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commit 4495e46fe18f198366961bb2b324a694ef8a9b44 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 98682063549bedd6e2d2b6b7222f150c6fbce68c upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f1ca493b0b5e8f42d3b2dc8877860db2983f47b6 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 417a1178f1bf3cdc606376b3ded3a22489fbb3eb upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0d2b6422529a26ac4dee06196524ba9da70cf735 upstream.
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a80cc734282805e15b5e023751a4d02f7ffbcc91 upstream.
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 5f960294e2031d12f10c8488c3446fecbf59628d upstream.
These are not supported
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0cc411b934c4317b363d1af993549f391852b980 upstream.
These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2a5f431592343b78896013b055582f94c12a5049 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 267f8fa2e1eef0612b2007e1f1846bcbc35cc1fa upstream.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a3adb1432d7a3ad86bb17a1638e44414537e4118 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 961a7aeafab477f63d9eef26afde9cbb8badcd0f upstream.
soc-dmaengine-pcm library need to be part of the snd-soc-core in order to
be able to compile ASoC as modules when dmaengine is enabled on the platform.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 445632ad6dda42f4d3f9df2569a852ca0d4ea608 upstream.
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 55c6f4cb6ef49afbb86222c6a3ff85329199c729 upstream.
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 034940a6b3afbe79022ab6922dd9d2982b78e6d5 upstream.
It is not Vinrator but Vibrator.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a1b98e12b7f8fad2f0aa3c08a3302bcac7ae1ec7 upstream.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 5ae9eb4cbdfd640269dbd66aa3c92ea8e11cc838 upstream.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 57451e437796548d658d03c2c4aab659eafcd799 upstream.
shdma doesn't support transfer re-scheduling or triggering from callbacks
or from atomic context. The fsi driver issues DMA transfers from a tasklet
context, which is a bug. To fix it convert tasklet to a work.
Reported-by: Do Q.Thang <dq-thang@jinso.co.jp>
Tested-by: Do Q.Thang <dq-thang@jinso.co.jp>
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 689185b78ba6fbe0042f662a468b5565909dff7a upstream.
Help UIs associate it with the matching gain control.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40 upstream.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 57b2d68863f281737d8596cb3d76d89d9cc54fd8 upstream.
The pause and resume operations indicate that the stream can be
un-paused/resumed from the exact location they were paused/suspended.
This is not true for this driver, the pause and suspend triggers share
the same code path with stop, they flush all pending DMA transfers.
This drops all pending samples. The pause_release/resume triggers are
the same as start, except that prepare won't be called beforehand,
nothing will be enqueued to the DMA engine and nothing will happen (no
audio). Removing the pause flag will let apps know that it isn't
supported. Removing the resume flag will cause user space to call
prepare and start instead of resume, so audio will continue playing when
the system wakes up.
Before removing the pause and resume flags, I tested this on an exynos
5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a
write error. Suspend/resume testing led to the same result. Removing
the two flags fixes suspend/resume (since snd_pcm_prepare is called
again). And leads to a proper reporting of pause not supported.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c921928661eda599d73a6a86e58bdd5aecfa18cb upstream.
Both the schematics and practical testing show that the HP detect GPIO
is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio
should not specify to invert the signal.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d0db84e713eaaccea2a435e1625fb3150b335f4a upstream.
The check for the mux_signal callback was wrong which prevents us to
configure the 6pin port's FSR/CLKR signal mux.
Reported-by: CF Adad <cfadad@rocketmail.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ccf795847a38235ee4a56a24129ce75147d6ba8f upstream.
Currently the microphone input source is not selectable as while there is
a DAPM widget it's not connected to anything so it won't be properly
instantiated. Add something more correct for the input structure to get
things going, even though it's not hooked into the rest of the routing
map and so won't actually achieve anything except allowing the relevant
register bits to be written.
Reported-by: Christop Fritz <chf.fritz@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit b8edf3e5522735c8ce78b81845f7a1a2d4a08626 upstream.
Otherwise if someone tries to use all four channels on AIF1 with the
device in master mode we won't be able to clock out all the data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 9d40e5582c9c4cfb6977ba2a0ca9c2ed82c56f21 upstream.
Required for reliable power up from cold.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit e4dd76788c7e5b27165890d712c8c4f6f0abd645 upstream.
Ensure robust startup of the part by going through the reset procedure
prior to resyncing the full register cache, avoiding potential intermittent
faults in some designs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0ff97ebf0804d2e519d578fcb4db03f104d2ca8c upstream.
Ever since the DAPM performance improvements we've been marking all widgets
as not dirty after each DAPM run. Since _PRE and _POST events aren't part
of the DAPM graph this has rendered them non-functional, they will never be
marked dirty again and thus will never be run again.
Fix this by skipping them when marking widgets as not dirty.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c9fe573a6584034670c1a55ee8162d623519cbbf upstream.
In sound/soc/codecs/tlv320aic3x.c
data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
data | (pll_p << PLLP_SHIFT));
In the above code, pll-p value is OR'ed with previous value without
clearing it. Bug is not seen if pll-p value doesn't change across
Sampling frequency.
However on some platforms (like AM335x EVM-SK), pll-p may have different
values across different sampling frequencies. In such case, above code
configures the pll with a wrong value.
Because of this bug, when a audio stream is played with pll value
different from previous stream, audio is heard as differently(like its
stretched).
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit b0dfa4541e48ac4cc5f017285432c89923ad0f58 upstream.
Without this very high BCLKs will be configured incorrectly.
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 433897f7408b556f7dfbb98c94deea02e634d2a7 upstream.
We no longer have a flat ASoC cache so can't peer directly into the array
any more but should instead use the register I/O functions to update the
cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit bfd37bb5f681961e255fd2f346c20fdae2ef3f27 upstream.
Volume updates may not be acted upon if there is no clock applied when
the volume update is written. Ensure this doesn't happen by writing out
registers with volume updates after we enable each of the clocks.
There are more registers updated than before as previously we were
relying on wm_hubs to set those for controls it manages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c8fdc1b56611faa7b38eab6b99da5e20113661ff upstream.
Ensure that all the actions get taken at appropriate times by calling the
_PRE and _POST events for the aifNclk_ev functions explicitly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few last-minute regression fixes for 3.4 final kernel. All trivial,
and Cc'ed to stable kernel."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Fix AIF2ADC power down
ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops
ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute fixes
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.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:
"Slightly more than expected as rc7, but all are reasonablly small
fixes. A few additions of HD-audio fixup entries, a couple of other
regression fixes including a revert, and a few other trivial
oneliners."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: sh: fix migor.c compilation
ALSA: HDA: Lessen CPU usage when waiting for chip to respond
Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
ALSA: hdsp - Provide ioctl_compat
ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
ALSA: echoaudio: Remove incorrect part of assertion
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Build fix for SH in 3.4
An API update which wasn't sufficiently thorough in updating the tree...
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Fix a recent compilation breakage, caused by a change in SH clock API.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
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git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into fix/asoc
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Signed-off-by: Oleg Matcovschi <oleg.matcovschi@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
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As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised
s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai.
Without this call the snd_soc_dai_ops structure isn't initialised correctly.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.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:
"A workaround for an ASUS laptop and a few ASoC changes; most of the
commits are tagged for stable, too."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Improve sequencing of AIF channel enables
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
ASoC: core: Fix card RTD count for deferred probe.
ASoC: cs42l73: don't use negative array index
ASoC: dapm: Ensure power gets managed for line widgets
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Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC
machine drivers to use the dai_links dai_fmt field to setup their DAI format.
For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt,
but missed to set the dai_links dai_fmt field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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