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>
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>
Some BIOS version of Fujitsu Lifebook T731 seems to set up the
headphone pin (0x21) without the assoc number 0x0f while it's set only
to the output on the docking port (0x1a). With the recent commit
[03ad6a8c93: ALSA: hda - Fix "PCM" name being used on one DAC when
there are two DACs], this resulted in the weird mixer element
mapping where the headphone on the laptop is assigned as a shared
volume with the speaker and the docking port is assigned as an
individual headphone.
This patch improves the situation by correcting the headphone pin
config to the more appropriate value.
Reported-and-tested-by: Taylor Smock <smocktaylor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Some M-Audio devices require to receive bootup command just after
powering on, while codes in BeBoB driver doesn't work properly in
big-endian machine because the command should be aligned by
little-endian.
This commit fixes this bug. This fix should go to stable kernel.
Cc: Takayuki Shiroma <t.shiroma.oki@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, snd_hdac_power_up()/down() helpers checks whether the codec
is being in pm (suspend/resume), and skips the call of runtime get/put
during it. This is needed as there are lots of power up/down
sequences called in the paths that are also used in the PM itself. An
example is found in hda_codec.c::codec_exec_verb(), where this can
power up the codec while it may be called again in its power up
sequence, too.
The above works in most cases, but sometimes we really want to wait
for the real power up. For example, the control element get/put may
want explicit power up so that the value change is assured to reach to
the hardware. Using the current snd_hdac_power_up(), however,
results in a race, e.g. when it's called during the runtime suspend is
being performed. In the worst case, as found in patch_ca0132.c, it
can even lead to the deadlock because the code assumes the power up
while it was skipped due to the check above.
For dealing with such cases, this patch makes snd_hdac_power_up() and
_down() to two variants: with and without in_pm flag check. The
version with pm flag check is named as snd_hdac_power_up_pm() while
the version without pm flag check is still kept as
snd_hdac_power_up(). (Just because the usage of the former is fewer.)
Then finally, the patch replaces each call potentially done in PM with
the new _pm() variant.
In theory, we can implement a unified version -- if we can distinguish
the current context whether it's in the pm path. But such an
implementation is cumbersome, so leave the code like this a bit messy
way for now...
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96271
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
... so that user-space can know that the whole nodes have been
created. Unfortunately, this can't be implemented easily in race-free
way, so it's a kind of compromise.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell create new platform with ALC288 codec.
This patch will enable headset mode for Dino platform.
[slight code refactoring and compile fix by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pin sense will active when power pin is wake up.
Power pin will not wake up immediately during resume state.
Add some delay to wait for power pin activated.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI/DP codec on SKL/BSW is in the power well.
The power well must be turned on before probing the
HDMI/DP codec.
This is a temporary patch, which will power on the
powerwell by adding AZX_DCAPS_I915_POWERWELL for SKL
and BSW. After restructuring and new flag is added,
this patch will be reverted.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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:
7b3d165a28 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 Audigy 5/Rx is essentially an Audigy 4 behind a PLX PCIe-
bridge with an additional TOSLINK output.
Signed-off-by: Michael Gernoth <michael@gernoth.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return a negative error code on failure.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
identifier ret; expression e1,e2;
@@
(
if (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
we are dereferencing pcm first then checking pcm. instead now lets put
them in same if condition so that pcm is checked first.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this quirk allows us to avoid the noisy
"cannot get freq at ep 0x1" message in dmesg output every time
playback starts.
This ought to affect other Benchmark DAC1 variations using the same
"Microchip Technology, Inc." chip as well, but I have only tested
with the "Pre" variant.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell new platform of ALC256 audio codec.
Support headset mode for Dell ALC256 platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recent Realtek codecs support the finer power state control on each
widget. Let's enable the new feature.
Tested-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we assumed that the node attributes like amp values remain
during the power state transition of the node itself. While this is
true for IDT/STAC codecs I've tested, but some other codecs don't seem
behaving in that way.
This patch implements a partial sync mechanism specific to the given
widget node. Now we've merged the regmap support, and it can be
easily written with regcache_sync_region().
Tested-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the support of regmap in HD-audio infrastructure.
Many in-house cache codes in HD-audio driver are relaced with the
more standard regmap base now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A regression was introduced in 7639a06c23: if AC_PAR_SUBSYSTEM_ID
reads as zero, one should retry using AC_VERB_GET_SUBSYSTEM_ID.
This seems to hit many codecs (my own laptop included), and causes
quirks for some machines not to apply correctly.
Reported-by: TienFu Chen <tienfu.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 95e0e07e71
(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>
This patchset include two new extcon driver and fix minor issue of extcon
driver.
Detailed description for patchset:
1. new extcon-max77843.c and extcon-usb-gpio.c extcon driver
- extcon-max77843.c driver support the MAXIM MAX77843 MUIC (Micor-USB Interface
Controller) device which handles the various external connectors such as TA/USB
/USB-HOST/JIG and so on.
- extcon-usb-gpio.c driver support the USB and USB-HOST cable detection by
using the GPIO pin which is connected to USB ID pin. This GPIO pin updates the
USB cable states.
2. Rename the filename of extcon core driver and add missing locking mechanism
- Rename the previous extcon-class driver.c as extcon.c because '-class'
postfix is not necessary word.
- extcon core driver (extcon.c) used the raw_notifier_chain. It must be
protected by locking mechanism to avoid the list changing while
extcon_update_state() is executed.
3. Fix minor issue of extcon drviers
- Fix cable name by using the capital letter instead of small letter on
extcon-max77693.c driver.
- Clean-up code of extcon-arizona.c to detect headphone cable.
- Fix the wrong return type and variable type on extcon-max77843.c.
- Fix the checkpatch warning of all extcon drivers.
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Merge tag 'extcon-next-for-4.1' of git://git.kernel.org/pub/scm/linux/kernel/git/chanwoo/extcon into char-misc-next
Chanwoo writes:
Update extcon for v4.1
This patchset include two new extcon driver and fix minor issue of extcon
driver.
Detailed description for patchset:
1. new extcon-max77843.c and extcon-usb-gpio.c extcon driver
- extcon-max77843.c driver support the MAXIM MAX77843 MUIC (Micor-USB Interface
Controller) device which handles the various external connectors such as TA/USB
/USB-HOST/JIG and so on.
- extcon-usb-gpio.c driver support the USB and USB-HOST cable detection by
using the GPIO pin which is connected to USB ID pin. This GPIO pin updates the
USB cable states.
2. Rename the filename of extcon core driver and add missing locking mechanism
- Rename the previous extcon-class driver.c as extcon.c because '-class'
postfix is not necessary word.
- extcon core driver (extcon.c) used the raw_notifier_chain. It must be
protected by locking mechanism to avoid the list changing while
extcon_update_state() is executed.
3. Fix minor issue of extcon drviers
- Fix cable name by using the capital letter instead of small letter on
extcon-max77693.c driver.
- Clean-up code of extcon-arizona.c to detect headphone cable.
- Fix the wrong return type and variable type on extcon-max77843.c.
- Fix the checkpatch warning of all extcon drivers.
A driver's device data should and can be const. This is a follow-up on
commit 33187fb4a2 (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>
HD-audio doesn't support the bulk access. Currently it works even
without this flag as implicitly assumed, but it's safer to set
explicitly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: ee2f615d6e (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
Although they can be written, handle a few verbs as read-only in
regmap interface: CONFIG_DEFAULT, CONV and CVT_CHAN_COUNT. These are
either updated in PCM or HDMI management code in a volatile manner, or
just needed only as parameter, thus they don't need to be written at
resume sync.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
struct kiocb now is a generic I/O container, so move it to fs.h.
Also do a #include diet for aio.h while we're at it.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
The WM8741 DAC supports the following typical audio sampling rates:
44.1kHz, 88.2kHz, 176.4kHz (eg: with a master clock of 22.5792MHz)
32kHz, 48kHz, 96kHz, 192kHz (eg: with a master clock of 24.576MHz)
For the rates lists, we should use 82000 instead of 88235, 176400
instead of 1764000 and 192000 instead of 19200 (seems to be a typo).
Signed-off-by: Sergej Sawazki <ce3a@gmx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
commit c66150824b ("ASoC: dapm: add code to configure dai link
parameters") introduced the following build warning:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm':
sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf'
discards 'const' qualifier from pointer target type
snprintf(w_param_text[count], len,
This patch fixes this by switching to using devm_kasprintf. This also
saves a couple of lines of code.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
My codec has a beep-generating node:
$ cat /proc/asound/card1/codec#0
Codec: VIA VT1802
...
Vendor Id: 0x11068446
Subsystem Id: 0x15587410
Revision Id: 0x100000
...
Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out
Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1
Amp-Out vals: [0x0a]
Power states: D0 D1 D2 D3
Power: setting=D0, actual=D0
...
But I was missing the:
Control: name=...
entries that I need to manage this widget from alsamixer. With this
patch (based on the similar Mono Amp-Out handling in
patch_conexant.c), I get a new:
input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/hdaudioC1D0/input15
entry in dmesg and controls to manage that beep:
$ cat /proc/asound/card1/codec#0 | grep -A5 Beep
Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out
Control: name="Beep Playback Volume", index=0, device=0
ControlAmp: chs=1, dir=Out, idx=0, ofs=0
Control: name="Beep Playback Switch", index=0, device=0
ControlAmp: chs=1, dir=Out, idx=0, ofs=0
Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1
Amp-Out vals: [0x12]
Power states: D0 D1 D2 D3
Power: setting=D0, actual=D0
[rebased and modified for the latest tree by tiwai]
Signed-off-by: W. Trevor King <wking@tremily.us>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just forgotten to remove. It's now in sound/hdaudio.h.
Reported-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current soc_init_card_debugfs() is called from snd_soc_register_card()
but, soc_cleanup_card_debugfs() is called from soc_cleanup_card_resources(),
not from paired function.
This differences don't matter for now. But if anyone wants to implement
a proper hotplug/unplug, this difference would become clearer.
Now, we can assume that snd_soc_instantiate_card() and
soc_cleanup_card_resources() are paired function.
soc_init_card_debugfs() / soc_cleanup_card_debugfs() paired function
should be called from these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set rule constraints to allow only combinations of sample-rate,
sample-format, and channels counts that can be played/captured with
reasonable sample-rate accuracy.
The logic with tdm-slots and serializers (=i2s data wires) goes like
this: The first wire will take all channels up to number of tdm-slots,
before following wires (if any) are used. If the first wire is used
fully, the remaining wires share the same clocks and the divider can
be calculated for the first wire.
Also, takes the number of tdm-slots into account when implicitly
selecting the BLCK divider.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace duplicated const keyword for 'dvc_ramp_rate' with proper
array of const pointers to const strings.
Signed-off-by: Krzysztof Kozlowski <k.kozlowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4642 has power-save mode for stereo line to reduce pop noise.
This patch enables it.
Signed-off-by: Takeshi Kihara <takeshi.kihara.df@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is worth to check the regmap_read result for ID check since it
is the first regmap_read. And we can check if there is any i2c
issue.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace duplicated const keyword for 'sampleclock_sources' with proper
array of const pointers to const strings.
Signed-off-by: Krzysztof Kozlowski <k.kozlowski@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently GPIO4 is hardcoded to output the pll-lock signal.
Unfortunately this is after the pll-out GPIO is configured which
is selectable in the device tree. Therefore it is not possible to
use GPIO4 for pll-out. Therefore this patch removes the
configuration of GPIO4.
Signed-off-by: Howard Mitchell <hm@hmbedded.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
This model uses the same dock port as the previous generation.
Signed-off-by: Sebastian Wicki <gandro@gmx.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to cap "ucontrol->id.index / num_busses_in(chip)" so the we
don't read beyond the end of the array.
I also adding a check on "in" and changing the type in
snd_echo_mixer_put() from short to unsigned int. Those changes are done
for symmetry and are cosmetic.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"err" is always a negative error code here, so there is no point in
checking. Removing the check silences a static checker warning and
makes the code a bit more clear. Also we don't need to initialize "err".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 16bit COEF read/write is pretty standard for many codecs, and they
can be cached in most cases -- more importantly, they need to be
restored at resume. For making this easier, add the cache support to
regmap. If the codec driver wants to cache the COEF access, set
codec->cache_coef flag and issue AC_VERB_GET_PROC_COEF with the coef
index in LSB 8 bits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio has quite a few asymmetrical ways of accessing verbs, and one
of typical ones is GET/SET_POWER_STATE verbs. While it takes only the
power state for setting, it returns a combination of states for
getting. For making the state handling simpler, this patch adds a
code to translate the value returned from GET_POWER_STATE to return
only the actual state or -1 for error. In that way, the driver can
simplify the power state management.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio spec is inconvenient regarding the handling of stereo volume
controls. It can set and get only single channel at once (although
there is a special option to set the same value to both channels).
This patch provides a fake pseudo-register via the regmap access so
that the stereo channels can be read and written by a single call.
It'd be useful, for example, for implementing DAPM widgets.
A stereo amp pseudo register consists of the encoding like the normal
amp verbs but it has both SET_LEFT (bit 13) and SET_RIGHT (bit 12)
bits set. The regmap reads and writes a 16bit value for this pseudo
register where the upper 8bit is for the right chanel and the lower
8bit for the left channel.
Note that the driver doesn't recognize conflicts when both stereo and
mono channel registers are mixed. Mixing them would certainly confuse
the operation. So, use carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patches, this patch converts also to the regmap, at
this time, the cached verb writes are the target. But this conversion
needs a bit more caution than before.
- In the old code, we just record any verbs as is, and restore them at
resume. For the regmap scheme, this doesn't work, since a few verbs
like AMP or DIGI_CONVERT are asymmetrical. Such verbs are converted
either to the dedicated function (snd_hda_regmap_xxx_amp()) or
changed to the unified verb.
- Some verbs have to be declared as vendor-specific ones before
accessing via regmap.
Also, the minor optimization with codec->cached_write flag is dropped
in a few places, as this would confuse the operation. Further
optimizations will be brought in the later patches, if any.
This conversion ends up with a drop of significant amount of codes,
mostly the helper codes that are no longer used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codecs may have own vendor-specific verbs, and we need to allow each
driver to give such verbs for cached accesses. Here a verb can be put
into a single array and looked through it at readable and writeable
callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp hash table was used for recording the cached reads of some
capability values like pin caps or amp caps. Now all these are moved
to regmap as well.
One addition to the regmap helper is codec->caps_overwriting flag.
This is set in snd_hdac_override_parm(), and the regmap helper accepts
any register while this flag is set, so that it can overwrite even the
read-only verb like AC_VERB_PARAMETERS. The flag is cleared
immediately in snd_hdac_override_parm(), as it's a once-off flag.
Along with these changes, the no longer needed amp hash and relevant
fields are removed from hda_codec struct now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch converts the amp access functions to the regmap helpers.
The amp values were formerly cached in the own hash table. Now it's
dropped by the regmap's cache.
The only tricky conversion is snd_hda_codec_amp_init(). This function
shouldn't do anything if the amp was already initialized. For
achieving this behavior, a value is read once at first temporarily in
the cache-only mode. Only if it returns an error, i.e. the item
still doesn't exist in the cache, it proceeds to the update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes we need the uncached reads, e.g. for refreshing the tree.
This patch provides the helper function for that and uses it for
refreshing widgets, reading subtrees and the whole proc reads.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's start converting the access functions to regmap.
The first one is the simplest, just converting the codec parameter
read helper function snd_hda_param_read().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds an infrastructure to support regmap-based verb
accesses. Because o the asymmetric nature of HD-audio verbs,
especially the amp verbs, we need to translate the verbs as a sort of
pseudo registers to be mapped uniquely in regmap.
In this patch, a pseudo register is built from the NID, the
AC_VERB_GET_* and 8bit parameters, i.e. almost in the form to be sent
to HD-audio bus but without codec address field. OTOH, for writing,
the same pseudo register is translated to AC_VERB_SET_* automatically.
The AC_VERB_SET_AMP_* verb is re-encoded from the corresponding
AC_VERB_GET_AMP_* verb and parameter at writing.
Some verbs has a single command for read but multiple for writes. A
write for such a verb is split automatically to multiple verbs.
The patch provides also a few handy helper functions. They are
designed to be accessible even without regmap. When no regmap is set
up (e.g. before the codec device instantiation), the direct hardware
access is used. Also, it tries to avoid the unnecessary power-up.
The power up/down sequence is performed only on demand.
The codec driver needs to call snd_hdac_regmap_exit() and
snd_hdac_regmap_exit() at probe and remove if it wants the regmap
access.
There is one flag added to hdac_device. When the flag lazy_cache is
set, regmap helper ignores a write for a suspended device and returns
as if it was actually written. It reduces the hardware access pretty
much, e.g. when adjusting the mixer volume while in idle. This
assumes that the driver will sync the cache later at resume properly,
so use it carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now let's take the basic tracepoints back to the HD-audio driver.
The three bus tracepoints, hda_send_cmd, hda_get_response and
hda_unsol_event are revived but in a slightly different form.
Since we don't assign the card number there, print the bus device name
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the driver is unloaded before the codec is bound, it still keeps
the runtime PM refcount up, and results in the unbalance. This patch
covers these cases by introducing a flag indicating the runtime PM
initialization and handling the codec registration procedure more
properly. It also fixes the missing input beep device as a gratis,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an overriding exec_verb op to struct hdac_device so that the call
via snd_hdac_exec_verb() can switch to a different route depending on
the setup. The codec driver sets this field so that it can handle the
errors or applying quirks appropriately. Furthermore, this mechanism
will be used for smooth transition for the regmap support in later
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the sysfs files assigned to the codec device on the
bus which were formerly identical with hwdep sysfs files. Now it
shows only a few core parameter, vendor_id, subsystem_id, revision_id,
afg, mfg, vendor_name and chip_name.
In addition, now a widget tree is added to the bus device sysfs
directory for showing the widget topology and attributes. It's just a
flat tree consisting of subdirectories named as the widget NID
including various attributes like widget capability bits. The AFG
(usually NID 0x01) is always found there, and it contains always
amp_in_caps, amp_out_caps and power_caps files. Each of these
attributes show a single value. The rest are the widget nodes
belonging to that AFG. Note that the child node might not start from
0x02 but from another value like 0x0a.
Each child node may contain caps, pin_caps, amp_in_caps, amp_out_caps,
power_caps and connections files. The caps (representing the widget
capability bits) always contain a value. The rest may contain
value(s) if the attribute exists on the node. Only connections file
show multiple values while other attributes have zero or one single
value.
An example of ls -R output is like below:
% ls -R /sys/bus/hdaudio/devices/hdaudioC0D0/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/:
01/ 04/ 07/ 0a/ 0d/ 10/ 13/ 16/ 19/ 1c/ 1f/ 22/
02/ 05/ 08/ 0b/ 0e/ 11/ 14/ 17/ 1a/ 1d/ 20/ 23/
03/ 06/ 09/ 0c/ 0f/ 12/ 15/ 18/ 1b/ 1e/ 21/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/01:
amp_in_caps amp_out_caps power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/02:
amp_in_caps amp_out_caps caps connections pin_caps pin_cfg
power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/03:
.....
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now some codes and functionalities of hda_codec struct are moved to
hdac_device struct. A few basic attributes like the codec address,
vendor ID number, FG numbers, etc are moved to hdac_device, and they
are accessed like codec->core.addr. The basic verb exec functions are
moved, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few basic codes for communicating over HD-audio bus are moved to
struct hdac_bus now. It has only command and get_response ops in
addition to the unsolicited event handling.
Note that the codec-side tracing support is disabled temporarily
during this transition due to the code shuffling. It will be
re-enabled later once when all pieces are settled down.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define the common hd-audio driver and device types to bind over
snd_hda_bus_type publicly. This allows to implement other type of
device and driver code over hd-audio bus.
Now both struct hda_codec and struct hda_codec_driver inherit these
new struct hdac_device and struct hdac_driver, respectively.
The bus registration is done in subsys_initcall() to assure it
before any other driver registrations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During probe, the Baytrail audio driver reports errors such as:
[44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0
[44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed
Those error messages are misleading, there is no error since the DMA
is explicitly not configured for Baytrail.
Add a test to remove DMA error checks when DMA is not configured
and return silently.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If den=1 and pllin_rate>20MHz then den and num are adjusted to 0
causing a divide by zero error a few lines further on. Therefore
this patch correctly scales num and den such that
pllin_rate/den < 20MHz as required in the device data sheet.
Signed-off-by: Howard Mitchell <hm@hmbedded.co.uk>
Signed-off-by: Mark Brown <broonie@sirena.org.uk>
Cc: stable@vger.kernel.org
ppc has special instruction forms to efficiently load and store values
in non-native endianness. These can be accessed via the arch-specific
{ld,st}_le{16,32}() inlines in arch/powerpc/include/asm/swab.h.
However, gcc is perfectly capable of generating the byte-reversing
load/store instructions when using the normal, generic cpu_to_le*() and
le*_to_cpu() functions eaning the arch-specific functions don't have much
point.
Worse the "le" in the names of the arch specific functions is now
misleading, because they always generate byte-reversing forms, but some
ppc machines can now run a little-endian kernel.
To start getting rid of the arch-specific forms, this patch removes them
from all the old Power Macintosh drivers, replacing them with the
generic byteswappers.
Signed-off-by: David Gibson <david@gibson.dropbear.id.au>
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
This patch adds missing dai_link stream_name
which is used when DPCM
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver sometimes fixups debug/error message format
30cc4faf70
(ASoC: rsnd: tidyup debug message format and timing)
337b0b4c5f
(ASoC: rsnd: error meesage indicates its port)
But, it still exist un-fomated error message. This patch fixup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It will be SRC interrupt endless loop f unusual situation happen.
This patch adds restart limit for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It will be SSI interrupt endless loop f unusual situation happen.
This patch adds restart limit for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is to ensure that 'alsactl restore' does not apply default
initialisation as the chip reset defaults are preferred.
Signed-off-by: Howard Mitchell <hm@hmbedded.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
David suggested that the name "power_mgmt" is too ambiguous. Rename
the flag with a bit clearer one "power_save_node".
Also, add the corresponding description to HD-Audio.txt, too.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some pins are used for controlling the LED with the VREF value.
This patch changes the power behavior of such pins to be constantly
up. A new state, pin_fixed, is introduced to nid_path to indicate
that the path contains the fixed pin. This improves also the
readability a bit for other static routes, too.
Then a helper function snd_hda_gen_fix_pin_power() is called from the
codec driver for such fixed pins, and it will create fake paths
containing only these pins with pin_fixed=1 flag.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Constify local structures (snd_soc_dai_ops, snd_soc_codec_driver,
regmap_config) and array (reg_defaults) which are not modified by the
driver and passed to core as pointer to const.
Signed-off-by: Krzysztof Kozlowski <k.kozlowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_device_id is always used as const.
(See driver.of_match_table and open firmware functions)
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the sst_acpi_probe memory allocation error path by setting right error
code and initiating the cleanup insteadof just returning
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check waves state before set parameter through ipc to prevent unexpected
operation. Also remove redundant check.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel MID DMA driver is going to be removed, commit should be a few lines
down near to dmaengine_slave_config() call in order to not confuse and at
quick look Synopsys DesignWare does seem to use some of the slave config
structure fields (see drivers/dma/dw/core.c: dwc_config()).
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel MID DMA driver is going to be removed by the coming commit
36111da783 ("dmaengine: intel-mid-dma: remove the driver") in spi.git
tree. Since there are no users for SST_DMA_TYPE_MID type the support for it
can be removed from here in advance.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the widget PM may turn off the pins, this might lead to the silent
output for beep when no explicit paths are given. This patch adds
fake output paths for the beep widget so that the output pins are
dynamically powered upon beep on/off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs used to have the own power controls but they were disabled
at transition to the generic parser due to the coding assuming the
fixed routes. Now we get the proper support of equivalently fine
power management in the generic parser, and the old kludges can be
replaced with it. This results in the reduction of lots of dead
codes.
The advanced PM feature is disabled as default like before for keeping
the compatible behavior. It's enabled via "Dynamic Power-Control"
mixer element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables the finer power state control of each widget
depending on the jack plug state and streaming state in addition to
the existing power_down_unused power optimization. The new feature is
enabled only when codec->power_mgmt flag is set.
Two new flags, pin_enabled and stream_enabled, are introduced in
nid_path struct for marking the two individual power states: the pin
plug/unplug and DAC/ADC stream, respectively. They can be set
statically in case they are static routes (e.g. some mixer paths),
too.
The power up and down events for each pin are triggered via the
standard hda_jack table. The call order is hard-coded, relying on the
current implementation of jack event chain (a la FILO/stack order).
One point to be dealt carefully is that DAC/ADC cannot be powered
on/off while streaming. They are pinned as long as the stream is
running. For controlling the power of DAC/ADC, a new patch_ops is
added. The generic parser provides the default callback for that.
As of this patch, only IDT/Sigmatel codec driver enables the flag.
The support on other codecs will follow.
An assumption we made in this code is that the widget state (e.g. amp,
pinctl, connections) remains after the widget power transition (not
about FG power transition). This is true for IDT codecs, at least.
But if the widget state is lost at widget power transition, we'd need
to implement additional code to sync the cached amp/verbs for the
specific NID.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add sampling rate control for ADC/DAC for ESI W192M.
Allow to switch between 48K/96K/192K sampling rate.
All DAC need to be mute when changing samplerate.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add text Line in/Mic for selecting input gain state in mixer for ESI W192M.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add TLV support to control volume using dB scale for input and ouput on ESI W192M.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update eeprom structure to C99 standard to be compliant with change in alsa.
It's just a notation change, no configuration change.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct copy/paste name from prodigy driver, no behaviour change, only name.
Signed-off-by: Clément Guedez <klem.dev@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dai-link params for codec-codec links were fixed. The fixed
link between codec and another chip which may be another codec,
baseband, bluetooth codec etc may require run time configuaration
changes. This change provides an optional alsa control to select
one of the params from a list of params.
Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch does two things:
- code refactoring with a local helper function,
- allow codec drivers to provide the specific PCM stream info pointers
only for overriding the non-NULL entries, instead of copying the
whole.
This simplifies the codec driver side (currently the only user is
alc269's 44kHz fixed rate).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As well as the usual collection of driver specific fixes there's a few
more generic things:
- Lots of fixes from Takashi for drivers using the wrong field in the
control union to communicate with userspace, leading to potential
errors on 64 bit systems.
- A fix from Lars for locking of the lists of devices we maintain,
mostly only likely to trigger during device probe and removal.
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Merge tag 'asoc-fix-v4.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.0
As well as the usual collection of driver specific fixes there's a few
more generic things:
- Lots of fixes from Takashi for drivers using the wrong field in the
control union to communicate with userspace, leading to potential
errors on 64 bit systems.
- A fix from Lars for locking of the lists of devices we maintain,
mostly only likely to trigger during device probe and removal.
It needs DAC -> Playback route instead of direct settings via
SND_SOC_DAPM_DAC. otherwise, it can't find correct path if
sound card used prefix name
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The jack creation code was in rt5670 codec driver before due to the
jack resources (gpio/irq) were defined under the node of codec device
in ACPI on Braswell. We used the snd_soc_jack_new() to create a jack
instance.
But now snd_soc_jack_new() is removed from upstream and we can't
use snd_soc_card_jack_new() in codec driver, so we move the jack
creation code to machine driver and pass the jack instance to codec
driver for further processing.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Jin Yao <yao.jin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On Braswell, we need to add some machine specific setting before suspend
and after resume. For example, disable/enable jack detection in codec so
use snd_soc_card suspend_pre and resume_post ops for this purpose.
Signed-off-by: Jin Yao <yao.jin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
