The dapm field of the snd_soc_codec struct will eventually be removed
(replaced with the DAPM context from the component embedded inside the
CODEC). Replace its usage with the card's DAPM context. The idea is that
DAPM is hierarchical and with the card at the root it is possible to access
widgets from other contexts through the card context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct will eventually be removed
(replaced with the DAPM context from the component embedded inside the
CODEC). Replace its usage with the card's DAPM context. The idea is that
DAPM is hierarchical and with the card at the root it is possible to access
widgets from other contexts through the card context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently both the oscillator and the PLL are powered up in
set_bias_level. This can be problematic when using output clocks from
the wm8804 for other devices. The snd_soc_codec_set_pll API defines that
a clock should be available once the call returns, however, with all the
clocking controlled in set_bias_level this is not currently the case.
This patch enables pm_runtime for the wm8804, enabling both the
regulators and the oscillator when the chip resumes, and enabling the
PLL in the snd_soc_codec_set_pll call. Naturally the enabling the PLL
will also cause the chip to resume.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This change converts the driver to use DAPM to control the power for the
various blocks on the chip. As part of this change the existing controls
"TX Playback Switch" (controlled power for the SPDIF TX block) and "AIF
Playback Switch" (controlled power for the AIF block) are both removed,
as they are now redundant since the power state of those blocks is
controlled automatically by DAPM.
There are several benefits of this change, the most important of which
is this change adds support for powering down the SPDIF RX block. The RX
block will automatically assume control of the PLL on the chip when it
is receiving a signal, so leaving this enabled all the time as was
currently done in the driver can be problematic. An incoming SPDIF signal
that is not being used can completely destroy the clocking for an in use
TX signal. But this change ensures that the RX block will only be
powered when the user intends to be receiving data, thus avoiding this
issue.
Additional benefits include the chip being simpler to operate as the
power no longer needs to be manually controlled between use-cases and a
small power saving (although it is acknowledged that this is likely
unimportant in the typical use-cases for this chip).
Signed-off-by: Sapthagiri Baratam <sapthagiri.baratam@incubesol.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To fix pop noise when shutdown,the pop noise during shutdown
is the pmic cutoff power of codec without any notice.
Signed-off-by: jay.xu <xjq@rock-chips.com>
Signed-off-by: zhengxing <zhengxing@rock-chips.com>
Signed-off-by: Caesar Wang <wxt@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
0day robot reported a buffer overflow issue:
...
sound/soc/intel/haswell/sst-haswell-pcm.c:1107 hsw_pcm_probe() error: buffer\
overflow 'hsw_dais' 4 <= 4
sound/soc/intel/haswell/sst-haswell-pcm.c:1109 hsw_pcm_probe() error: buffer\
overflow 'hsw_dais' 4 <= 4
...
Fix it by initializing the index(i) to correct value.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is best to use the physical reset if it is available. This patch adds
support for a GPIO controlled physical reset for the chip.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Passing &mcasp->ruledata[dir] to snd_pcm_hw_rule_add() is not correct since
commit:
7b3d165a2821 ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream
now sets up the struct based on the substream->stream (0 or 1) while we pass
a pointer which we take with dir (1 or 2). This will lead kernel crash.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Restructure the sound/soc/intel/ directory: create atom folder, and move
sst atom platform files here.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Restructure the sound/soc/intel/ directory: create baytrail folder, and move
sst baytrail platform files here.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Restructure the sound/soc/intel/ directory: create boards folder, and move
sst boards files here.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Restructure the sound/soc/intel/ directory: create haswell folder, and
move haswell platform files here.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Restructure the sound/soc/intel/ directory: create common folder, and move
sst common files here.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The earpiece on wm5102 is mono, thus there is no output 3R. Don't toggle
the volume update bits for this output, although worth noting that doing
so had no negative effects it is just redundant.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The delay time after a reset in the codec probe callback was too short,
and did not work on certain hw because the codec needs more time to
power on. This increases the delay time from 1us to 1ms.
Signed-off-by: Pascal Huerst <pascal.huerst@gmail.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc.
Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at
192kHz) to work correctly. By default, keep the prealloc at 64kB.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The serializer direction definitions runs from 1 to 2, which does not
suite the purpose. The substream->stream is perfect for the purpose
and should have been used from the beginning.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode
and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when
applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this
fix does not affect the functionality in any way.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm8350 driver is the last driver that still uses the delayed_work field
from the snd_soc_dapm_context struct. Moving this over to the driver's
private data struct will allow us to remove the field from the DAPM context,
which will drastically reduce its size.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The only two users of the suspend_bias_level field were two rather old
drivers which weren't exactly doing things by the book. Those drivers have
been updated and field is now unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When being powered on, either initially on probe or when resuming from
suspend, the wm8971 configures the device for quick output capacitor
charging. Since the charging can take a rather long time (up to multiple
seconds) it is done asynchronously without blocking. A delayed work item is
run once the charging is finished and the device is switched to the target
bias level.
This all done asynchronously to the regular DAPM sequence accessing the same
data structures and registers without any looking, which can lead to race
conditions. Furthermore this potentially delays the start of stream on the
CODEC while the rest of the system is already up and running, meaning the
first bytes of audio are lost. It also does no comply with the assumption of
the DAPM core that if set_bias_level() returned successfully the device will
be at the requested bias level.
This patch slightly refactors things and makes sure that the caps charging
is properly integrated into the DAPM sequence. When transitioning from
SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast
charging mode and a work item will be scheduled that puts it back into
standby charging once the charging period has elapsed. If a playback or
capture stream is started while charging is in progress the driver will now
wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure
that charging is done asynchronously in the background when the chip is
idle, but at the same time makes sure that playback/capture is not started
before the charging is done.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When being powered on, either initially on probe or when resuming from
suspend, the wm8971 configures the device for quick output capacitor
charging. Since the charging can take a rather long time (up to multiple
seconds) it is done asynchronously without blocking. A delayed work item is
run once the charging is finished and the device is switched to the target
bias level.
This all done asynchronously to the regular DAPM sequence accessing the same
data structures and registers without any looking, which can lead to race
conditions. Furthermore this potentially delays the start of stream on the
CODEC while the rest of the system is already up and running, meaning the
first bytes of audio are lost. It also does no comply with the assumption of
the DAPM core that if set_bias_level() returned successfully the device will
be at the requested bias level.
This patch slightly refactors things and makes sure that the caps charging
is properly integrated into the DAPM sequence. When transitioning from
SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast
charging mode and a work item will be scheduled that puts it back into
standby charging once the charging period has elapsed. If a playback or
capture stream is started while charging is in progress the driver will now
wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure
that charging is done asynchronously in the background when the chip is
idle, but at the same time makes sure that playback/capture is not started
before the charging is done.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The delayed work used by the wm8971 driver to manage the caps charging
doesn't have any special requirements that would justify using a custom
workqueue, just use the generic system_power_efficient_wq instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
atmel-pcm.c was split into two files to create a generic framework for both PDC
and DMA.
atmel-pcm-dma.c is using the generic dmaengine framework since 95e0e07e710e
(ASoC: atmel-pcm: use generic dmaengine framework).
Merge atmel-pcm.c in atmel-pcm-pdc.c as this is now the only user.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas R-Car sound SRC (= Sampling Rate Converter) has
Asynchronous/Synchronous SRC mode. Asynchronous mode is already
supported via DPCM. This patch adds Synchronous mode on it.
The condition of enabling Synchronous mode are
- SoC is clock master
- Sound uses SRC
- Sound doesn't use DVC
- Sound card uses DPCM (= rsrc-card card)
amixer set "SRC Out Rate" on
aplay xxx.wav &
amixer set "SRC Out Rate" 48000
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A driver's device data should and can be const. This is a follow-up on
commit 33187fb4a203 (ASoC: rsnd: constify of_device_id array) which
marked the of_device_id as const.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
A driver's platform_device_id and device data should and can be const.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the file header only GPL v2 applies to it. Fix the
MODULE_LICENSE parameter accordingly.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch supports DPCM based sampling rate convert on Renesas sound
driver. It assumes...
1. SRC is implemented as FE
2. BE dai_link supports .be_hw_params_fixup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes useless debug message. especially some kind of
"probed" message will be printed from core.c if it has #define DEBUG
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
clk_prepare_enable()/clk_disable_unprepare() uses mutex inside,
in concretely clk_prepare()/clk_unprepare().And it uses __schedule().
Then, raw_spin_lock/unlock_irq() is called, and it breaks Renesas
sound driver's spin lock irq.
This patch separates thesse into clk_prepare()/clk_unprepare() and
clk_enable/clk_disable. And call clk_prepare()/clk_unprepare() from
probe/remove function. Special thanks to Das Biju.
Reported-by: Das Biju <biju.das@bp.renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd-dpcm-card is supporting DPCM FE/BE sound card.
This patch adds .be_hw_params_fixup and enabled sampling convert rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound card has "sampling rate convert" feature which
should be implemented via DPCM.
But, sound card driver point of view, it is difficult to add
this DPCM feature on simple-card driver. Especially, DT binding
support is very difficult.
This patch implements DPCM feature on DT as Renesas specific sound card.
This new driver is copied from current simple-card driver.
Main difference between simple-card and this driver are...
1. removed unused feature from simple-card
2. removed driver named prefix from DT property
3. CPU will be FE, CODEC will be BE with snd-soc-dummy
4. it supports sampling rate convert via .be_hw_params_fixup
5. board specific routing is implemented in driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial typo fix.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Anish Kumar <Anish.Kumar@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver will set RT5645_DEPOP_MAN bit in headphone power up
depop process. We need to restore it in headphone power down
process. Otherwise, we will get headphone noise when push button
function is enabled.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In codec bias level off, we need to disable gate mode with MCLK
for power saving. It is set by one bit. We don't need to write
while register for that.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
RT5645 doesn't support auto incrementing writes so driver should set
the use_single_rw flag for regmap.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5650 and rt5645 use different register bits for format configuration.
This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support
both codecs.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The smatch tool report warning:
...
CHECK sound/soc/intel/sst-haswell-pcm.c
sound/soc/intel/sst-haswell-pcm.c:1110 hsw_pcm_probe() error: buffer overflow\
'hsw_dais' 4 <= 4
sound/soc/intel/sst-haswell-pcm.c:1112 hsw_pcm_probe() error: buffer overflow\
'hsw_dais' 4 <= 4
...
fix it by use its own struct member for post-process module, rather than sharing
unused pcm member.
Signed-off-by: Lu, Han <han.lu@intel.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Having to set different formats on the CPU side and the CODEC side of a DAI
link is usually indication that something is terribly wrong and in most
cases is a result of a broken driver that implements a set_fmt() callback
which does not follow the specification. In the past this feature has been
used to work around broken drivers, rather than fixing them. We don't really
want to encourage this, so remove support for setting different formats on
both ends of the link.
Along the way switch to static DAI format setup by setting the the dai_fmt
field of the snd_soc_dai_link rather than calling snd_soc_dai_fmt().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As davinci card gets registered using 'devm_' api
there is no need to unregister the card in 'remove'
function.
Hence drop the 'remove' function.
Fixes: ee2f615d6e59c (ASoC: davinci-evm: Add device tree binding)
Signed-off-by: Manish Badarkhe <manishvb@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
We have a HP machine which use the codec node 0x17 connecting the
internal speaker, and from the node capability, we saw the EAPD,
if we don't set the EAPD on for this node, the internal speaker
can't output any sound.
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1436745
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The total stream number of Sunrise Point's input and output stream
exceeds 15, which will cause some streams do not work because
of the overflow on SDxCTL.STRM field if using the legacy
stream tag allocation method.
This patch uses the new stream tag allocation method by add
the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
