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The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.
Also drop the regcache_cache_only() calls from the suspend and resume
handlers. There shouldn't be any IO happening after suspend and before
resume.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.17
A few more driver specific fixes on top of the currently pending fixes
(which are already in your tree but not Linus').
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The DACs on Sigmatel/IDT codecs do mute at the lowest volume level,
and in the earlier drivers, we passed TLV_DB_SCALE_MUTE bit for each
volume control element like Speaker and Headphone as well as Master.
Along with the translation to the generic parser, however, the TLV bit
was lost for the slave controls (e.g. Speaker) but set only to
Master. In theory this should have sufficed, but apps, particularly
PA, do care the slave volume bits, so we seem to see a regression in
the volume controls.
This patch adds a flag to hda_gen_spec to specify the DAC mute
feature, and adds the TLV bit properly for all relevant volume
controls. Also, the TLV bit for vmaster is set in hda_generic.c, so
that we can get rid of all tricks from the codec driver side.
As the similar hack is applied to Conexant 5051 stuff, we can get rid
of it as well.
BugLink: https://bugs.launchpad.net/bugs/1357928
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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All MAX98090 input clocks MCLK, LRCLK and BCLK must be running and stable
before powering on the codec in slave mode. Otherwise the PLL may not lock
to LRCLK causing silence in playback and capture. How often that happens is
somewhat hardware and clock configuration specific.
Now if wanting to follow strictly this clocks must be active before
powering the codec on requirement we should have a notification from DAI
driver to codec driver when clocks are activated and take codec out of
shutdown only after that. Plus take care of possible active bypass paths.
However, when PLL unlock occurs, MAX98090 asserts the PLL Unlock Flag which
can be configured as an IRQ source. This allows to workaround around the
issue by toggling the codec power shortly in case of PLL lock failure.
In order to prevent needlessly toggling codec power in case of short PLL
unlocks at the beginning of stream this patch implements delayed activation
for PLL unlock interrupt. Then workaround is run only when the PLL doesn't
lock at all.
Power toggling workaround for PLL unlock comes originally from
Liam Girdwood <liam.r.girdwood@linux.intel.com> and delayed activation from
me.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This simplifies aic31xx_divs table. There is no more need for p_val or
separate lines for 12 and 24 MHz mclks.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-tlv320aic31xx
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I2S format requires bitclock to have an exact amount of cycles in a
frame for audio to work cleanly. With dsp formats that is not so
important.
Updates aic31xx_setup_pll() to look for a line in aic31xx_divs table
that produces the best match for the bitclock and adds lines to
aic31xx_divs for 12MHz mclk and 24bit samples.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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f687d900d30a61dda38db2a99239f5284a86a309
(ASoC: simple-card: cpu_dai_name creates confusion when DT case)
cleared cpu_dai_name for caring fmt_single_name case,
and
179949bc04c7157a4b2279f62a842638b61f78f9
(ASoC: simple-card: remove dai_link->cpu_dai_name when DT)
cared multi dai-link case.
but, cpu_dai_name matching is required when fmt_multiple_name was used
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The previous commit for the non-atomic PCM ops added more codes to
snd_pcm_stream_lock() and its variants. Since they are inlined
functions, it resulted in a significant code size bloat. For reducing
the size bloat, this patch changes the inline functions to the normal
function calls. The export of rwlock and rwsem are removed as well,
since they are referred only in pcm_native.c now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently, many PCM operations are performed in a critical section
protected by spinlock, typically the trigger and pointer callbacks are
assumed to be atomic. This is basically because some trigger action
(e.g. PCM stop after drain or xrun) is done in the interrupt handler.
If a driver runs in a threaded irq, however, this doesn't have to be
atomic. And many devices want to handle trigger in a non-atomic
context due to lengthy communications.
This patch tries all PCM calls operational in non-atomic context.
What it does is very simple: replaces the substream spinlock with the
corresponding substream mutex when pcm->nonatomic flag is set. The
driver that wants to use the non-atomic PCM ops just needs to set the
flag and keep the rest as is. (Of course, it must not handle any PCM
ops in irq context.)
Note that the code doesn't check whether it's atomic-safe or not, but
trust in 100% that the driver sets pcm->nonatomic correctly.
One possible problem is the case where linked PCM substreams have
inconsistent nonatomic states. For avoiding this, snd_pcm_link()
returns an error if one tries to link an inconsistent PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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...by factoring out common parts to the just added pin macros.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This will be used in a later patch to make the pin quirk table shorter.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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'asoc/fix/da732x', 'asoc/fix/omap', 'asoc/fix/rsnd', 'asoc/fix/rt5640', 'asoc/fix/rt5677', 'asoc/fix/simple' and 'asoc/fix/tegra' into asoc-linus
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ALC1150 codec seems to need the COEF- and PLL-setups just like its
compatible ALC882 codec. Some machines (e.g. SunMicro X10SAT) show
the problem like too low output volumes unless the COEF setup is
applied.
Reported-and-tested-by: Dana Goyette <danagoyette@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit ff795d614bfa ("ASoC: ab8500: Convert register I/O to regmap")
initially converted the ab8500 CODEC driver to use regmap rather than
legacy ASoC IO. This was reverted though in commit 63e6d43bf80d ("ASoC:
ab8500: Revert to using custom I/O functions") since the inital conversion
was not working properly. This was presumebly because the SOC_SINGLE_XR_SX
controls, which are used by this driver, did not properly support regmap at
that point. This has since been fixed in commit 6137a5ca326d ("ASoC: Prepare
SOC_SINGLE_XR_SX controls for regmap"). So revert back to regmap again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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DAI links's cpu_of_node's and codec_of_node's refcounts shouldn't
be decremented immediately at the end of the probe() fucntion.
Because we will still use them before the audio card is removed.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The 'big-endian-data' property is originally used to indicate whether the
LSB firstly or MSB firstly will be transmitted to the CODEC or received
from the CODEC, and there has nothing relation to the memory data.
Generally, if the audio data in big endian format, which will be using the
bytes reversion, Here this can only be used to bits reversion.
So using the 'lsb-first' instead of 'big-endian-data' can make the code
to be readable easier and more easy to understand what this property is
used to do.
This property used for configuring whether the LSB or the MSB is transmitted
first for the fifo data.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-sai
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Acer Aspire 3830TG with CX20588 codec has a digital built-in mic that
has the same problem like many others, the inverted signal in stereo.
Apply the same fixup to this machine, too.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix to return a negative error code from the 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@kernel.org>
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Rename iface_reg3 to iface_reg2 since this variable is actually used for
interface register 2.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-esai
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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6a91a17bd7b92b2d2aa9ece85457f52a62fd7708
(ASoC: simple-card: Handle many DAI links)
added multi DAI support on simple-card.
This means priv->dai_link might be pointer of multi DAI.
dai_link->init is needed for all DAI.
This patch cares it for all DAIs on DT/non-DT
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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f687d900d30a61dda38db2a99239f5284a86a309
(ASoC: simple-card: cpu_dai_name creates confusion when DT case)
removed dai_link->cpu_dai_name when DT case,
since it uses DT phand in soc_bind_dai_link().
This binding will fail if it has cpu_dai_name.
6a91a17bd7b92b2d2aa9ece85457f52a62fd7708
(ASoC: simple-card: Handle many DAI links)
added multi DAI link support to simple-card driver.
Then, removing cpu_dai_name was cared only single DAI.
But, it is needed in all DT cases.
This patch moves it to asoc_simple_card_dai_link_of()
so that care about all DAIs.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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simple-card driver is using asoc_simple_xxx() prefix.
simple_card_dai_link_of() should be
asoc_simple_card_dai_link_of().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The platform_name should be omap-mcasp3 for the 2nd link which is used for
voice connection.
Reported-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie+linaro@kernel.org>
Cc: stable@vger.kernel.org
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Dice quirk
In IEC 61883-6, one data block transfers one event. In ALSA, the event equals one PCM frame,
hence one data block transfers one PCM frame. But Dice has a quirk at higher sampling rate
(176.4/192.0 kHz) that one data block transfers two PCM frames.
Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete
CIP_HI_DUALWIRE") moved some codes related to this quirk into Dice driver. But the commit
forgot to add arrangements for PCM period interrupts and DMA pointer updates. As a result, Dice
driver cannot work correctly at higher sampling rate.
This commit adds 'double_pcm_frames' parameter to amdtp structure for this quirk. When this
parameter is set, PCM period interrupts and DMA pointer updates occur at double speed than in
IEC 61883-6.
Reported-by: Daniel Robbins <drobbins@funtoo.org>
Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.16
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The channel mapping is initialized by amdtp_stream_set_parameters(), however
Dice driver set it before calling this function. Furthermore, the setting is
wrong because the index is the value of array, and vice versa.
This commit moves codes for channel mapping after the function and set it correctly.
Reported-by: Daniel Robbins <drobbins@funtoo.org>
Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.16
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use single snd_soc_update_bits() call to update the register bits.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The regulator_bulk_free() call is not required because current code is using
devm_regulator_bulk_get().
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The regulator_bulk_free() call is not required because current code is using
devm_regulator_bulk_get().
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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There are no real users for cache_only in "struct snd_soc_codec" so remove
it and needless debugfs node.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Return true or false instead of 1 and 0
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Return true or false instead of 1 and 0
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Return true or false instead of 1 and 0
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Reported-by: Zoltán Szenczi <zoltan@raspberrypi.org>
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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Reported-by: Zoltán Szenczi <zoltan@raspberrypi.org>
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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Reported-by: Zoltán Szenczi <zoltan@raspberrypi.org>
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
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 fixes from Takashi Iwai:
"Here contains not many exciting changes but just a few minor ones: An
off-by-one proc write fix, a couple of trivial incldue guard fixes,
Acer laptop pinconfig fix, and a fix for DSD formats that are still
rarely used"
* tag 'sound-3.17-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Set up initial pins for Acer Aspire V5
ALSA: pcm: Fix the silence data for DSD formats
ALSA: ctxfi: ct20k1reg: Fix typo in include guard
ALSA: hda: ca0132_regs.h: Fix typo in include guard
ALSA: core: fix buffer overflow in snd_info_get_line()
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When using a DT-based multi-platform kernel, there's not always Kconfig
logic that selects the right codec driver.
Allow the user to manually select WM8978.
This is needed for Armadillo 800 EVA using a generic r8a7740 multi-platform
kernel.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
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