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 Atmel ASoC machine drivers don't have any compile time arch dependencies
anymore. Make it possible to select them when COMPILE_TEST is enabled to get
better compile test coverage.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_AT91_SOC_SAM9G20_WM8731 and SND_AT91_SOC_SAM9X5_WM8731 machine
driver symbols select SND_SOC_WM8731 which depends on SND_SOC_I2C_AND_SPI.
So the machine driver symbols need to depend on SND_SOC_I2C_AND_SPI as well,
otherwise we might end up with a invalid configuration, which will sooner or
later upset the randconfig auto-builders.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit [39d118677b: ALSA: ctl: evaluate macro instead of
numerical value] replaced the numbers with constants, but one place
was replaced wrongly with a different type. Fixed now.
Fixes: 39d118677b ('ALSA: ctl: evaluate macro instead of numerical value')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate.
Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed.
Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961.
Signed-off-by: Adam Honse <calcprogrammer1@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
readv() and writev() should _not_ ignore all but the first ->iov_len,
among other things. Really weird abuse of those syscalls - it
expects a vector element per channel, with identical lengths (it
actually assumes them to be identical - no checking is done).
readv() and writev() are really bad match for that. Unfortunately,
userland API is userland API and we can't do anything about them.
Converted to ->read_iter/->write_iter. Please, _please_ don't do
anything of that kind when designing new interfaces.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
This patch backs out a change that came in during the merge window which
selects a configuration for GPIO4 on pcm512x CODECs that may not be
suitable for all systems using the device. Changes for v4.1 will make
this properly configurable but for now it's safest to revert to the
v3.19 behaviour and leave the pin configuration alone.
Sorry for sending this direct at the last minute but due to the GPIO
misuse it'd be really good to get it in the release and I'd not realised
it hadn't been sent yet - between some travel, a job change and other
non-urgent fixes coming in I'd lost track of the urgency. It's been in
-next for several weeks now, is isolated to the driver and fairly clear
to inspection.
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Merge tag 'asoc-fix-v4.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound
Pull last-minute ASoC fix from Mark Brown:
"This patch backs out a change that came in during the merge window
which selects a configuration for GPIO4 on pcm512x CODECs that may not
be suitable for all systems using the device. Changes for v4.1 will
make this properly configurable but for now it's safest to revert to
the v3.19 behaviour and leave the pin configuration alone.
Sorry for sending this direct at the last minute but due to the GPIO
misuse it'd be really good to get it in the release and I'd not
realised it hadn't been sent yet - between some travel, a job change
and other non-urgent fixes coming in I'd lost track of the urgency.
It's been in -next for several weeks now, is isolated to the driver
and fairly clear to inspection"
* tag 'asoc-fix-v4.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound:
ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4
The r-car sound driver only works when CONFIG_OF is set, and
after a recent change has a compile-time dependency as well:
sound/built-in.o: In function `rsnd_dma_request_channel':
:(.text+0x9fb84): undefined reference to `of_dma_request_slave_channel'
This could be fixed either by adding a static inline wrapper
for the function, or by adding a Kconfig dependency. This
implements the second approach, which seems appropriate
because the driver in fact has a hard dependency.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 72adc61f46 ("ASoC: rsnd: 1st DMAC dma-names cares subnode")
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The optical ports on the E-mu 1010 (and dock) can be configured
for ADAT- or S/PDIF-mode, which is currently hardcoded to ADAT.
Add two mixer elements to expose this setting.
Tested on an E-mu 1010 PCIe with connected Micro Dock.
Signed-off-by: Michael Gernoth <michael@gernoth.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
currently some members related identical information are not fiiled
in returned parameter of SNDRV_CTL_IOCTL_ELEM_ADD. This is not better
for userspace application.
This commit copies information to returned value. When failing to copy
into userspace, the added elements are going to be removed. Then, no
applications can lock these elements between adding and removing because
these are already locked.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In operations of SNDRV_CTL_IOCTL_ELEM_INFO, identical information in
returned value is cleared. This is not better to userspace application.
This commit confirms to return full identical information to the
operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When event originator doesn't set numerical ID in identical information,
the event data includes no numerical ID, thus userspace applications
cannot identify the control just by unique ID in event data.
This commit fix this bug so as the event data includes all of identical
information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the dock on an E-mu 1010 card is disconnected, all outputs get
muted by the hardware. Add logic to detect a disconnect and unmute.
Signed-off-by: Michael Gernoth <michael@gernoth.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This model uses the same dock port as the previous generation.
Signed-off-by: Yves-Alexis Perez <corsac@debian.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The returned value of 'get/seq client pool' operation has zeroed value
for its client ID, against requested client ID.
This commit fix the bug by filling it with index value of referred
client object.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the generic IPC/mailbox APIs to replace the original processing
code for Broadwell platform.
Signed-off-by: Jin Yao <yao.jin@linux.intel.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the generic IPC/mailbox APIs to replace the original processing
code for Baytrail platform.
Signed-off-by: Jin Yao <yao.jin@linux.intel.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently in Intel SST driver, some similar IPC/mailbox processing
code are used in different platforms (e.g. in baytrail/broadwell).
This patch extracts the common code and creates new files
(sst-ipc.c/sst-ipc.h) to contain the common code and provide the generic
APIs for IPC/mailbox processing.
Signed-off-by: Jin Yao <yao.jin@linux.intel.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To keep consistency with the other Kconfig entries, use the audio
interface acronyms (SSI and SPDIF) in the Kconfig menu text.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_runtime_set_dai_fmt() is called after
soc_probe_link_dais(). this means snd_soc_dai_set_fmt() will be
called after soc_new_pcm().
Before appling 1efb53a220
(ASoC: simple-card: Remove support for setting differing DAI formats)
simple-card user had (1) snd_soc_dai_set_fmt() -> soc_new_pcm(),
but, after that it is (2) soc_new_pcm() -> snd_soc_dai_set_fmt().
At least rsnd driver is assuming (1) pattern.
This patch move snd_soc_dai_set_fmt() into soc_probe_link_dais()
after the dai_link->init section to solve this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using platform_get_resource() and devm_ioremap_resource() can make the
code a bit simpler.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The functions snd_emu10k1_proc_spdif_read and snd_emu1010_fpga_read
acquire the emu_lock before accessing the FPGA. The function used
to access the FPGA (snd_emu1010_fpga_read) also tries to take
the emu_lock which causes a deadlock.
Remove the outer locking in the proc-functions (guarding only the
already safe fpga read) to prevent this deadlock.
[removed superfluous flags variables too -- tiwai]
Signed-off-by: Michael Gernoth <michael@gernoth.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using devm_snd_soc_register_component() can make the code shorter and
cleaner.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4642 doesn't have Mono record, ak4643 have it, but not supported.
This patch fixes channel mismatch
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A comma was used instead of a semicolon, which may lead to a build
error.
Fixes: cffd396681 ('ALSA: hda/realtek - Fix the regression by widget power-saving')
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SNDRV_CTL_TLV_OP_XXX is defined but not used in core code. Instead,
raw numerical value is evaluated.
This commit replaces these values to these macros for better looking.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3237"
Signed-off-by: Dmitry M. Fedin <dmitry.fedin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the case for Realtek, VIA codec driver needs this ops as
well for making the widget power-save working.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the registration of a debugfs directory fails this is treated as a
non-fatal error in ASoC and operation continues as normal. This means we
need to be careful and check if the parent debugfs directory exists if we
try to register a debugfs file or sub-directory. Otherwise we might end up
passing NULL for the parent and the file or directory will be registered in
the top-level debugfs directory.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Failing to register the debugfs entries is not fatal and will not affect
normal operation of the sound card. Don't abort the card registration if
soc_dpcm_debugfs_add() fails.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
RT286 can't do register reset. If the hardware power is still existing
in power off, rt286 will keep the register settings. So, we need to
restore the default register value in probe to make sure the cache value
is the same as the real register value.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Create the card debugfs directory at the begining of the initilization
rather then the end as various steps in the initilization sequence will try
to register files and sub-directories in the card directory.
Fixes: 4e2576bd36 ("ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card()")
Reported-by: Fabio Estevam <festevam@gmail.com>
Reported-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
While enabling the widget power-saving for ALC269 & co, the important
setup was forgotten -- stream_pm ops. Without this setup, the paths
for PCM won't be powered up at all.
Also, the power_filter callbacks used in ALC269 & co need to chain to
the default snd_hda_gen_path_power_filter().
Tested-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My static checker complains about these because it looks like the
multiply can overflow and then we cast to a larger data type. I don't
think this is a problem, but it's also harmless to do the cast earlier
so let's silence the static checker warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the generic parser sets codec->power_filter when
power_save_node flag is set. But this overrides the existing filter
that has been already set by the codec driver, thus it looses some
features. Instead, set the default power_filter only when it's not
set yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the check of power_save_node flag at the beginning of the function
so that it skips the rest if the flag isn't set. In this way, we can
call this function safely no matter whether the widget power-saving is
really used or not.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the widget power-saving is enabled, the first automute hook
invocation checks through the whole pins and it also tries to
synchronize the power state. However, this results in a wrong state
because it calls unconditionally snd_hda_jack_detect_state().
This patch adds a check of jack detectability before the actual jack
detection call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new widget power-saving tries to apply the power change no matter
whether the node has a power cap or not. It's bad (although most of
codecs chip just ignore it). Check the capability properly
beforehand.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hdac regmap code checks whether the codec is powered on while
accessing, but there must be an exception -- the verbs to control the
power state.
Currently HD-audio driver doesn't access them via regmap, so this
patch doesn't fix any current behavior, but it's just for future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
trivial conflict in net/socket.c and non-trivial one in crypto -
that one had evaded aio_complete() removal.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
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>
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>
