| Commit message (Collapse) | Author | Age | Files | Lines |
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Convert all uses of devm_request_and_ioremap() to the newly introduced
devm_ioremap_resource() which provides more consistent error handling.
devm_ioremap_resource() provides its own error messages so all explicit
error messages can be removed from the failure code paths.
Signed-off-by: Thierry Reding <thierry.reding@avionic-design.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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These are not supported
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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According to the defines in wm2200.h:
/*
* R1284 (0x504) - Audio IF 1_5
*/
We should not left shift 1 bit for fmt_val when setting dai format.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Otherwise we won't run correctly on systems that require this for larger
data transfers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Both the mask and mode settings are wrong in current code.
According to the datasheet:
S2PCFG0 (0x0A)
BIT[3:1] DATA_FORMAT
serial interface protocol format:
000: left Justified
001: I2S (default)
010: right justified
100: PCM no delay
101: PCM delay
111: DSP
Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and
RIGHT_J_DATA_FORMAT.
Also adds define for DATA_FORMAT_MSK.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3.
From Eric Nelson:
"We also found that for the microphones we have here (commodity PC boom mics) a
default value of 2 for the gain gives the best results."
So change the default microphone gain as well.
Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Update lm49453_reg_defs values as per LM49453 HW revision-B
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.
Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>
--
sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++-------------------
1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The mode variable is either 0 or 1.
To update mode setting, the mask should be BIT(0) rather than BIT(1).
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In the case of no-match, return -EINVAL instead of 0.
Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The driver had the property check for 'cirrus,amutec_eq_bmutec' the
wrong way around. That happens if you misspell the property in the
bindings during tests.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When probing aux_dev, initializing is as below:
device_initialize()
device_add()
So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.
So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced
the following build warning:
sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable]
Remove the unused 'ret' variable.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.
Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This update contains overall only driver-specific fixes. Slightly
large LOC are seen in usb-audio driver for a couple of new device
quirks and cs42l71 ASoC driver for enhanced features. The others are
a few small (regression) fixes HD-audio, and yet other small / trival
ASoC fixes."
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
ALSA: HDA: Fix sound resume hang
ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
ASoC: atmel-ssc: change disable to disable in dts node
ASoC: Prevent pop_wait overwrite
ALSA: usb-audio: ignore-quirk for HP Wireless Audio
ALSA: hda - Always turn on pins for HDMI/DP
ALSA: hda - Fix pin configuration of HP Pavilion dv7
ASoC: core: Fix splitting of log messages
ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
ASoC: cs42l73: Add DAPM events for power down.
ASoC: cs42l73: Add DMIC's as DAPM inputs.
ASoC: sigmadsp: Fix endianness conversion issue
ASoC: tpa6130a2: Use devm_* APIs
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Converted to use devm_gpio_request and devm_regulator_get APIs.
These are device managed and make error handling and cleanup
a bit simpler.
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Don't wrap log messages over multiple lines, it makes them hard to grep
for.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Since VSP only has one power bit, only provide one DAPM widget.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The 'addr' field of the sigma_action struct is stored as big endian in the
firmware file.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This update contains a fairly wide range of changes all over in sound
subdirectory, mainly because of UAPI header moves by David and __dev*
annotation removals by Bill. Other highlights are:
- Introduced the support for wallclock timestamps in ALSA PCM core
- Add the poll loop implementation for HD-audio jack detection
- Yet more VGA-switcheroo fixes for HD-audio
- New VIA HD-audio codec support
- More fixes on resource management in USB audio and MIDI drivers
- More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite,
Roland VG-99, etc
- Add support for FastTrack C400 usb-audio
- Clean ups in many drivers regarding firmware loading
- Add PSC724 Ultiimate Edge support to ice1712
- A few hdspm driver updates
- New Stanton SCS.1d/1m FireWire driver
- Standardisation of the logging in ASoC codes
- DT and dmaengine support for ASoC Atmel
- Support for Wolfson ADSP cores
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090
- Lots of other ASoC driver fixes and developments"
Fix up trivial conflicts. And go out on a limb and assume the dts file
'status' field of one of the conflicting things was supposed to be
"disabled", not "disable" like in pretty much all other cases.
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits)
ALSA: hda - Move runtime PM check to runtime_idle callback
ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522
ALSA: hda - Avoid doubly suspend after vga switcheroo
ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3
ALSA: hda - Check validity of CORB/RIRB WP reads
ALSA: hda - use usleep_range in link reset and change timeout check
ALSA: HDA: VIA: Add support for codec VT1808.
ALSA: HDA: VIA Add support for codec VT1705CF.
ASoC: codecs: remove __dev* attributes
ASoC: utils: remove __dev* attributes
ASoC: ux500: remove __dev* attributes
ASoC: txx9: remove __dev* attributes
ASoC: tegra: remove __dev* attributes
ASoC: spear: remove __dev* attributes
ASoC: sh: remove __dev* attributes
ASoC: s6000: remove __dev* attributes
ASoC: OMAP: remove __dev* attributes
ASoC: nuc900: remove __dev* attributes
ASoC: mxs: remove __dev* attributes
ASoC: kirkwood: remove __dev* attributes
...
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CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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