Check return value from call to devm_kzalloc() in order to prevent a
potential NULL pointer dereference.
Also, notice that it makes no sense to allocate any resources if
res = platform_get_resource(pdev, IORESOURCE_MEM, 0); fails,
so move the call to devm_kzalloc() below the mentioned code.
Lastly, improve the use of sizeof in the call to devm_kzalloc() by
changing it from sizeof(struct i2s_dev_data) to sizeof(*adata)
This issue was detected with the help of Coccinelle.
Fixes: ac289c7ec0 ("ASoC: amd: add ACP3x PCM platform driver")
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.com>
Cc: Dan Carpenter <dan.carpenter@oracle.com>
Cc: Kees Cook <keescook@chromium.org>
Cc: Will Deacon <will.deacon@arm.com>
Cc: Greg KH <greg@kroah.com>
Signed-off-by: Willy Tarreau <w@1wt.eu>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a potential NULL pointer dereference in case devm_kzalloc()
fails and returns NULL.
Fix this by adding a NULL check on rt5514_dsp.
This issue was detected with the help of Coccinelle.
Fixes: 6eebf35b0e ("ASoC: rt5514: add rt5514 SPI driver")
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Dan Carpenter <dan.carpenter@oracle.com>
Cc: Kees Cook <keescook@chromium.org>
Cc: Will Deacon <will.deacon@arm.com>
Cc: Greg KH <greg@kroah.com>
Signed-off-by: Willy Tarreau <w@1wt.eu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix typo which causes headphone no sound while using BCLK
as PLL source.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
cpu and platform are optional components in DAI links. For example
codec-codec links usually have no platform set.
Call snd_soc_find_component only if the name or of_node of
a cpu or platform is set. Otherwise it will return NULL and
soc_init_dai_link bails out immediately with -EPROBE_DEFER,
meaning registering a card with NULL cpu or platform in DAI links
can never succeed.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Signed-off-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are some use cases where you're checking for a lot of things on a
card and it makes sense that you might end up trying to call
snd_soc_find_component() without either a name or an of_node. Currently
in that case we try to dereference the name and crash but it's more
useful to allow the caller to just treat that as a case where we don't
find anything, that error handling will already exist.
Inspired by a patch from Ajit Pandey fixing some callers.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_init_platform initializes pointers to snd_soc_dai_link which is
statically allocated and it does this by devm_kzalloc. In the event of
an EPROBE_DEFER the memory will be freed and the pointers are left
dangling. snd_soc_init_platform sees the dangling pointers and assumes
they are pointing to initialized memory and does not reallocate them on
the second probe attempt which results in a use after free bug since
devm has freed the memory from the first probe attempt.
Since the intention for snd_soc_dai_link->platform is that it can be set
statically by the machine driver we need to respect the pointer in the
event we did not set it but still catch dangling pointers. The solution
is to add a flag to track whether the pointer was dynamically allocated
or not.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
AES channel status carries various audio parameters. If channel status is
detected, current patch extracts sample rate and bit depth parameters of
the incoming stream during capture.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build error:
ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined!
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes unused variables which also fixes below warnings:
msm8916-wcd-digital.c:245:30: warning: 'rx2_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:234:30: warning: 'rx_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:224:26: warning: 'adc2_mux_text'
defined but not used [-Wunused-const-variable=]
static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
^~~~~~~~~~~~~
msm8916-wcd-digital.c:223:26: warning: 'rx_mix2_text'
defined but not used [-Wunused-const-variable=]
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver cs4341 can be built with SPI and/or I2C, but it has to be one
of them at least. When I2C is set as a module we see the warning below:
sound/soc/codecs/cs4341.c:213:12: warning: ‘cs4341_probe’
defined but not used [-Wunused-function]
static int cs4341_probe(struct device *dev)
^~~~~~~~~~~~
Rework so that we use IS_ENABLED instead of defined. Also change so
SND_SOC_CS4341 depends on SND_SOC_I2C_AND_SPI to we dont' get a link
error when SND_SOC_CS4341=y, I2C=m and REGMAP_I2C=m is set.
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the refactoring of HD-audio display power management, the
display power status is managed per domain. Meanwhile the ASoC
hdac_hdmi driver still keeps and relies (incorrectly) on the
refcounting together with ASoC skl driver, and this leads to the
display state always on.
This patch is an attempt to address the regression by simplifying the
PM code of ASoC skl and hdac_hdmi drivers. Basically, since the
refactoring, we don't have to manage the display power at HD-audio
controller suspend / resume but only at HD-audio HDMI codec suspend /
resume. So the patch drops the superfluous snd_hdac_display_power()
calls in skl driver.
Meanwhile, in hdac_hdmi side, we rewrite the PM call just to re-use
the runtime PM callbacks like other drivers do. Now the logic is
simple: turn off at suspend and turn on at resume.
The patch also fixes the possibly missing display-power off at skl
driver removal as well as some error paths at probe.
Fixes: 029d92c289 ("ALSA: hda: Refactor display power management")
Reported-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix typo for model alc255-dell1 to alc225-dell1.
Enable headset mode support for new WYSE NB platform.
Fixes: a26d96c780 ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've always had a weird situation around dma_zalloc_coherent. To
safely support mapping the allocations to userspace major architectures
like x86 and arm have always zeroed allocations from dma_alloc_coherent,
but a couple other architectures were missing that zeroing either always
or in corner cases. Then later we grew anothe dma_zalloc_coherent
interface to explicitly request zeroing, but that just added __GFP_ZERO
to the allocation flags, which for some allocators that didn't end
up using the page allocator ended up being a no-op and still not
zeroing the allocations.
So for this merge window I fixed up all remaining architectures to zero
the memory in dma_alloc_coherent, and made dma_zalloc_coherent a no-op
wrapper around dma_alloc_coherent, which fixes all of the above issues.
dma_zalloc_coherent is now pointless and can go away, and Luis helped
me writing a cocchinelle script and patch series to kill it, which I
think we should apply now just after -rc1 to finally settle these
issue.
-----BEGIN PGP SIGNATURE-----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=VIBB
-----END PGP SIGNATURE-----
Merge tag 'remove-dma_zalloc_coherent-5.0' of git://git.infradead.org/users/hch/dma-mapping
Pull dma_zalloc_coherent() removal from Christoph Hellwig:
"We've always had a weird situation around dma_zalloc_coherent. To
safely support mapping the allocations to userspace major
architectures like x86 and arm have always zeroed allocations from
dma_alloc_coherent, but a couple other architectures were missing that
zeroing either always or in corner cases.
Then later we grew anothe dma_zalloc_coherent interface to explicitly
request zeroing, but that just added __GFP_ZERO to the allocation
flags, which for some allocators that didn't end up using the page
allocator ended up being a no-op and still not zeroing the
allocations.
So for this merge window I fixed up all remaining architectures to
zero the memory in dma_alloc_coherent, and made dma_zalloc_coherent a
no-op wrapper around dma_alloc_coherent, which fixes all of the above
issues.
dma_zalloc_coherent is now pointless and can go away, and Luis helped
me writing a cocchinelle script and patch series to kill it, which I
think we should apply now just after -rc1 to finally settle these
issue"
* tag 'remove-dma_zalloc_coherent-5.0' of git://git.infradead.org/users/hch/dma-mapping:
dma-mapping: remove dma_zalloc_coherent()
cross-tree: phase out dma_zalloc_coherent() on headers
cross-tree: phase out dma_zalloc_coherent()
soc_init_dai_link() calls soc_find_component() which needs
to be within client_mutex lock. Add client_mutex lock around
soc_init_dai_link() in snd_soc_register_card() to avoid
lockdep warning.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_byt_cht_es8316_mc_remove() use the platform drvdata as a type
of 'struct byt_cht_es8316_private', but snd_byt_cht_es8316_mc_probe()
set it to 'struct snd_soc_card', as suggested by Dan Carpenter, fix
the usage in snd_byt_cht_es8316_mc_remove().
Fixes: 0d3e91da07 ("ASoC: Intel: bytcht_es8316: Add external speaker mux support")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
move the codec PLL to rt5682_codec_init, because codec only need to config the clock source/PLL once.
As the result, remove the platform_clock_controls since no need to control clock anymore.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY.
MICBIAS voltage wasn't supplied to the microphone with the older
SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work.
This patch fixes the problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By making MCLK parent of DAI clocks, when querying the rate of the
clock the rate returned is now given from the parent clock so
gives the MCLK rate rather than 0 as previously returned. This is
a bit misleading, and actually there's no major reason why we can't
at least return the DAI WCLK rate, as set in HW, so that's what we
now do.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms using the Common Clock Framework to control the
codec's DAI clocks, MCLK should be enabled prior to DAI clocks
being turned on. For some platforms the codec is already
provided with an MCLK reference and can therefore control MCLK
itself as it needs to.
To improve functionality MCLK is now added as a parent to the
DAI clocks, if MCLK was provided, so that if they are enabled MCLK
will automatically be enabled as a prerequisite by the CCF.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI component probe is not called if it is not present
in component list during sound card registration.
Check if component is available in component list for
platform and cpu dai before soundcard registration.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable Headset Mic VREF for headset mode of ALC225.
This will be controlled by coef bits of headset mode functions.
[ Fixed a compile warning and code simplification -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to add unplug function to unplug state of headset mode
for ALC225.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix typo after a recent commit causing headphones to have no sound
Fixes: ad43d528a7 (ALSA: usb-audio: Define registers for CM6206)
Signed-off-by: Amadeusz Sławiński <amade@asmblr.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function rt274_jack_detect(), local variable "buf" could
be uninitialized if function regmap_read() returns -EINVAL.
However, it will be used to calculate "hp" and "mic" and
make their value unpredictable while those value are used
in the caller. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture through some of dmic we observe a glitch at the
start of record. This is because we start capturing even before
dmic is ready to send out data.
The optional delay will be applied after enabling the mic.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We already need to zero out memory for dma_alloc_coherent(), as such
using dma_zalloc_coherent() is superflous. Phase it out.
This change was generated with the following Coccinelle SmPL patch:
@ replace_dma_zalloc_coherent @
expression dev, size, data, handle, flags;
@@
-dma_zalloc_coherent(dev, size, handle, flags)
+dma_alloc_coherent(dev, size, handle, flags)
Suggested-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Luis Chamberlain <mcgrof@kernel.org>
[hch: re-ran the script on the latest tree]
Signed-off-by: Christoph Hellwig <hch@lst.de>
The "chip->dsp_spos_instance" can be NULL on some of the ealier error
paths in snd_cs46xx_create().
Reported-by: "Yavuz, Tuba" <tuba@ece.ufl.edu>
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a DMI quirk for the Point of View TAB-P1006W-232 (v1.0) tablet, this
tablet is special in a number of ways:
1) It uses the 2nd GPIO resource in the ACPI tables for jack-detect rather
then using the rt5651 codec's builtin jack-detect functionality
2) It uses the 3th GPIO resource in the ACPI tables to control the
external amplifier rather then the usual first non GpioInt resource and
the GPIO is active-low.
3) It is a BYTCR device, without a CHAN package and it uses SSP0-AIF1
rather then the default SSP0-AIF2.
4) Its internal mic is a digital mic (the first x86 rt5651 device that
I'm aware of which does this), combined with having its headset-mic
connected to IN2.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some board designs hook the jack-detect up to an external GPIO, rather
then to one of the codec pins, add support for this.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks module parameter to allow manually specifying quirks
from the kernel commandline (or modprobe.conf).
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map
headsetmic mapping"), changed the headsetmic mapping from IN3P to IN2P,
this was based on the observation that all bytcr_rt5651 devices I have
access to (7 devices) where all using IN3P for the headsetmic. This was
an attempt to unifify / simplify the mapping, but it was wrong.
None of those devices was actually using a digital internal mic. Now I've
access to a Point of View TAB-P1006W-232 (v1.0) tabler, which does use a
DMIC and it does have its headsetmic connected to IN2P, showing that the
original mapping was correct, so this commit reverts the change changing
the mapping back to IN2P.
Fixes: 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map ... mapping")
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some board designs hook the jack-detect up to an external GPIO,
rather then to one of the codec pins, add support for this.
Figuring out which GPIO to use is pretty much board specific so I've
chosen to let the machine driver pass the gpio_desc as data argument to
snd_soc_component_set_jack() rather then add support for getting the
GPIO to the codec driver. This keeps the codec code nice and clean.
Note that using an external GPIO for this conflicts with button-press
support, so this commit disables button-press support when an
external GPIO is used.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some BYT platforms have a RT5651 codec while using an ACPI node with
a HID of 10EC5640 to describe the coded. Add the 10EC5640 HID to the
acpi_device_id list, so that the rt5651 will bind to the codec on these
devices.
Like the rt5645 and rt5670 drivers which also have the 10EC5640 ACPI HID
in their acpi_device_id list for similar reasons, the rt5651 driver checks
the codecs device-id register so that it will only bind if the codec
actually is a rt5651 and it will ignore actual rt5640 codecs.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Point of View TAB-P1006W-232 (v1.0) tablet uses 10EC5640 as
ACPI HID, but it has a rt5651 codec add a quirk for this.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current BSDSR/BSDISR are using temporary/generic settings, but it can't
handle all SRCx/SoC. It needs to handle correctry.
Otherwise, sampling rate converted sound channel will be broken if it
was TDM. One note is that it needs to overwrite settings on E3 case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: chaoliang qin <chaoliang.qin.jg@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch introduces "sclk-strength" property to allow SCLK pad drive
strength to be changed via device tree.
When running playback test on LS1028ARDB, Tx Frame sync error interrupt
will occur sometimes. Some noises also exist. After changing SCLK pad
drive strength to the maximum value, the issues are gone.
Signed-off-by: Alison Wang <alison.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_lib_malloc_pages() may fail, so let's check its status and
return its error code upstream.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Added SPDIF driver build related changes.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added SPDIF audio driver. This provides playback and capture of
AES audio over SPDIF interface.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During the bootup of the kernel, the DAPM bias level is in the OFF
state. As soon as the DAPM framework kicks in it pushes the codec
into STANDBY state.
The probe function doesn't prepare the clock, and STANDBY state
does a clk_disable_unprepare() without checking the previous state.
This leads to an OOPS.
Not transitioning from an OFF state to the STANDBY state fixes the
problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add Digital Audio Interface driver that convers PDM bitstream to PCM
format.
Features:
- Fixed filtering characteristics for audio application.
- Full or partial set of channels operation with individual enable control.
- Programmable PDM clock generator.
- Programmable decimation rate.
- 16-bit signed output result.
- Overall stopband attenuation more than 80dB.
- Overall passband ripple less than 0.2dB.
Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit f84a6273dd ("ASoC: pxa: remove raumfeld machine driver")
removed the Raumfeld ASoC machine driver but forgot to kill one line
in the Makefile.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt298.c:992:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:995:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:317:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:320:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:348:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:351:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The buf in rl6347a_hw_read is __be32.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The eq parameters binary is stored in __be. However, it is unsigned short
in rt5645_eq_param_s{} which will cause incorrect type assignment. So add
struct rt5645_eq_param_s_be16{} to store the eq binary and convert it to
unsigned short in rt5645->eq_param.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse:
da7219.c:841:57: warning: dubious: x & !y
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse.
da7219.c:440:44: warning: cast to restricted __le16
da7219.c:461:13: warning: incorrect type in assignment (different base types)
da7219.c:461:13: expected unsigned short [unsigned] [usertype] val
da7219.c:461:13: got restricted __le16 [usertype] <noident>
da7219.c:1451:16: warning: incorrect type in assignment (different base types)
da7219.c:1451:16: expected unsigned short [unsigned] [usertype] offset
da7219.c:1451:16: got restricted __le16 [usertype] <noident>
da7219-aad.c:150:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:150:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:150:37: got restricted __le16 [usertype] <noident>
da7219-aad.c:157:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:157:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:157:37: got restricted __le16 [usertype] <noident>
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
mt6351.c:1418:5-8: Unneeded variable: "ret". Return "0" on line 1437
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/tscs42xx.c:392:5-31: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
nau8824.c:810:6-12: ERROR: Assignment of bool to non-0/1 constant
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5651.c:750:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:754:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:2192:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98927.c:508:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:889:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:891:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:893:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5640.c:980:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:984:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:2825:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt286.c:927:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:930:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:299:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:302:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt274.c:958:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:961:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:384:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:387:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/cs4271.c:226:2-16: WARNING: Assignment of bool to 0/1
sound/soc/codecs/cs4271.c:229:2-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98373.c:411:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:922:2-27: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:924:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some mux/mixer are not used. Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645_if3_adc_in_mux, rt5645_inr_mux, and rt5645_inl_mux are not used.
Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Missing or spurious parameter descriptions. Fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful this driver should be modified to expose
them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful then this driver should be modified to
expose them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No reason why this is global, fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AK4497 is a 32-bit 2ch DAC and has the same register
map as AK4458 with few exceptions:
* AK4497 has one more register at the end of register space
DFS_READ which is a read only register that allows users
to read FS Auto Detection mode. We currently do not use
this register so we use the same regmap structure as for ak4458.
* Because AK4458 is an 8ch DAC there are some fields that are
only used by AK4458 and marked as reserved for AK4497, so for
this reason we need to have a distinct set of controls, widgets
and routes.
Datasheet for AK4497 is at:
https://www.akm.com/akm/en/file/ev-board-manual/AK4497EQ.pdf
Datasheet for AK4458 is at:
https://www.akm.com/akm/en/file/datasheet/AK4458VN.pdf
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell has new platform for ALC274.
This will support to enable headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Reported-by: Hui Peng <benquike@163.com>
Tested-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had some sanity checks of the mixer unit descriptors but they
are too loose and some corner cases are overlooked. Add more strict
checks in uac_mixer_unit_get_channels() for avoiding possible OOB
accesses by malformed descriptors.
This also changes the semantics of uac_mixer_unit_get_channels()
slightly. Now it returns zero for the cases where the descriptor
lacks of bmControls instead of -EINVAL. Then the caller side skips
the mixer creation for such unit while it keeps parsing it.
This corresponds to the case like Maya44.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser for the processing unit reads bNrInPins field before the
bLength sanity check, which may lead to an out-of-bound access when a
malformed descriptor is given. Fix it by assignment after the bLength
check.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All McASP pin can be configured as GPIO.
Add gpiochip support for McASP and only enable it when the
gpio-controller is present in the DT node.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
McASP can loose it's context when runtime_pm is disabled.
Save and restore the context when suspending and resuming the device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the platform drivers are selected by the DAI drivers (including
McASP) there is no longer a need to check whether the modules are built-in
or module.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks to select the correct input map, jack-detect options
and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C).
Note: Although sound works out of the box, jack detection currently
requires overriding the ACPI DSDT table. This is necessary because
the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as
interrupt (one of the Bluetooth GPIOs).
The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the
Intel(R) Audio Machine Driver - AMCR0F28 device).
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices detected as BYT-T by the PMIC-type based detection
have only a single IRQ listed in the 80860F28 ACPI device. This
causes -ENXIO later when attempting to get the IRQ at index 5.
It turns out these devices behave more like BYT-CR devices,
and using the IRQ at index 0 makes sound work correctly.
This patch adds a fallback for these devices to is_byt_cr():
If there is no IRQ resource at index 5, treating the device
as BYT-T is guaranteed to fail later, so we can safely treat
these devices as BYT-CR without breaking any working device.
Link: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143176.html
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
is_byt_cr() and its usage can be simplified by returning the bool
directly, instead of through a pointer. This works because the
return value is just treated as bytcr = false and is not used
otherwise.
This patch also removes the extra check of
IS_ENABLED(CONFIG_IOSF_MBI) in favor of checking
iosf_mbi_available() directly. The header already takes care
of returning false if the config option is not enabled.
No functional change.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some BYTCR devices use an ES8316 codec, add an ACPI match table entry
for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on the input-map and on if 1 or 2 speakers are connected,
userspace needs to use a different UCM profile.
Since we already deal with quirks in the kernel driver and set the
input-map from the kernel, add a quirk for devices with a single / mono
speaker and set the card's long_name based on the input and speaker
quirks, so that userspace can use the long_name to pick the right UCM
profile.
This change, including how the long_name is build-up mirrors how we do
this in the bytcr_rt5640 and bytcr_rt5651 machine drivers.
Note since all devices I have access to use a mono speaker setup I've
chosen to default the speaker setting to mono.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After adding jack-detect support we have 3 microphone input switches:
"Microphone 1", "Microphone 2" and "Headset Mic". But the ES8316 has only
2 microphone inputs.
In the app-note explaining how to use the codec and on the 3 boards I
have one input is used for an internal microphone and one for the headset
microphone. On the 2 CHT boards I have the internal mic is on on MIC1 and
the headset mic is on MIC2, on the BYTCR board I have it is the other way
around.
This commit replaces the 2 "Microphone 1" and "Microphone 2" input switches
with a single "Internal Mic" switch and adds support for selecting either
possible input mapping.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ES8316 only has a single (amplified) output. The ES8316 appnote showing
the intended usage uses a jack-receptacle which physically disconnects the
speakers from the output when a jack is plugged in.
But all 3 devices using the es8316 which I have (2 Cherry Trail devices and
one Bay Trail CR device), use an analog mux to disconnect the speakers,
driven by a GPIO.
This commit adds support for this, modelling this as a separate speaker
widget / dapm pin-switch which sets the mux to drive the speakers when
selected.
The intend is for userspace to use the recently added jack-detect support
and then automatically select either the Headphone or Speaker output based
on that.
Note this commit includes a workaround for an ACPI table bug which is
present on 2 of the 3 devices I have, see the added comment in the code.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Hookup the jack-detect support added to the codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for having the codec connected to SSP0 instead of SSP2. This
is controlled through a new quirk parameter, similar to how this is done
in the bytcr_rt5640 and bytcr_rt5651 machine drivers.
Bay Trail CR (cost reduced) SoCs do not have an SSP2, so we default to SSP0
there.
Note the SPP0 quirk gets BIT(16) because bits 0-15 are reserved for non
boolean quirks like the input-map added in a later commit in this series.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some minor refactoring:
1) Group the code setting the card dev and prive pointers together with
registering the card
2) Properly put the comment about registering the card at the place where
we actually register the card and add a new comment for getting the clk
3) Add a struct device *dev helper variable (this will be used more in
follow up commits)
4) Reword error message to have the same "foo failed: %d" wording as others
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For lack of a better (non-random) way of sorting includes more and more
files in the kernel are moving over to sorting the includes alphabetically.
Move the bytcht_es8316 driver over to this sorting before we add a
bunch of more includes.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Export the DAC functionality to mix left + right together and then output
the same (mixed) signal on both outputs.
Various (x86) tablets with an ES8316 codec use a single speaker
connected between the headhpone LOUT and ROUT pins, expecting the output
to be in a mono differential mode. Presumably this is done to use the
power of both the left and right outputs to allow the speaker to be
louder.
The ES8316 codec does not have a differential output mode, but we can
emulate this by making both channels output the same through the mono mix
switch, combined with setting the Playback Polarity control to "R Invert",
which applias a 180 degrees phase inversion to the right channel.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding jack-detect support may seem weird for a codec with only
a single output, but it is necessary. The ES8316 appnote showing
the intended usage uses a jack-receptacle which physically disconnects
the speakers from the output when a jack is plugged in.
But all 3 devices using the es8316 which I have (2 Cherry Trail
devices and one Bay Trail CR device), use an analog mux to disconnect
the speakers, driven by a GPIO. In order to enable/disable the speakers
at the right time, we need jack-detect.
The same goes for the microphone where we must correctly set the mux
for the single ADC to either the internal or the headset microphone.
All devices I have support the es8316's builtin jack-detect functionality.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Inside function rt274_i2c_probe(), if regmap_read() function
returns -EINVAL, then local variable "val" leaves uninitialized
but used in if statement. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Nobody has actually used the type (VERIFY_READ vs VERIFY_WRITE) argument
of the user address range verification function since we got rid of the
old racy i386-only code to walk page tables by hand.
It existed because the original 80386 would not honor the write protect
bit when in kernel mode, so you had to do COW by hand before doing any
user access. But we haven't supported that in a long time, and these
days the 'type' argument is a purely historical artifact.
A discussion about extending 'user_access_begin()' to do the range
checking resulted this patch, because there is no way we're going to
move the old VERIFY_xyz interface to that model. And it's best done at
the end of the merge window when I've done most of my merges, so let's
just get this done once and for all.
This patch was mostly done with a sed-script, with manual fix-ups for
the cases that weren't of the trivial 'access_ok(VERIFY_xyz' form.
There were a couple of notable cases:
- csky still had the old "verify_area()" name as an alias.
- the iter_iov code had magical hardcoded knowledge of the actual
values of VERIFY_{READ,WRITE} (not that they mattered, since nothing
really used it)
- microblaze used the type argument for a debug printout
but other than those oddities this should be a total no-op patch.
I tried to fix up all architectures, did fairly extensive grepping for
access_ok() uses, and the changes are trivial, but I may have missed
something. Any missed conversion should be trivially fixable, though.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Among a few HD-audio fixes, the only significant one is the
regression fix on some machines like Dell XPS due to the default
binding changes. We ended up reverting the whole since the fix for
ASoC HD-audio driver won't be available immediately.
-----BEGIN PGP SIGNATURE-----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=93xe
-----END PGP SIGNATURE-----
Merge tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Among a few HD-audio fixes, the only significant one is the regression
fix on some machines like Dell XPS due to the default binding changes.
We ended up reverting the whole since the fix for ASoC HD-audio driver
won't be available immediately"
* tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Revert DSP detection on legacy HD-audio driver
ALSA: hda/tegra: clear pending irq handlers
ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, if platform_get_irq_byname() fails, the returned error
turns into a huge value, once it is being store into a variable
of type unsigned int, hence never actually reporting any error
and causing unexpected behavior when using the values stored
in aud_drv_data->s2mm_irq and aud_drv_data->mm2s_irq.
Fix this by changing the type of variables s2mm_irq and mm2s_irq in
structure xlnx_pcm_drv_data from unsigned int to int.
Addresses-Coverity-ID: 1476096 ("Unsigned compared against 0")
Fixes: 796175a94a7f ("ASoC: xlnx: add pcm formatter platform driver")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds reset and precharge in shutdown of PCM device.
ACODEC goes to silence if we change Fs to 44.1kHz from 48kHz. This
workaround seems to work but I don't know this workaround is correct
sequence or not for ACODEC.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for audio CODEC core of rk3328.
Rockchip does not publish detail specification of this core
but driver source code is opened on their GitHub repository.
https://github.com/rockchip-linux/kernel
So I ported this code to linux-next and added some trivial fixes.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is using asoc_simple_card_xxx() for
function / data naming. Because of this long prefix, it is easy to be
80 character over.
Let's reduce prefix from asoc_simple_card_xxx() to simple_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch cleanups the code by using asoc_simple_card_for_each_link()
which judges normal link / DPCM link.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch adds/modifies counting and parsing function for
"normal sound" and "DPCM sound", and call it from link loop.
This is prepare for cleanup DAI link loop method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
To preparing cleanup code, this patch adds link_info which handles
number of DAIs/Links/Codec Conf, and CPU/Codec turn.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= simple-scu-card) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged simple-card is completely forgeting about it.
This patch re-support it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= simple-scu-card) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged simple-card is completely forgeting about it.
To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(),
it need to judge whether it is DPCM by checking convert-rate/channel.
For this purpose, this patch adds asoc_simple_card_get_conversion()
as preparation
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is using asoc_graph_card_xxx() for
function / data naming. Because of this long prefix, it is easy to be
80 character over.
Let's reduce prefix from asoc_graph_card_xxx() to graph_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch cleanups the code by using asoc_graph_card_for_each_link()
which judges normal link / DPCM link.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch adds/modifies counting and parsing function for
"normal sound" and "DPCM sound", and call it from link loop.
This is prepare for cleanup DAI link loop method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
To preparing cleanup code, this patch adds link_info which handles
number of DAIs/Links/Codec Conf, and CPU/Codec turn.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= audio-graph-scu) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged audio-graph-card is completely forgeting about it.
This patch re-support it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio formatter PL IP supports DMA of two streams -
mm2s and s2mm for playback and capture respectively. Apart from
DMA, IP also does conversions like PCM to AES and viceversa.
This patch adds DMA component driver for the IP.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is already merged into simple-card.
simple-scu-card is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is already merged into audio-graph-card.
audio-graph-scu-card is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= audio-graph-scu) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged audio-graph-card is completely forgeting about it.
To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(),
it need to judge whether it is DPCM by checking convert-rate/channel.
For this purpose, this patch adds asoc_graph_card_get_conversion()
as preparation
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We will get DAI ID from "reg" property if it has on DT, otherwise get
it by counting port/endpoint.
But in below case, we need to get DAI ID = 0 via port reg = <0>, but
current implementation returns ID = 1, because it can't judge ID = 0 was
from "non reg" or "reg = <0>".
Thus, it will count port/endpoint number as "non reg" case.
of_graph_parse_endpoint() implementation itself is not a problem,
but because asoc_simple_card_get_dai_id() need to count port/endpoint
number when "non reg" case, it need to know ID = 0 was from
"non reg" or "reg = <0>".
This patch fix this issue.
port {
reg = <0>;
xxxx: endpoint@0 {
};
=> xxxx: endpoint@1 {
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix Sparse warnings with two machine drivers which weren't updated
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Detected with Coccinelle
skl-messages.c:419:5-32: WARNING: Comparison to bool
skl-pcm.c:1426:6-33: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Detected with Coccinelle
sound/soc/intel/skylake/skl-topology.c:3106:16-20: WARNING: casting
value returned by memory allocation function to (char *) is useless.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK input is needed when accessing any register after enabling SYSCLK.
This also fixes imbalance of clk_enable / clk_disable when transitioning
between ON -> STANDBY -> ON bias levels.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Save 2x unsigned int of .rodata.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms that use the audio-graph-card driver, the codec is
not selected by SoC-platform driver. Make it available.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Cirrus Logic CS4341.
This is a very simple, playback only, stereo DAC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC mixer setting needs to restore to default value
after calibration.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After 'commit 5d32a66541 ("PCI/ACPI: Allow ACPI to be built without
CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were
satisfied implicitly through dependencies on CONFIG_ACPI have to be
specified directly. This code relies on IOSF_MBI and IOSF_MBI depends
on PCI. For this reason, add a direct dependency on CONFIG_PCI to the
IOSF_MBI driver.
Fixes: 5d32a66541 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")
Signed-off-by: Sinan Kaya <okaya@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The problem is seen in the q6asm_dai_compr_set_params() function:
ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
(prtd->pcm_size / prtd->periods),
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
prtd->periods);
In this code prtd->pcm_size is the buffer_size and prtd->periods comes
from params->buffer.fragments. If we allow the number of fragments to
be zero then it results in a divide by zero bug. One possible fix would
be to use prtd->pcm_count directly instead of using the division to
re-calculate it. But I decided that it doesn't really make sense to
allow zero fragments.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can't return directly if snd_dma_alloc_pages() fails; we first need
to free prtd->audio_client and prtd.
Fixes: 22930c79ac ("ASoC: qdsp6: q6asm-dai: Add support to compress offload")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The q6asm_audio_client_alloc() doesn't return NULL, it returns error
pointers.
Fixes: 22930c79ac ("ASoC: qdsp6: q6asm-dai: Add support to compress offload")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The q6asm_fe_dais[] array has MAX_SESSIONS (8) elements so the >
comparison should be >= or we access one element beyond the end of the
array.
Fixes: 22930c79ac ("ASoC: qdsp6: q6asm-dai: Add support to compress offload")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We accidentally call mutex_unlock(&pcm512x->mutex); twice in a row.
I re-wrote the error handling to use "goto unlock;" instead of returning
directly. Hopefully, it makes the code a little simpler.
Fixes: 3500f1c589 ("ASoC: pcm512x: Implement the digital_mute interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviwed-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Changed License header from C to C++ style comment block.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason this field was set to zero when all other drivers use
.dynamic = 1 for front-ends. This change was tested on Dell XPS13 and
has no impact with the existing legacy driver. The SOF driver also works
with this change which enables it to override the fixed topology.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The intent was to print the address as a hexadecimal but there is an
extra "u" in the "0x%08ulx" format specification so it is displayed as
decimal.
Fixes: aef3b06ac6 ("[ALSA] SH7760 ASoC support")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Headset codec is connected over PRIMARY_MI2S interface. Call
set_jack for codec associated with Primary Mi2s interface.
Also, set_jack to NULL when jack is freed.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This essentially reverts the commits
c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present
and Skylake driver selected")
and
d82b51c855 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+
driver selection")
for the path of legacy HD-audio controller (snd-hda-intel).
The automatic DSP detection and skip of binding with the legacy driver
caused regressions on several machines like Dell XPS13. They give the
PCI class 0x40380 indicating the availability of DSP while they don't
work with ASoC SKL driver (yet).
As the support of ASoC driver for such devices isn't available, it's
better to revert the whole DSP-detection-and-skip behavior of the
legacy driver, so that we can get the old good driver working on such
devices.
The pci_binding option for ASoC SKL driver is still kept so that it
can work without blacklisting.
Fixes: c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Reported-by: Hans de Goede <hdegoede@redhat.com>
Reported-by: Azat Khuzhin <dohardgopro@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after disabling interrupts on the module, it could be possible
that irq handlers are still running. System hang is seen during
suspend path. It was found that, there were pending writes on the
HDA bus and clock was disabled by that time.
Above mentioned issue is fixed by clearing any pending irq handlers
before disabling clocks and returning from hda suspend.
Suggested-by: Mohan Kumar <mkumard@nvidia.com>
Suggested-by: Dara Ramesh <dramesh@nvidia.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose
CODEC is Realtek ALC294 has jack auto detection feature. This patch
enables the feature.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-----BEGIN PGP SIGNATURE-----
iHUEABYIAB0WIQRTLbB6QfY48x44uB6AXGG7T9hjvgUCXBvKlAAKCRCAXGG7T9hj
vmIoAP0XpLCE+0Z1hhxcDcJ0hKah1NIniRSIGGr6Af+gxe8F4wEA0Vm55gtEZerU
9mL5S7e2EcuTo93XCIjsxU8uPLGtegQ=
=59wi
-----END PGP SIGNATURE-----
Merge tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip
Pull xen updates from Juergen Gross:
"Xen features and fixes:
- a series to enable KVM guests to be booted by qemu via the Xen PVH
boot entry for speeding up KVM guest tests
- a series for a common driver to be used by Xen PV frontends (right
now drm and sound)
- two other fixes in Xen related code"
* tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip:
ALSA: xen-front: Use Xen common shared buffer implementation
drm/xen-front: Use Xen common shared buffer implementation
xen: Introduce shared buffer helpers for page directory...
xen/pciback: Check dev_data before using it
kprobes/x86/xen: blacklist non-attachable xen interrupt functions
KVM: x86: Allow Qemu/KVM to use PVH entry point
xen/pvh: Add memory map pointer to hvm_start_info struct
xen/pvh: Move Xen code for getting mem map via hcall out of common file
xen/pvh: Move Xen specific PVH VM initialization out of common file
xen/pvh: Create a new file for Xen specific PVH code
xen/pvh: Move PVH entry code out of Xen specific tree
xen/pvh: Split CONFIG_XEN_PVH into CONFIG_PVH and CONFIG_XEN_PVH
Pull sparc updates from David Miller:
- Automatic system call table generation, from Firoz Khan.
- Clean up accesses to the OF device names by using full_name instead
of path_component_name.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-next:
ALSA: sparc: Use of_node_name_eq for node name comparisons
sbus: Use of_node_name_eq for node name comparisons
sparc: generate uapi header and system call table files
sparc: add system call table generation support
sparc: add __NR_syscalls along with NR_syscalls
sparc: move __IGNORE* entries to non uapi header
sparc: Use DT node full_name instead of name for resources
sparc: Remove unused leon_trans_init
sparc: Use device_type helpers to access the node type
sparc: Use of_node_name_eq for node name comparisons
sparc: Convert to using %pOFn instead of device_node.name
sparc: prom: use property "name" directly to construct node names
of: Drop full path from full_name for PDT systems
sparc: Convert to using %pOF instead of full_name
fs/openpromfs: Use of_node_name_eq for node name comparisons
fs/openpromfs: use full_name instead of path_component_name
There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates.
A large diff pattern appears in ASoC TI part which now merges both
OMAP and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial
-----BEGIN PGP SIGNATURE-----
iQJCBAABCAAsFiEEIXTw5fNLNI7mMiVaLtJE4w1nLE8FAlwbbCIOHHRpd2FpQHN1
c2UuZGUACgkQLtJE4w1nLE+RoQ//TNKliUP3bOv4BdnSmUHcCSAP3st96Ror5lC+
RZ103UyjlCsfa7hSPfH7/4WHAjk7wXYdazjA3m2swsxsbcjPMW4uBIBJlegQNM/9
PmNt4y60UgdgMCT/uu10BlEO8GsqBkRpxFYHtUJ3Lq6h9ECa+VDLazNjK9jABItK
BVG668/LZp0le94cnJsLICmZ7fwKpAvi58hOsgOJLrPP4gzSGTj1gJXw/yyZ99QC
MPLVj1PruXq1l8zfxM+MUuOa7hayafx64bCbftlITPonWfEr7OvBCQ7Vf9HqzJIJ
OzEXAclVSc89R7RQT2omPiRKC7AyL85M9PCkpTtXh2D7DGXw9CFj5IDL6eIC4Ip4
sycArLOo6LA8ZYu45zCiY3rfh9Hx0Zn2qHz3qJeQtBrv5XYULuf6ZNwq6xJDEUz/
jxS558wSGHYWAyjv/IaPha1+JD6Us7zkSQgum+/qqnYYnYlDyxXnBatj4HzWZP/M
Khuhj+k+Y8UXK1MbRiekVCGIwCHGf3cUv4H7tq+qpEzvJZwTFnOBh7/twVL8gN96
x4ozZogaiaenZwlEZlkzowAMBBYMYb537Y+Y5suxzWGz3HN/Z8raaL8GTkIIB/BO
bNprJtPR8GJLfdu8NE++dUR2VuHfMnWat+frXk0eAsWutoviRyB4JLCRppfpgrA/
vDhIreU=
=SG61
-----END PGP SIGNATURE-----
Merge tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates. A
large diff pattern appears in ASoC TI part which now merges both OMAP
and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx
I2S controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial"
* tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection
ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected
ALSA: HDA: export process_unsol_events()
ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294
ALSA: bebob: fix model-id of unit for Apogee Ensemble
ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
ALSA: rme9652: Fix potential Spectre v1 vulnerability
ASoC: ti: Kconfig: Remove the deprecated options
ARM: davinci_all_defconfig: Update the audio options
ARM: omap1_defconfig: Do not select ASoC by default
ARM: omap2plus_defconfig: Update the audio options
ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options
ARM: OMAP2: Update for new MCBSP Kconfig option
ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option
MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support
ASoC: ti: Merge davinci and omap directories
ALSA: hda: add mute LED support for HP EliteBook 840 G4
ALSA: fireface: code refactoring to handle model-specific registers
ALSA: fireface: add support for packet streaming on Fireface 800
ALSA: fireface: allocate isochronous resources in mode-specific implementation
...
For HDaudio and Skylake drivers, add module parameter "pci_binding"
When pci_binding == 0 (AUTO), the PCI class/subclass info is used to
select drivers based on the presence of the DSP.
pci_binding == 1 (LEGACY) forces the use of the HDAudio legacy driver,
even if the DSP is present.
pci_binding == 2 (ASOC) forces the use of the ASOC driver. The
information on the DSP presence is bypassed.
The value for the module parameter needs to be identical for both
drivers. This parameter is intended as a back-up solution if the
automatic detection fails or when the DSP usage fails. Such cases
should be reported on the alsa-devel mailing list for analysis.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the SST/Skylake driver supports per platform selectors, we
can add logic to automatically select the right driver.
If the Skylake driver is selected for a specific platform, and the DSP
is detected at run-time based on the PCI class/subclass/prog-if
information, the legacy HDaudio driver aborts the probe. This will
result in a single driver probing and remove the need for modprobe
blacklists.
Follow-up patches will add a module parameter to bypass the logic if
this automatic detection fails, or if the Skylake driver is unable to
actually support the platform (firmware authentication, missing
topology file, hardware issue, etc).
The same mechanism will be used to conflicts generated by the same PCI
ID being registered by both legacy HDAuudio and SOF drivers for Intel
platforms. In other words SOF will not require changes to the HDaudio
legacy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOF implementation does not rely on the hdac_bus library, however
for HDMI and HDaudio codec support it does need to deal with
unsolicited events. Instead of re-inventing the wheel, export this
symbol to reuse this part of the library directly.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default, there is no sound on Asus UX391UA on Linux.
This patch adds sound support on Asus UX391UA. Tested working by three
different users.
The problem has also been described at
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485
Signed-off-by: Wandrille RONCE <w@ndrille.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ipcm->substream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->channel is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap)
Fix this by sanitizing info->channel before using it to index hdsp->channel_map
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
Also, notice that I refactored the code a bit in order to get rid of the
following checkpatch warning:
ERROR: do not use assignment in if condition
FILE: sound/pci/rme9652/hdsp.c:4103:
if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use page directory based shared buffer implementation
now available as common code for Xen frontend drivers.
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Boris Ostrovsky <boris.ostrovsky@oracle.com>
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlwY7KsTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0MFqB/4m9nlfUeXTpiSD1FgVIlMmdPNUg4V2
6Ybztaw4kRc5LuQN8PXQmaFLx020yAnLvI7Zzj7l3K8r6a9lfyFs+pKL0wtBnNK2
9QEFmOVQ3QFpt31Yb2IQeO4dfNbiyKeczjLau4mXWTl0j5dc/UH+HasE1dRZOxsC
rqJ8IsdibIVxVtQ7ZmcnU+y6XK0inBHAAh6ksMehsufShGrfrLs/nRBaXRZcRqJg
ciSFY5uYRYkDxTgogTpNRfVy4nr17N10+0sgrQ3RtaaqgG3gBXsHca1meyxKYW3Q
TssOJGIl3+uGiAMNyZqzxe5pAwwuGhZ3hAAAODtfYJQtAuAOW3/45Wqh
=phWF
-----END PGP SIGNATURE-----
Merge tag 'asoc-v4.21' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.21
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
We no longer have these options used anywhere.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Create new directory to contain all Texas Instruments specific DAI,
platform and machine drivers instead of scattering them under davinci and
omap directories.
There is already inter dependency between the two directories becasue of
McASP (on dra7x it is serviced by sDMA, not EDMA).
With the upcoming AM654 we will need to introduce new platform driver for
UDMA and it does not fit under davinci, nor under omap.
With the move I have restructured the Kconfig to be more usable in the era
of simple-sound-card:
CPU DAIs can be selected individually and they will select the platform
driver they can be served with.
To avoid breakage, I have moved over deprecated Kconfig options so
defconfig builds will work without regression.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
For sound/soc/{omap => ti}:
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tested with 4.19.9.
v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because
that's what the existing fixups for EliteBooks use.
Signed-off-by: Mantas Mikulėnas <grawity@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a result of investigation for Fireface 800, 'struct snd_ff_spec.regs'
is just for higher address to receive tx asynchronous packets of MIDI
messages, thus it can be simplified.
This commit simplifies it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to multiplex PCM frames into isochronous
packets and demultiplex PCM frames from isochronous packets for ALSA PCM
applications.
Fireface 800 voluntarily maintains resources for tx isochronous
communication. It performs reservation of isochronous channel and
allocation/update of bandwidth in some cases below:
- at a first request to allocation after bus resets
- at requests to allocation when further bandwidth is required
When request is grant and the unit is prepared, read data from
0x0000801c0008 represents isochronous channel for tx stream, then
the unit can handle requests to start communication. If driver
send the request without checking the register, the unit takes
panic to continue bus resets. The unit starts transmission of
tx packets after receiving several rx packets from driver.
I note that the unit can process tx/rx packets and generate/record
sound regardless of HOST LED.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The way to maintain isochronous resources on bus is different between
Fireface 400/800.
This commit is a preparation. This commit moves a function to allocate resource to
model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400/800 use three modes against the number of data channels in
data block for both tx/rx packets.
This commit adds refactoring for it. Some enumerators are added to
represent each of mode and a function is added to calculate the mode
from sampling frequency code (sfc).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both of Fireface 400/800 have the same register to switch frame fetching
mode regardless of difference of available number of PCM frames in
rx isochronous packet.
This commit moves a helper function from model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to my memo at hand and saved records, writing 0x00000001 to
SND_FF_REG_FETCH_PCM_FRAMES disables fetching PCM frames in corresponding
channel, however current implement uses reversed logic. This results in
muted volume in device side during playback.
This commit corrects the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e1 ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
uses different print formats for added tracepoints. However this is not
convenient for users/developers to prepare debug tools.
This commit uses the same format for the two tracepoints.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
introduces a wrong assignment to 'data_blocks' value of
'out_packet_without_header' tracepoint.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1/6 engine of ALSA firewire stack, a packet handler has a
second argument for 'the number of bytes in payload of isochronous
packet'. However, an incoming packet handler without CIP header uses the
value as 'the number of quadlets in the payload'. This brings userspace
applications to receive the number of PCM frames as four times against
real time.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 3b196c394d ('ALSA: firewire-lib: add no-header packet processing')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support to Display_port_rx mixers required to
select path between ASM stream and AFE ports.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support of AFE DAI for Display_port_rx port.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for Display_Port_Rx
port in AFE.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds MP3 playback support in q6asm dais, adding other codec
support should be pretty trivial.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to mp3 format in ASM module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Default copy function uses kmalloc to allocate buffers, lets check
if the runtime buffers are setup before making this allocations.
This can be useful if the buffers are dma buffers.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current SKYLAKE kconfig is a all-you-can-eat selection that will
support all known plaforms. This is however not necessarily a good
thing: most platforms for SKL and KBL don't support the DSP, but a
number of CNL/WHL ones do. Selecting this driver in all cases isn't
really smart and will require users to muck with blacklists.
Partition the configs to allow distributions to select on which
platform this driver is used. Keep the existing SND_SOC_INTEL_SKYLAKE
config to select everything for backwards compatibility. This patch does
not provide new functionality, only finer-grained choices in supported
platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,prefix = "xxx"; // initial
simple-audio-card,dai-link {
prefix = "xxx"; // overwrite
cpu {
...
};
codec {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "convert_rate/channel"
by many ways. But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,convert_channels = <xxx>; // initial
simple-audio-card,dai-link {
convert_channels = <xxx>; // overwrite
cpu {
convert_channels = <xxx>; // overwrite
};
codec {
convert_channels = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "mclk-fs" by many way.
But, it is not useful and readable.
We want to do is that allow having mclk-fs everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,mclk-fs = <xxx>; // for initial
simple-audio-card,dai-link {
mclk-fs = <xxx>; // overwrite
cpu {
mclk-fs = <xxx>; // overwrite
};
codec {
mclk-fs = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card and simple-scu-card are very similar driver,
but the former is supporting normal sound card,
the latter is supporting DPCM sound card.
We couldn't use normal sound and DPCM sound in same time by
one sound card. This patch merges both sound card into
simple-card. Now we can use both feature on same driver.
simple-card is now supporting .compatible = "simple-scu-audio-card".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
prefix = "xxx"; // initial
};
codec {
audio-graph-card,prefix = "xxx"; // overwrite
ports {
prefix = "xxx"; // overwrite
port {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "convert_rate/channel"
by many ways. But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
convert-channels = <xxx>; // initial
};
codec {
audio-graph-card,convert-channels = <xxx>; // overwrite
ports {
convert_channels = <xxx>; // overwrite
port {
convert_channels = <xxx>; // overwrite
endpoint {
convert_channels = <xxx>; // overwrite
};
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "mclk-fs" by many way.
But, it is not useful and readable.
We want to do is that allow having mclk-fs everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
mclk-fs = <xxx>; // initial
};
codec {
ports {
mclk-fs = <xxx>; // overwrite
port {
mclk-fs = <xxx>; // overwrite
endpoint {
mclk-fs = <xxx>; // overwrite
};
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card and audio-graph-scu-card are very similar driver,
but the former is supporting normal sound card,
the latter is supporting DPCM sound card.
We couldn't use normal sound and DPCM sound in same sound card by
audio-graph-card.
This patch merges both sound card into it.
Now we can use both feature on same driver.
audio-grap-card is now supporting .compatible = "audio-graph-scu-card".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit b6f3fc005a ("ASoC: simple-card-utils: fixup
asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method.
It will get DAI ID from OF graph "port", but, we want to consider about
"endpoint", too.
And, we also want to keep compatibility.
This patch fixup it as
if (driver has specified DAI ID)
use it as DAI ID
else if (OF graph endpoint has reg)
use it as DAI ID
else if (OF graph port has reg)
use it as DAI ID
else
use endpoint count as DAI ID
Fixes: commit b6f3fc005a ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting")
Reported-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove no_pcm check to invoke pcm_new() for backend dai-links
too. This fixes crash in hdmi codec driver during hdmi_codec_startup()
while accessing chmap_info struct. chmap_info struct memory is
allocated in pcm_new() of hdmi codec driver which is not invoked
in case of DPCM when hdmi codec driver is part of backend dai-link.
Below is the crash stack:
[ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018
..
[ 61.666696] CM = 0, WnR = 1
[ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000
[ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000
[ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP
[ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21
..
[ 61.740269] PC is at hdmi_codec_startup+0x124/0x164
[ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clicks and pops of various volumes can be produced while the device is
opened, closed, put into and taken out of standby, or reconfigured.
Fix this, by implementing the digital_mute interface, so that the
output is muted during such operations.
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Even if this spdif input driver is only supposed to be used on 64bits
platform, there is possible problem with 32bits and do_div, as reported
by the kbuild robot. Just fix it.
Fixes: 5ce5658375 ("ASoC: meson: add axg spdif input")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error logs to make probe debug easier.
Also remove hard-coded dependency on NHLT. NHLT literally stands for
NonHdaudioLinkTable and is only required for SSP/DMIC interfaces.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
bus->ppcap is now tested upfront, there is no need to re-check if the
hardware is exposed as needed. Remove tests and remove indentation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check immediately if required HDaudio capabilities can't be found (no
PPCAP or no streams exposed in GCAP), and move all DMA inits after the
error tests.
PPCAP and GCAP are not reliable indicators of DSP presence, but if
they don't exist then the driver will not work.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing PPCAP and GCAP fields cannot be used reliably to
determine if the DSP is enabled by the BIOS. Instead rely on the
class/subclass information to find out if this driver can run or
not. The values in the code don't seem to be documented in publicly
available documents but are part of recommendations made to BIOS
writers and have been verified to be accurate on a number of
platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's with CNP, supposed to be equivalent with CNL entry.
Keep the existing declaration style for now, at a later point we may
transition and use PCI_DEVICE_DATA().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S IP instance can work in transmitter/playback or receiver/capture mode
exclusively. The patch registers corresponding instance as ASoC component
with audio framework.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The new Dell IoT platform uses kabylake + alc3277 codec, and alc3277
shares the driver with the codec rt5660, here we generate a new
machine driver based on kbl_da7219_max98357a.
The audio design on this IoT platform is as below:
- Intel kabylake platform
- connect the codec ALC3277 via SSP0
- line-out and line-in with Micbias jacks
- line-out mute control and jack detection of line-out and line-in
- two HDMI ports with audio capability
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the spdif input decoder of the axg SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add IEC958_SUBFRAME_LE to the list of format accepted by the fifo frontend.
As opposed to what was initially noted in the toddr dai driver, the spdifin
does not place the msb at bit 28, it just output a whole spdif subframe.
Placing the msb at bit 28 in the toddr driver just filters out the parity,
user, channel status and validity bits. It is better to just provide the
whole spdif subframe to the userspace and let the iec958 plugin deal with
it.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
