user of the clock support due to
the problems relying on the clock API.
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Merge tag 'asoc-v3.10-4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A few more bug fixes, the DAPM clock fix is actually a driver specific
one since currently there's only one user of the clock support due to
the problems relying on the clock API.
According documentation bit ACLKRPOL is set to 0 (receiver samples data
on falling edge) and when set to 1 (receiver samples data on rising edge).
I2S data are always sampled on falling edge and valid during rising edge
of bit clock. So in case of capture data transmitter sample data on falling
edge and macsp must read then on rising edge.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
More multiplatform enablement for ARM platforms. The ones converted in
this branch are:
- bcm2835
- cns3xxx
- sirf
- nomadik
- msx
- spear
- tegra
- ux500
We're getting close to having most of them converted!
One of the larger platforms remaining is Samsung Exynos, and there are
a bunch of supporting patches in this merge window for it. There was a
patch in this branch to a early version of multiplatform conversion,
but it ended up being reverted due to need of more bake time. The
revert commit is part of the branch since it would have required
rebasing multiple dependent branches and they were stable by then.
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Merge tag 'multiplatform-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC multiplatform updates from Olof Johansson:
"More multiplatform enablement for ARM platforms. The ones converted
in this branch are:
- bcm2835
- cns3xxx
- sirf
- nomadik
- msx
- spear
- tegra
- ux500
We're getting close to having most of them converted!
One of the larger platforms remaining is Samsung Exynos, and there are
a bunch of supporting patches in this merge window for it. There was
a patch in this branch to a early version of multiplatform conversion,
but it ended up being reverted due to need of more bake time. The
revert commit is part of the branch since it would have required
rebasing multiple dependent branches and they were stable by then"
* tag 'multiplatform-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (70 commits)
mmc: sdhci-s3c: Fix operation on non-single image Samsung platforms
clocksource: nomadik-mtu: fix up clocksource/timer
Revert "ARM: exynos: enable multiplatform support"
ARM: SPEAr13xx: Fix typo "ARCH_HAVE_CPUFREQ"
ARM: exynos: enable multiplatform support
rtc: s3c: make header file local
mtd: onenand/samsung: make regs-onenand.h file local
thermal/exynos: remove unnecessary header inclusions
mmc: sdhci-s3c: remove platform dependencies
ARM: samsung: move mfc device definition to s5p-dev-mfc.c
ARM: exynos: move debug-macro.S to include/debug/
ARM: exynos: prepare for sparse IRQ
ARM: exynos: introduce EXYNOS_ATAGS symbol
ARM: tegra: build assembly files with -march=armv7-a
ARM: Push selects for TWD/SCU into machine entries
ARM: ux500: build hotplug.o for ARMv7-a
ARM: ux500: move to multiplatform
ARM: ux500: make remaining headers local
ARM: ux500: make irqs.h local to platform
ARM: ux500: get rid of <mach/[hardware|db8500-regs].h>
...
Here is a collection of cleanup patches. Among the pieces that stand out are:
- The deletion of h720x platforms
- Split of at91 non-dt platforms to their own Kconfig file to keep them separate
- General cleanups and refactoring of i.MX and MXS platforms
- Some restructuring of clock tables for OMAP
- Convertion of PMC driver for Tegra to dt-only
- Some renames of sunxi -> sun4i (Allwinner A10)
- ... plus a bunch of other stuff that I haven't mentioned
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Merge tag 'cleanup-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC cleanup from Olof Johansson:
"Here is a collection of cleanup patches. Among the pieces that stand
out are:
- The deletion of h720x platforms
- Split of at91 non-dt platforms to their own Kconfig file to keep
them separate
- General cleanups and refactoring of i.MX and MXS platforms
- Some restructuring of clock tables for OMAP
- Convertion of PMC driver for Tegra to dt-only
- Some renames of sunxi -> sun4i (Allwinner A10)
- ... plus a bunch of other stuff that I haven't mentioned"
* tag 'cleanup-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (119 commits)
ARM: i.MX: remove unused ARCH_* configs
ARM i.MX53: remove platform ahci support
ARM: sunxi: Rework the restart code
irqchip: sunxi: Rename sunxi to sun4i
irqchip: sunxi: Make use of the IRQCHIP_DECLARE macro
clocksource: sunxi: Rename sunxi to sun4i
clocksource: sunxi: make use of CLKSRC_OF
clocksource: sunxi: Cleanup the timer code
ARM: at91: remove trailing semicolon from macros
ARM: at91/setup: fix trivial typos
ARM: EXYNOS: remove "config EXYNOS_DEV_DRM"
ARM: EXYNOS: change the name of USB ohci header
ARM: SAMSUNG: Remove unnecessary code for dma
ARM: S3C24XX: Remove unused GPIO drive strength register definitions
ARM: OMAP4+: PM: Restore CPU power state to ON with clockdomain force wakeup method
ARM: S3C24XX: Removed unneeded dependency on CPU_S3C2412
ARM: S3C24XX: Removed unneeded dependency on CPU_S3C2410
ARM: S3C24XX: Removed unneeded dependency on ARCH_S3C24XX for boards
ARM: SAMSUNG: Fix typo "CONFIG_SAMSUNG_DEV_RTC"
ARM: S5P64X0: Fix typo "CONFIG_S5P64X0_SETUP_SDHCI"
...
Pull VFS updates from Al Viro,
Misc cleanups all over the place, mainly wrt /proc interfaces (switch
create_proc_entry to proc_create(), get rid of the deprecated
create_proc_read_entry() in favor of using proc_create_data() and
seq_file etc).
7kloc removed.
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (204 commits)
don't bother with deferred freeing of fdtables
proc: Move non-public stuff from linux/proc_fs.h to fs/proc/internal.h
proc: Make the PROC_I() and PDE() macros internal to procfs
proc: Supply a function to remove a proc entry by PDE
take cgroup_open() and cpuset_open() to fs/proc/base.c
ppc: Clean up scanlog
ppc: Clean up rtas_flash driver somewhat
hostap: proc: Use remove_proc_subtree()
drm: proc: Use remove_proc_subtree()
drm: proc: Use minor->index to label things, not PDE->name
drm: Constify drm_proc_list[]
zoran: Don't print proc_dir_entry data in debug
reiserfs: Don't access the proc_dir_entry in r_open(), r_start() r_show()
proc: Supply an accessor for getting the data from a PDE's parent
airo: Use remove_proc_subtree()
rtl8192u: Don't need to save device proc dir PDE
rtl8187se: Use a dir under /proc/net/r8180/
proc: Add proc_mkdir_data()
proc: Move some bits from linux/proc_fs.h to linux/{of.h,signal.h,tty.h}
proc: Move PDE_NET() to fs/proc/proc_net.c
...
The missing break here means that we always return early and the
function is a no-op.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
When McASP is bit clock and frame clock master enable pin output for rx clocks.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update dapm_clock_event to use clk_prepare_enable and
clk_disable_unprepare.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pull media update from Mauro Carvalho Chehab:
- OF documentation and patches at core and drivers, to be used by for
embedded media systems
- some I2C drivers used on go7007 were rewritten/promoted from staging:
sony-btf-mpx, tw2804, tw9903, tw9906, wis-ov7640, wis-uda1342
- add fimc-is driver (Exynos)
- add a new radio driver: radio-si476x
- add a two new tuners: r820t and tuner_it913x
- split camera code on em28xx driver and add more models
- the cypress firmware load is used outside dvb usb drivers. So, move
it to a common directory to make easier to re-use it
- siano media driver updated to work with sms2270 devices
- several work done in order to promote go7007 and solo6x1x out of
staging (still, there are some pending issues)
- several API compliance fixes at v4l2 drivers that don't behave as
expected
- as usual, lots of driver fixes, improvements, cleanups and new device
addition at the existing drivers.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (831 commits)
[media] cx88: make core less verbose
[media] em28xx: fix oops at em28xx_dvb_bus_ctrl()
[media] s5c73m3: fix indentation of the help section in Kconfig
[media] cx25821-alsa: get rid of a __must_check warning
[media] cx25821-video: declare cx25821_vidioc_s_std as static
[media] cx25821-video: remove maxw from cx25821_vidioc_try_fmt_vid_cap
[media] r820t: Remove a warning for an unused value
[media] dib0090: Fix a warning at dib0090_set_EFUSE
[media] dib8000: fix a warning
[media] dib8000: Fix sub-channel range
[media] dib8000: store dtv_property_cache in a temp var
[media] dib8000: warning fix: declare internal functions as static
[media] r820t: quiet gcc warning on n_ring
[media] r820t: memory leak in release()
[media] r820t: precendence bug in r820t_xtal_check()
[media] videodev2.h: Remove the unused old V4L1 buffer types
[media] anysee: Grammar s/report the/report to/
[media] anysee: Initialize ret = 0 in anysee_frontend_attach()
[media] media: videobuf2: fix the length check for mmap
[media] em28xx: save isoc endpoint number for DVB only if endpoint has alt settings with xMaxPacketSize != 0
...
* patchwork: (831 commits)
[media] cx88: make core less verbose
[media] em28xx: fix oops at em28xx_dvb_bus_ctrl()
[media] s5c73m3: fix indentation of the help section in Kconfig
[media] cx25821-alsa: get rid of a __must_check warning
[media] cx25821-video: declare cx25821_vidioc_s_std as static
[media] cx25821-video: remove maxw from cx25821_vidioc_try_fmt_vid_cap
[media] r820t: Remove a warning for an unused value
[media] dib0090: Fix a warning at dib0090_set_EFUSE
[media] dib8000: fix a warning
[media] dib8000: Fix sub-channel range
[media] dib8000: store dtv_property_cache in a temp var
[media] dib8000: warning fix: declare internal functions as static
[media] r820t: quiet gcc warning on n_ring
[media] r820t: memory leak in release()
[media] r820t: precendence bug in r820t_xtal_check()
[media] videodev2.h: Remove the unused old V4L1 buffer types
[media] anysee: Grammar s/report the/report to/
[media] anysee: Initialize ret = 0 in anysee_frontend_attach()
[media] media: videobuf2: fix the length check for mmap
[media] em28xx: save isoc endpoint number for DVB only if endpoint has alt settings with xMaxPacketSize != 0
...
Conflicts:
drivers/media/pci/cx25821/cx25821-video.c
drivers/media/platform/Kconfig
Current code does this:
be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1])
Which is effectively (neglecting the index):
be16_to_cpu(be16_to_cpu(*((u16 *) buf)))
This means the int16 in the buffer is not converted at all.
Daniel Mack confirmed that the driver works on little endian
CPUs, leading to the conclusion that the device-side structure
is actually little endian.
This changes the code to use le16_to_cpu().
Caught by sparse.
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here's the big char / misc driver update for 3.10-rc1
A number of various driver updates, the majority being new functionality
in the MEI driver subsystem (it's now a subsystem, it started out just a
single driver), extcon updates, memory updates, hyper-v updates, and a
bunch of other small stuff that doesn't fit in any other tree.
All of these have been in linux-next for a while
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'char-misc-3.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc
Pull char/misc driver update from Greg Kroah-Hartman:
"Here's the big char / misc driver update for 3.10-rc1
A number of various driver updates, the majority being new
functionality in the MEI driver subsystem (it's now a subsystem, it
started out just a single driver), extcon updates, memory updates,
hyper-v updates, and a bunch of other small stuff that doesn't fit in
any other tree.
All of these have been in linux-next for a while"
* tag 'char-misc-3.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc: (148 commits)
Tools: hv: Fix a checkpatch warning
tools: hv: skip iso9660 mounts in hv_vss_daemon
tools: hv: use FIFREEZE/FITHAW in hv_vss_daemon
tools: hv: use getmntent in hv_vss_daemon
Tools: hv: Fix a checkpatch warning
tools: hv: fix checks for origin of netlink message in hv_vss_daemon
Tools: hv: fix warnings in hv_vss_daemon
misc: mark spear13xx-pcie-gadget as broken
mei: fix krealloc() misuse in in mei_cl_irq_read_msg()
mei: reduce flow control only for completed messages
mei: reseting -> resetting
mei: fix reading large reposnes
mei: revamp mei_irq_read_client_message function
mei: revamp mei_amthif_irq_read_message
mei: revamp hbm state machine
Revert "drivers/scsi: use module_pcmcia_driver() in pcmcia drivers"
Revert "scsi: pcmcia: nsp_cs: remove module init/exit function prototypes"
scsi: pcmcia: nsp_cs: remove module init/exit function prototypes
mei: wd: fix line over 80 characters
misc: tsl2550: Use dev_pm_ops
...
Some Asihpi formats are not supported or invalid, and their mapping to
ALSA format is set to -1.
Before performing the format conversion into ALSA bitwise formats,
add a consistency check for the requested format, as done in
snd_card_asihpi_playback_formats().
Compile tested only.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
for reading compressed data, we need to allow when we are paused, draining or
stopped.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Cc: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables better volume controls than the current model parser.
Also, because the original quirk for X220 was added to fix
docking station support, add the TP410 fixup instead.
Reported-by: Willian Jon McCann <william.jon.mccann@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes in the USB API ("implement new semantics for
URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the
default, and changed this flag to mean that URBs can be delayed.
This is not the behaviour wanted by any of the audio drivers because
it leads to discontinuous playback with very small period sizes.
Therefore, our URBs need to be submitted without this flag.
Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merging in fixes since there's a conflict in the omap4 clock tables caused by
it.
* fixes: (245 commits)
ARM: highbank: fix cache flush ordering for cpu hotplug
ARM: OMAP4: hwmod data: make 'ocp2scp_usb_phy_phy_48m" as the main clock
arm: mvebu: Fix the irq map function in SMP mode
Fix GE0/GE1 init on ix2-200 as GE0 has no PHY
ARM: S3C24XX: Fix interrupt pending register offset of the EINT controller
ARM: S3C24XX: Correct NR_IRQS definition for s3c2440
ARM i.MX6: Fix ldb_di clock selection
ARM: imx: provide twd clock lookup from device tree
ARM: imx35 Bugfix admux clock
ARM: clk-imx35: Bugfix iomux clock
+ Linux 3.9-rc6
Signed-off-by: Olof Johansson <olof@lixom.net>
Conflicts:
arch/arm/mach-omap2/cclock44xx_data.c
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The virt_to_bus/bus_to_virt functions have been deprecated
for as long as I can remember, and they are used in very
few remaining instances, usually in obscure ISA device
drivers. The OSS sound drivers are the only ones that are
still used on the ARM architecture, and only on some of
the earliest StrongARM machines.
The problem for converting the OSS subsystem to use
dma_map_single instead is that the caller of virt_to_bus
does not have a device pointer, since the subsystem has
never been ported to use the common device infrastructure.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM cannot handle udelay for more than 2 miliseconds, so we
should use mdelay instead for those.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
....
It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.
Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.
Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.
There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.
When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.
Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We get a lot of build warnings from the msp driver like:
In file included from sound/soc/ux500/ux500_msp_dai.h:21:0,
from sound/soc/ux500/mop500.c:25:
sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: 'struct msp_i2s_platform_data' declared inside parameter list [enabled by default]
struct msp_i2s_platform_data *platform_data);
^
sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default]
The easiest solution is to add a declaration of the struct name.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length).
I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels.
Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback"
number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dma_request_slave_channel() is a more appropriate API for dmaengine
clients that adopt generic DMA bindings to call. Let's use it instead
of of_dma_request_slave_channel() to save <linux/of_dma.h> include.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The examples in Documentation/devicetree/bindings/dma/dma.txt recommends
the name for dma channel doing both RX and TX to be "rx-tx". This
becomes a common pattern that has been adopted by platforms that
converts to generic DMA bindings. Let's follow this common pattern in
generic-dmaengine-pcm.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These are being reported as being so noisy at high mic boost levels,
so they are unusable in practice.
Therefore artificially limit the boosts.
BugLink: https://bugs.launchpad.net/bugs/1089795
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting mxs to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MXS SAIF is only half-duplex so set the SNDRV_PCM_INFO_HALF_DUPLEX flag for
the PCM in order to prevent playback and capture from running at the same time.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some platforms which are half-duplex share the same DMA channel between the
playback and capture stream. Add support for this to the generic dmaengine PCM
driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.
More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.
Signed-off-by: Daniel Schürmann <daschuer@mixxx.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patch by Dan, we should clear the data to be
returned from certain compress ioctls, namely,
snd_compr_get_codec_caps() and snd_compr_get_params().
This time, we can simply replace kmalloc() with kzalloc().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the ->get_caps() function doesn't clear the buffer then there would
stack information leaked to userspace. For example,
soc_compr_get_caps() can return success without clearing the buffer.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously we just hard coded all streams as playback streams, this
patch checks the DAI to see if it is a capture or playback stream. It is
worth noting that at this time only unidirectional streams are
supported.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The app_pointer is managed locally by the compress core for memory
mapped DSPs but for DSPs that are not memory mapped this would have to
be manually updated from within the DSP driver itself, which is hardly
very idiomatic.
This patch switches to using the cumulative values to calculate the
available buffer space because these are already gracefully passed out
of the DSP driver to the compress core and otherwise should be
functionally equivalent.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is my example conversion of a few existing mmap users. The pcm
mmap case is one of the more straightforward ones.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
Use the generic dmaengine PCM driver instead of a custom implemention. There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.
That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.
The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.
ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.
This patch adds support for this by adding a boolean flag to the
audio format struct.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.
The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.
To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.
The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pin default configs are overridden via patch option, these are
evaluated before fixups are applied. Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.
We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The flag bus->shutdown implies that the control elements might have
been already destroyed. When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.
This patch just adds a check of the flag in the caller side for
avoiding such a crash.
Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily. So let it be for a while.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware revision of the codec is based at 0x40. Subtract that
before convering to ASCII. The same as it is done for 98095.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.
This patch is needed for playback and capturing on a codec-less systems,
as otherwise the PCM device nodes are not even created.
Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting imx to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platforms which don't use devicetree yet or have to optionally
support a non-devicetree way to request the DMA channel. The patch adds the
compat_request_channel and compat_filter_fn callbacks to the
snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it
will be used to request the DMA channel. If not dma_request_channel with
compat_filter_fn as the filter function will be used to request the channel.
The patch also exports the snd_dmaengine_pcm_request_chan() function, since
compat platforms will want to use it to request their DMA channel.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a generic dmaengine PCM driver. It builds on top of the
dmaengine PCM library and adds the missing pieces like DMA channel management,
buffer management and channel configuration. It will be able to replace the
majority of the existing platform specific dmaengine based PCM drivers.
Devicetree is used to map the DMA channels to the PCM device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and
snd_soc_unregister_platform with the difference that they won't allocate and
free the snd_soc_platform structure.
Also add snd_soc_lookup_platform which looks up a platform by the device it has
been registered for.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the dmaengine PCM library to allow the DMA channel to be requested
before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA
channel instead of a filter function and filter parameter as its parameters.
snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows
a dmaengine based PCM driver to request its channels before the substream is
opened.
The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan()
and snd_dmaengine_pcm_close_release_chan(), which have the same signature and
behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new
variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are
updated to use snd_dmaengine_pcm_open_request_chan() and
snd_dmaengine_pcm_close_release_chan().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When graphics initializes the HDMI chip, sometimes this leads to
pins going into D3 and right channel being muted. If the audio driver
finishes initialization before the graphic driver does, this situation
becomes permanent.
This is a workaround that checks for this situation and corrects it on
playback prepare. It has been verified working on at least one machine.
BugLink: https://bugs.launchpad.net/bugs/1167270
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default. Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add definitions for AC97 control register.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not. But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.
Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing. This leaves the aamix path even though the
loopback control is turned off.
This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D. Rename the codec time stamp
function appropriately.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices. However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions. With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.
Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.
Commit f5f165418c (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.
To work around all this, just disable autopm for all USB MIDI devices.
Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.
Buglink: https://bugs.launchpad.net/bugs/1169138
Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.
On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.
This patch implements that functionality as different capture sources.
Buglink: https://bugs.launchpad.net/bugs/1169143
Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.
Userspace expected: L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1
Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.
Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b2ca78717c (ARM: S3C24XX: make gta02.h local) already replaced
the GTA02_GPIO_* constants in neo1973-wm8753.c but forgot to remove the
inclusion of mach/gta02.h before moving the file out of mach/.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few updates, more than I'd like, fixing some relatively small issues
but mostly driver specific ones. Nothing wildly exciting so if it
doesn't make v3.9 it won't be the end of the world but it'd be nice.
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Merge tag 'asoc-v3.9-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.9
A few updates, more than I'd like, fixing some relatively small issues
but mostly driver specific ones. Nothing wildly exciting so if it
doesn't make v3.9 it won't be the end of the world but it'd be nice.
The plat/regs-iis.h and plat/regs-ac97.h files in the samsung platform
are only needed by the ASoC drivers, so they can be moved into the same
directory, as one more step towards a multiplatform build.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Firmwares may provide some firmware wide configuration regions which can
be configured by the coefficient files using the firmware ID as the
algorithm ID, include these in the algorithm list.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The idma_reg_addr_init function is used by the samsung i2s driver,
which can be a loadable module, so we have to export this function.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The second argument to the module_device_table macro must be the
name of the device id array. In the samsung i2s driver, there
was a small typo, resulting in a build error when building it
as a loadable module.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With multiplatform kernels, we cannot use hardwired IRQ
numbers in device drivers. This changes the idma driver
to use a proper resource, like all other drivers do.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.
Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only part of proc_dir_entry the code outside of fs/proc
really cares about is PDE(inode)->data. Provide a helper
for that; static inline for now, eventually will be moved
to fs/proc, along with the knowledge of struct proc_dir_entry
layout.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.
Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
There's already a device revision stored in the core data structure,
don't duplicate it in the CODEC driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For playback add the codec-side delay to the timestamp, for capture
subtract it. This brings the timestamps in line with the time that
was recently added to the delay reporting.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there
never was a corresponding Kconfig symbol so this test has always
evaluated to true. And since AB5500 support was removed in v3.5 it
appears safe to remove this test and a few lines of code.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This removes <mach/hardware.h> and <mach/db8500-regs.h>
from the Ux500, merging them into the local include
"db8500-regs.h" in mach-ux500. There is some impact
outside the ux500 machine, but most of it is dealt with
in earlier patches.
Contains portions of a clean-up patch from Arnd Bergmann.
Cc: Samuel Ortiz <sameo@linux.intel.com>
Cc: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Mike Turquette <mturquette@linaro.org>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
This header file only contains platform data structure definitions,
so it's straightforward to move.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
[Delete one include rather than move it]
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
The snd_pcm_hardware structs for playback and capture in the ux500 PCM are
identical, so remove one of them and use the same snd_pcm_hardware struct for
both playback and capture. Also move the defines used to initialize the
snd_pcm_hardware fields from ux500_pcm.h to ux500_pcm.c since that's the only
place where they are used.
Also drop the assignment of the snd_pcm_hardware struct to runtime->hw since
that is what the call to snd_soc_set_runtime_hwparams() right above it already
does, so the second assignment is redundant.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()
Caught by sparse:
sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct pin configs for the Acer AC700. Most importantly indicate
that SPDIF is connected, it routes to HDMI out.
Similar to Aspire models, chain in the DMIC fixup and allow it to be
applied to this codec (ALC269VB) as well.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Channel size settings will be made at the end of
davinci_mcasp_hw_params() routine and thus overwrite frame
format settings made for DIT mode. This patch fixes this issue
by taking op_mode into account. Tested with official PSP 3.2
kernel and sii9022a HDMI transmitter.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AFSX won't be used in DIT mode. The related pins are AHCLKX and
the data pins.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 453807f3 ("ASoC: ep93xx: Use ep93xx_dma_params instead of
ep93xx_pcm_dma_params") introduced a small compile error by not updating the
name of the 'dma_port' field to 'port'. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the common DAI DMA data struct for fsl/imx, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done. Add a new patch op to return that
latency for capture and playback streams. The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count. When it's NULL, nothing changes.
This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6ab317419c.
The commit [6ab317419c: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability. This assumed that this option
is rarely changed dynamically unlike power_save option. Too naive.
It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing. This enabled forcibly the
runtime PM of the controller, which is known to be broken om many
chips thus disabled as default.
So, the only sane fix is to revert this commit again. It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171
Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config
Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).
In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.
If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.
This patch does not introduce any logic flow change.
It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor style fix, following a general code style in the kernel.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge branch 'fix/samsung' of
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into
asoc-component to resolve trivial conflict
Conflicts:
sound/soc/samsung/i2s.c
Use the common DAI DMA data struct for tegra, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the common DAI DMA data struct for omap, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
For omap-dmic and omap-mcpdm also move the DMA data from a global variable to
the driver state struct.
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@opensource.wolfsonmicro.com>
This patch adds a common DMA data struct which can be used by DAI drivers to
communicate their DMA configuration requirements to the DMA pcm driver. Having
a common data structure for this allows us to implement common functions on top
of them, which can be used by multiple platforms.
This patch also introduces a new function to initialize certain fields of a
dma_slave_config struct from the common DAI DMA data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Usually device_fc should be set to false for audio DMAs. Initialize it in a
common place so drivers don't have to do this manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tegra dmaengine driver does not support pausing and resuming a DMA stream.
The tegra PCM driver still claims to support pause and resume though and
implements them by stopping and restarting the stream. This is not what an
application using pause/resume would expect. Usually applications have support
for working around PCMs which do not support suspend and resume, so don't set
the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and
use the default snd_dmaengine_pcm_trigger callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().
No functional change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, a new platform device is created for secondary device
by calling platform_device_register_resndata and then the drvdata
is set for this device.
The following patch has been added to driver core:
"driver core: fix possible missing of device probe".
This results in the added device getting probed immediately but
the drvdata for the secondary device is not yet set.
This patch removes the platform_device_register_resndata call and
instead calls platform_device_alloc, platform_set_drvdata and
platform_device_add which fixes the above issue.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a possible crash in case drvdata for the secondary
device is not set.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.
Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.
The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.
Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.
'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone. Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Kconfig symbol SND_SOC_OF_SIMPLE got removed in commit
f0fba2ad1b ("ASoC: multi-component - ASoC
Multi-Component Support"). But that commit missed one instance. Remove
it now, together with the prompt it has effectively hidden ever since.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When clearing the walked flags there is no need to clear all paths, we
only need to clear the paths we actually walked. This means we can split
dapm_clear_walk() into input and output versions and rather than going
through all DAPM paths we can recurse down the path until we encounter
paths we have not yet walked.
This reduces the number of operations we need to perform and improves
cache locality.
[Pulled out of the vendor tree that the patch was originally generated
for by me, any bugs were introduced in that process -- broonie]
Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We already clear the walked state in dapm_widget_power_check(), no need
to do it again.
Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Merge tag 'v3.9-rc5' into patchwork
Linux 3.9-rc5
* tag 'v3.9-rc5': (1080 commits)
Linux 3.9-rc5
Revert "lockdep: check that no locks held at freeze time"
dw_dmac: adjust slave_id accordingly to request line base
dmaengine: dw_dma: fix endianess for DT xlate function
PNP: List Rafael Wysocki as a maintainer
rbd: don't zero-fill non-image object requests
ia64 idle: delete stale (*idle)() function pointer
Btrfs: don't drop path when printing out tree errors in scrub
target: Fix RESERVATION_CONFLICT status regression for iscsi-target special case
tcm_vhost: Avoid VIRTIO_RING_F_EVENT_IDX feature bit
Revert "mm: introduce VM_POPULATE flag to better deal with racy userspace programs"
usb: ftdi_sio: Add support for Mitsubishi FX-USB-AW/-BD
mg_disk: fix error return code in mg_probe()
Btrfs: fix wrong return value of btrfs_lookup_csum()
Btrfs: fix wrong reservation of csums
Btrfs: fix double free in the btrfs_qgroup_account_ref()
Btrfs: limit the global reserve to 512mb
Btrfs: hold the ordered operations mutex when waiting on ordered extents
Btrfs: fix space accounting for unlink and rename
Btrfs: fix space leak when we fail to reserve metadata space
...
The header <mach/hardware.h> is not needed at all, and <mach/mxs.h> is
needed only for macros MXS_SET_ADDR and MXS_CLR_ADDR. Define the macros
and remove the mach header inclusions.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the equivalent of commit af46800 "ASoC: Implement mux control
sharing", but applied to mixers instead of muxes.
This allows a single control to affect multiple mixer widgets at once,
which is useful when there is a single set of register bits that affects
multiple mixers in HW, for example both the L and R mixers of a stereo
path.
Without this, you either:
1) End up with multiple controls that affect the same register bits, but
whose DAPM state falls out of sync with HW, since the DAPM state is only
updated for the specific control that is modified, and not for other
paths that are affected by the register bit(s).
2) False paths through DAPM, since you end up merging unconnected stereo
paths together into a single widget which hosts the single control, and
then branching back out again, thus conjoining the enable states of the
two input paths.
Now that the kcontrol creation logic is split out into a separate
function, dapm_create_or_share_mixmux_kcontrol(), also use that to
replace most of the body of dapm_new_mux(). This should produce no
functional change, but simply eliminates some mostly duplicated code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
while at it, update the copyright timeline too
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have regular register mapped controls we should be splitting
the control sets for ADSP1 and ADSP2 as the register maps are not
identical. Do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When a new stream is being opened it is necessary to cancel any delayed
power down of the audio.
[Fixed unused variable -- broonie]
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does no not modify the driver of a platform. Making it const
allows ASoC platform drivers to declare the snd_soc_platform_driver struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this, modules will fail to link against those symbols.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
siu_dai.c is using snd_soc_register_dais(),
even though array size of siu_i2s_dai is 1.
OTOH, new API snd_soc_register_component() uses properly
snd_soc_register_dai() (henceforth dai()) or
snd_soc_register_dais() (henceforth dais()) via num_dai.
Then, cpu_dai_name will be "siu-i2s-dai" if dais() was used,
and it will be "siu-pcm-audio" if dai() was used.
Therefore this patch fixup migor_dai :: cpu_dai_name too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_register_dai() uses fmt_single_name(), and
snd_soc_register_dais() uses fmt_multiple_name()
for dai->name which is used for name based matching.
This patch uses properly snd_soc_register_dai() it it was single driver,
and uses snd_register_dais() if it were multiple drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compress core added metadata apis in 9727b4, so add same in ASoC
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into asoc-arizona
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into char-misc-next
Mark writes:
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
Running HPDET while the headphone outputs are enabled can disrupt the
operation of HPDET. In order to avoid this HPDET needs to disable the
headphone outputs and ASoC needs to not enable them while HPDET is
running.
Do the ASoC side of this by storing the enable state in the core driver
structure and only writing to the device if a flag indicating that the
accessory detection side is in a state where it can have the headphone
output stage enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions were initially added to be able to support some oddball dma
drivers, but all users have been updated to deal with the situation without the
help of snd_dmaengine_pcm_{set,get}_data, so these two functions can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the mxs_dma_data struct, which gets passed to the dmaengine driver, is
allocated in the pcm driver's open callback. The mxs_dma_data struct has exactly
one field which is initialized from the the same field in the mxs_pcm_dma_params
struct. The mxs_pcm_dma_params struct gets passed to the pcm driver from the dai
driver. Instead of taking this indirection embed the mxs_dma_data struct
directly in the mxs_pcm_dma_params struct. This allows us to simplify the pcm
driver quite a bit, since we don't have to care about memory managing the
mxs_dma_data struct anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the imx_dma_data struct, which gets passed to the dmaengine driver, is
allocated and constructed in the pcm driver from the data stored in the
dma_params struct. The dma_params struct gets passed to the pcm driver from the
dai driver. Instead of going this route of indirection embed the dma_data struct
directly into the dma_params struct and let the dai driver fill it in. This
allows us to simplify the imx-pcm-dma driver quite a bit, since it doesn't have
care about memory managing the imx_dma_data struct anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma filter parameters are only used within filter callback, so there is no
need to allocate them on the heap and keep them around until the PCM has been
closed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the ep93xx_dma_params struct which is passed to the dmaengine driver
is constructed at runtime from the ep93xx_pcm_dma_params that gets passed to the
ep93xx PCM driver from one of the ep93xx DAI drivers. The ep93xx_pcm_dma_params
struct is almost identical to the ep93xx_dma_params struct. The only missing
field is the 'direction' field, which is computed at runtime in the PCM driver
based on the current substream. Since we know in advance which
ep93xx_pcm_dma_params struct is being used for which substream at compile time,
we also already know which direction to use at compile time. So we can easily
replace all instances of ep93xx_pcm_dma_params with their ep93xx_dma_params
counterpart. This allows us to simplify the code in the ep93xx pcm driver quite
a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We want to get rid of snd_dmaengine_pcm_{set,get}_data(). All instances of
snd_dmaengine_pcm_get_data() in the atmel pcm driver can easily be replaced with
snd_soc_dai_get_dma_data().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver never uses snd_dmaengine_pcm_get_data(), so there is no need to use
snd_dmaengine_pcm_set_data().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver never uses snd_dmaengine_pcm_get_data(), so there is no need to use
snd_dmaengine_pcm_set_data().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Stephen Warren <swarren@wwwdotorg.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
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@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The generic dmaengine based PCM driver code takes care of setting this
constraint, there is no need of doing it manually in the ux500 driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Used PTR_RET function instead of IS_ERR and PTR_ERR.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The omap PCM driver provides a set_threshold callback which gets called by the
PCM driver when either playback or capture is started. The only DAI driver which
sets this callback is the mcbsp driver. This patch removes the callback from the
PCM driver and moves the invocation of the omap_mcbsp_set_threshold() function
to the mcbsp hw_params callback since this is the only place where the threshold
size can change. Doing so allows us to use the default dmaengine PCM trigger
callback in the omap PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is
success. So just check return value is negative.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Tegra114 requires different PLL rates. Modify the code to know about
this.
On Tegra114 only for now, use regular clk_get() rather than clk_get_sys()
to retrieve clocks. This assumes that the clocks will be represented in
device tree. We can assure that from the start of any Tegra114 audio
support. For older chips, I'll add the required clocks properties to the
device trees this kernel cycle, and switch this code to only support the
"new_clocks" path next cycle.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tegra114's AHUB shares a design with Tegra30, with the followin changes:
* Supports more (10 vs. 4) bi-directional FIFO channels into RAM.
* Requires a separate block of registers to support the above.
* Supports more attached clients, i.e. new audio multiplexing and
de-multiplexing modules.
* Is affected by more clocks due to the above.
This change fully defines the device tree binding changes required to
represent these changes, and minimally extends the driver to support
the new hardware, without exposing any of the new FIFO channels.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since imx-sgtl5000 is only used on DT platform, it makes more sense to
use cpu_of_node rather than cpu_dai_name to match cpu dai.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ioctl is defined as IOW, so pass the argument as const.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Acked-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Alexey Klimov <klimov.linux@gmail.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
This ioctl is defined as IOW, so pass the argument as const.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs. Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.
Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
The dma-sh7760 currently fails with the following compile error:
sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer
sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer
sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast
sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer
sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe':
sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type
include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *'
This is due the misnaming of the snd_soc_platform_driver type name and 'ops'
field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC
Multi-Component Support").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Use the newly introduced mute_stream DAI operation, and don't mute the
codec if it's called for the _CAPTURE stream.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC does not yet really have the framework features needed to
support propagating sample rates through the device well yet implement
basic support for the ISRCs equivalent to that we currently have for the
ASRCs. The user can opt for 8kHz or 16kHz as the rate for the DSP blocks
in addition to the main audio rate, these being the primary use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the device state does not diverge from the state we have set
in the register map in order to make the behaviour clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some system designs have been identified which repurpose portions of the
speaker driver circuits for other functions which will require that they
not be managed using DAPM. Prepare for this by factoring out the creation
of the speaker widgets into the core driver, the widgets will be replaced
by dummy ones when the additional functions are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems use the audio CODEC to clock a DAI with multiple data lines
in parallel, meaning that bit clocks are only required for a smaller number
of channels than data is sent for. In some cases providing the extra bit
clocks can take the other devices on the audio bus out of spec.
Support such systems by allowing a maximum number of channels to be
specified.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lars-Peter Clausen
Mark Brown
Dan Carpenter
Stephen Warren
Paul Bolle
Shawn Guo
Sachin Kamat
Fabio Baltieri
Uwe Kleine-König
Marek Belisko
Fabio Estevam
Michał Mirosław
Dimitris Papastamos
Linus Torvalds
Takashi Iwai
Axel Lin
Kent Overstreet
Wei Yongjun
Wang YanQing
Mike Travis
David Howells
Mauro Carvalho Chehab
Eldad Zack
Al Viro
Vinod Koul
David Henningsson
Clemens Ladisch
Olof Johansson
Arnd Bergmann
Kailang Yang
Trulan Martin
Daniel Mack
Pavel Machek
Michal Bachraty
Daniel Schürmann
Charles Keepax
Dylan Reid
Stas Sergeev
Markus Pargmann
Greg Kroah-Hartman
Calvin Owens
Masanari Iida
Heiko Stübner
Alban Bedel
Linus Walleij
Yegor Yefremov
Jiri Slaby
Rainer Koenig
Prathyush K
Torstein Hegge
Mengdong Lin
Chih-Chung Chang
Alexandru Gheorghiu
Ryo Tsutsui
Kuninori Morimoto
Jeeja KP
Joonyoung Shim
Hans Verkuil