| Commit message (Collapse) | Author | Age |
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This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
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This makes it possible to hook the device into a more complex board and
ensures it will continue to work with non-DAPM support removed from the
core.
Signed-off-by: Mark Brown <broonie@linaro.org>
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Support for loading the ak4554 codec module via devicetree.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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ak4554 is very simple DA/AD converter which has no setting register.
Note that it has hard coded asymmetric data format
playback : SND_SOC_DAIFMT_RIGHT_J
capture : SND_SOC_DAIFMT_LEFT_J
This driver has single DAI and doesn't have set_fmt.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Saves some code. We should also be able to manage the power up and reset
registers using DAPM but it's probably more trouble than it's worth in
mains powered systems.
Signed-off-by: Mark Brown <broonie@linaro.org>
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This makes it easer to integrate the device with other on-board components
and ensures correct operation following removal of support for non-DAPM
CODECs.
Signed-off-by: Mark Brown <broonie@linaro.org>
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No need to hold on to the `codec' pointer. We can use the `dsp'
pointer and grab all the information we need from there. This
makes the parameters for the functions a bit more sane and idiomatic.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Get rid off the wm_coeff struct and the wm_coeff_add_kcontrol()
function. We are now using the snd_soc_card_kcontrol() function to
get the kcontrol pointers. No need to call into ALSA code to
register the kcontrols.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This makes it easier to hook into boards and ensures the driver continues
to work with support for non-DAPM CODECs removed.
Signed-off-by: Mark Brown <broonie@linaro.org>
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This driver can be built as module, thus add module device table for adav801 to
support module auto loading.
To make the naming consistent, also rename adav80x_id to adav80x_i2c_id.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The GPIO manipulation done by this driver is never in atomic context so
we can use gpio_set_value_cansleep() and support GPIOs that can't be set
from atomic context.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and
not an unsigned long. The current tests are always false.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Both the adau1702 and the adau1401(a) are register compatible to the adau1701,
so add them to adau1701 driver's device id table.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9
("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due
to the dependency on the SigmaDSP firmware loader which was only available on a
limited set of platforms. This was fixed quite some time ago in commit 40216ce7
("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver
back again to SND_SOC_ALL_CODECS.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This makes it possible to hook up other devices in boards and is required
by removal of support for non-DAPM CODECs in the core.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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Since non-DAPM devices are not going to be supported provide DAPM input
and output widgets and hook them up to the DAIs.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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Ensure continued operation with DAPM being mandatory.
Signed-off-by: Mark Brown <broonie@linaro.org>
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Bit 9 of PLL2,3 and 4 is reserved as '0'. The 24bit fractional part
should be split across each register in 8bit chunks.
Signed-off-by: Mike Dyer <mike.dyer@md-soft.co.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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The SGTL5000 Capture Attenuate Switch (or "ADC Volume Range Reduction"
as it is called in the manual) is single bit only.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Reviewed-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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When a sound capture/playback is terminated while a playback/capture
is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in
the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other
channel.
Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active
to prevent this.
Signed-off-by: Lothar Waßmann <LW@KARO-electronics.de>
Reviewed-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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CS42L52 Beep control uses 2dB scale from -56dB
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Beep Volume Min/Max was backwards.
Change to SOC_SONGLE_SX_TLV for correct volume representation
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@kernel.org
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If kzalloc() fails for `img' then we are going to leak the memory
for `out'. We are freeing the memory of all the tx/rx transfers
but the tx/rx buf pointers will be NULL if we drop out earlier.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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It's not always the case that clock is already available when sgtl5000
get probed at the first time, e.g. the clock is provided by CPU DAI
which may be probed after sgtl5000. So let's defer the probe when
devm_clk_get() call fails and give it chance to try later.
It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000:
Let the codec acquire its clock).
[ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2
[ 1.934280] sgtl5000: probe of 0-000a failed with error -2
[ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered
[ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517)
[ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The registers of max98088 are 8 bits, not 16 bits. This bug causes the
contents of registers to be overwritten with bad values when the codec
is suspended and then resumed.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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This was committed in error.
Signed-off-by: Mark Brown <broonie@linaro.org>
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wm8978 needs .symmetric_rates = 1.
The playback/capture will be strange without this patch
when it used asymmetric rate in same time
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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SGTL5000_PLL_FRAC_DIV_MASK is used to mask bits 0-10 (11 bits in total) of
register CHIP_PLL_CTRL, so fix the mask to accomodate all this bit range.
Reported-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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According to the sgtl5000 reference manual, the default value of CHIP_SSS_CTRL
is 0x10.
Reported-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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The usage of strict_strto*() is not preferred, because
strict_strto*() is obsolete. Thus, kstrto*() should be
used.
Signed-off-by: Jingoo Han <jg1.han@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few small fixes (and cleanups) for HD-audio, USB-audio and ASoC"
* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: fix regression for fixed stream quirk
ALSA: hda - Keep halting ALC5505 DSP
ASoC: wm8962: fix NULL pdata pointer
ASoC: imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
ASoC: Samsung: Remove redundant comment
ALSA: hda - Fix EAPD vmaster hook for AD1884 & co
ASoC: samsung: Remove obsolete GPIO based DT pinmuxing
ASoC: mxs: register saif mclk to clock framework
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.11
A few small fixes, all driver specific. The removal of the GPIO based
pinmuxing is a bug fix, since the obsolete nodes had been removed from
the DT it stopped the driver loading.
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There is an error in merge commit 384b834 on conflict resolution which
causes the following NULL pdata pointer bug.
wm8962 0-001a: customer id 0 revision D
Unable to handle kernel NULL pointer dereference at virtual address 00000004
pgd = 80004000
[00000004] *pgd=00000000
Internal error: Oops: 5 [#1] SMP ARM
Modules linked in:
CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0+ #1
task: bf870000 ti: bf874000 task.ti: bf874000
PC is at wm8962_probe+0x134/0x6c8
LR is at regmap_unlock_mutex+0x10/0x14
pc : [<80452100>] lr : [<80304cf4>] psr: a0000113
sp : bf875c98 ip : 00000000 fp : bf875cd4
r10: 00000000 r9 : bfb1830c r8 : 80779bc4
r7 : 00000000 r6 : 00000001 r5 : bfbac010 r4 : bfb33e00
r3 : 80304ce4 r2 : 00000000 r1 : 00000001 r0 : fffffffb
Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel
Control: 10c53c7d Table: 1000404a DAC: 00000017
Process swapper/0 (pid: 1, stack limit = 0xbf874238)
Stack: (0xbf875c98 to 0xbf876000)
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Fix the error by assigning pdata a correct pointer.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"A relative calm release at this time with a flat diffstat. The only
significant change in the ALSA core side is the support for more than
32 card instances, configurable via kconfig. Other than that, in both
ASoC and other parts, mostly some improvements and fixes on the driver
side.
- hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
- hda: Haswell HDMI audio fixes, runtime PM improvements
- hda: Intel BayTrail support, ALC5505 DSP support
- es1968: MediaForte M56VAP support
- usb-audio: Improved support for Yamaha/Roland devices
- usb-audio: M2Tech hiFace, Audio Advantage Micro II support
- hdspm: wordclock fixes
- ASoC: Pending fixes for WM8962
- ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
- ASoC: Generalisation of the Bluetooth and HDMI stub drivers
- ASoC: SSM2518 and RT5640 codec drivers.
- ASoC: Tegra CPUs with RT5640 machine driver
- ASoC: AC'97 refactoring bug fixes
- ASoC: ADAU1701 driver fixes
- Clean up of *_set_drvdata() in a wide range of drivers"
* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
ALSA: vmaster: Fix the regression of missing vmaster hook call
ALSA: hda - Add Dell SSID to support Headset Mic recording
ASoC: adau1701: remove control_data assignment
ASoC: adau1701: more direct regmap usage
ASoC: ac97: fixup multi-platform AC'97 module build failure
ASoC: pxa2xx: fixup multi-platform AC'97 build failures
ASoC: tegra20-ac97: Remove unused variable
ASoC: tegra20-ac97: Remove duplicate error message
ALSA: usb-audio: Add Audio Advantage Micro II
ASoC: tas5086: fix Mid-Z implementation
ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
ALSA: Replace the magic number 44 with const
ALSA: hda - Fix the max length of control name in generic parser
ALSA: hda - Guess what, it's two more Dell headset mic quirks
ALSA: hda - Yet another Dell headset mic quirk
ALSA: hda - Add support for ALC5505 DSP power-save mode
ASoC: mfld: Remove unused variable
ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
ALSA: usb-audio: claim autodetected PCM interfaces all at once
ALSA: usb-audio: remove superfluous Roland quirks
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