In a series of Fireface, later model supports different protocol
from former models.
This commit is a preparation to support both of protocols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unused variable was forgotten to be removed and now we get a
compiler warning:
sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend':
sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm'
Fixes: 17bc4815de ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system.
Because of this background, it is directly using
xxx_name / xxx_of_node / xxx_dai_name on dai_link.
Let's call it as legacy style here.
More complex style system like multi CPU multi Platform (= DMA) will
coming. To supporting it, we can use snd_soc_dai_link_component on
dai_link. Let's call it as modern style here.
But current ALSA SoC can't support it so far. Thus, we need to have
multi CPU / multi Codec / multi Platform style in the future on ALSA SoC.
Currently we already have multi Codec support. Platform is starting to
use modern style on dai_link, but still style only. Multi Platform is
not yet implemented. And we still don't have multi CPU support on ALSA
SoC, and not have modern style either.
Currently, if driver is using legacy style Codec/Platform, it will be
converted to modern style on soc-core. This means, we are using glue code
for legacy vs modern style so far on ALSA SoC.
We can fully switch to modern style on all drivers if ALSA SoC supported
modern style for CPU, and then, legacy style code will be removed from
ALSA SoC.
Untile then, we need to keep both legacy/modern style and its glue code.
This patch adds such future plan and background on soc.h
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlxBy/wTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0HNQB/wI7qx/bNGOX1p2C0M5ENdgsMcYDpMh
OHD073hL4wDkNJ/O2josnceCtrPUS5tuOxmx765IFXrAR4FWlNezQL4dHwTNGfG9
Femd6iAxv47lC2fROpuHfB0j32LIjVrHLYDG0wAiDvteXK2VrGbj8vedfWqflOBj
PX2kqkkgWfSCTrEYdrE09ExYoYKYdEqU/LEKFmIUnMuXc/HNHLo6e1sFNzTo8DIo
g5P8nQ//Qgi1U9UWMabKjy4lYL2Tcid9jcNlz0QIffK2qwIEpVBeuhqJaT4sQrf5
G51UrSk2y4lJ0t9WXv1SwQTDi2slCCGcOHA3fylGkJl1cxgOFMAR37Xs
=ev1L
-----END PGP SIGNATURE-----
gpgsig -----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlxCJZ0THGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0L+2B/0VnSutpVWaJnGyBuJ6zrydcSIW4183
G51jmioR10cl5LDV0DiI9l7IoiOwUyODbrIl/swoQPs7FWUsFRGFYytdmAoqaKIC
HK1j4D6Tlzac++e6bP6G6NzBMW6TGTu8c7hu3UtIGCz5uPRUKBthnndHmSbEB4h+
10N7RMs9+/BvH1Zt+x9VqEIP5OHpwc7rP/8yANYbQCY7CPehqDiGpE7SvUgFFl5t
IHx0nM1lJNeJyeu0Z/9BGAB5GWM/DzuMSrppNwl2k/QujSSyf1EPZvThpYNCBbnQ
6cQ51+7SZi06ejzMBB5h5z/9yFZTSKmTwFwzJwg/fJ/QngZVkksRRNLV
=9LPf
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v5.0-rc2' into asoc-5.1
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
When stopping audio, ASoC will first stop DMA then CPU DAI.
Sometimes there is a delay between DMA stop and CPU DAI stop, which
triggers an underrun error. Now, because of the delay introduced
by dev_err another underrun error will occur causing a vicious circle
making impossible to stop CPU DAI.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Deferred probes shouldn't cause error messages in the boot log, so
change the dev_err() to the more harmless dev_info().
Signed-off-by: Stefan Agner <stefan@agner.ch>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Removing the goto would not be strictly necessary. However, if
code gets added later, the cleanup in the EPROBE_DEFER case likely
would get missed.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Not finding the codec/SSI instance can be due to probe deferral.
Do not print error messages in those cases.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure to properly put the of node in case finding the codec
fails.
Fixes: 81e8e49261 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000")
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdmi-codec oopses the kernel when it is unbound from a successfully
bound audio subsystem, and is then rebound:
Unable to handle kernel NULL pointer dereference at virtual address 0000001c
pgd = ee3f0000
[0000001c] *pgd=3cc59831
Internal error: Oops: 817 [#1] PREEMPT ARM
Modules linked in: ext2 snd_soc_spdif_tx vmeta dove_thermal snd_soc_kirkwood ofpart marvell_cesa m25p80 orion_wdt mtd spi_nor des_generic gpio_ir_recv snd_soc_kirkwood_spdif bmm_dmabuf auth_rpcgss nfsd autofs4 etnaviv thermal_sys hwmon gpu_sched tda9950
CPU: 0 PID: 1005 Comm: bash Not tainted 4.20.0+ #1762
Hardware name: Marvell Dove (Cubox)
PC is at hdmi_dai_probe+0x68/0x80
LR is at find_held_lock+0x20/0x94
pc : [<c04c7de0>] lr : [<c0063bf4>] psr: 600f0013
sp : ee15bd28 ip : eebd8b1c fp : c093b488
r10: ee048000 r9 : eebdab18 r8 : ee048600
r7 : 00000001 r6 : 00000000 r5 : 00000000 r4 : ee82c100
r3 : 00000006 r2 : 00000001 r1 : c067e38c r0 : ee82c100
Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none[ 297.318599] Control: 10c5387d Table: 2e3f0019 DAC: 00000051
Process bash (pid: 1005, stack limit = 0xee15a248)
...
[<c04c7de0>] (hdmi_dai_probe) from [<c04b7060>] (soc_probe_dai.part.9+0x34/0x70)
[<c04b7060>] (soc_probe_dai.part.9) from [<c04b81a8>] (snd_soc_instantiate_card+0x734/0xc9c)
[<c04b81a8>] (snd_soc_instantiate_card) from [<c04b8b6c>] (snd_soc_add_component+0x29c/0x378)
[<c04b8b6c>] (snd_soc_add_component) from [<c04b8c8c>] (snd_soc_register_component+0x44/0x54)
[<c04b8c8c>] (snd_soc_register_component) from [<c04c64b4>] (devm_snd_soc_register_component+0x48/0x84)
[<c04c64b4>] (devm_snd_soc_register_component) from [<c04c7be8>] (hdmi_codec_probe+0x150/0x260)
[<c04c7be8>] (hdmi_codec_probe) from [<c0373124>] (platform_drv_probe+0x48/0x98)
This happens because hdmi_dai_probe() attempts to access the HDMI
codec private data, but this has not been assigned by hdmi_dai_probe()
before it calls devm_snd_soc_register_component(). Move the call to
dev_set_drvdata() before devm_snd_soc_register_component() to avoid
this oops.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlxBy/wTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0HNQB/wI7qx/bNGOX1p2C0M5ENdgsMcYDpMh
OHD073hL4wDkNJ/O2josnceCtrPUS5tuOxmx765IFXrAR4FWlNezQL4dHwTNGfG9
Femd6iAxv47lC2fROpuHfB0j32LIjVrHLYDG0wAiDvteXK2VrGbj8vedfWqflOBj
PX2kqkkgWfSCTrEYdrE09ExYoYKYdEqU/LEKFmIUnMuXc/HNHLo6e1sFNzTo8DIo
g5P8nQ//Qgi1U9UWMabKjy4lYL2Tcid9jcNlz0QIffK2qwIEpVBeuhqJaT4sQrf5
G51UrSk2y4lJ0t9WXv1SwQTDi2slCCGcOHA3fylGkJl1cxgOFMAR37Xs
=ev1L
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v5.0-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
Add a default pdata which can fit most HW design. So we don't need to
add a lot of DMI checking in this driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
snd_pcm_suspend() is no longer called from outside, so let's make it
local static. Also drop a superfluous NULL check there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATIIXP driver supports the full PCM resume and saves/restores the
running PCM pointer. This used to be done in the suspend and resume
callbacks together with snd_pcm_suspend() call. But since we moved
the snd_pcm_supsend*() call in PCM device PM ops, this should be moved
to a more appropriate place, i.e. the trigger callback.
Along with the movement of the PCM suspend/resume code, remove the
superfluous snd_pcm_suspend_all() call, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Until now we rely on each driver calling snd_pcm_suspend*() explicitly
at its own PM handling. However, this can be done far more easily by
setting the PM ops to each actual snd_pcm device object.
This patch adds the device_type object for PCM stream and assigns to
each PCM stream object. The type contains only the PM ops for system
suspend; we don't need to deal with the resume in general.
The suspend hook simply calls snd_pcm_suspend_all() for the given PCM
streams. This implies that the PM order is correctly put, i.e. PCM is
suspended before the main (or codec) driver, which should be true in
general. If a special ordering is needed, you'd need to adjust the
device PM order manually later.
This patch introduces a new flag, snd_pcm.no_device_suspend, too.
With this flag set, the PCM device object won't invoke
snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to
manage the PM call orders in its serialized way, and the flag is set
in soc_new_pcm() as default.
For the non-ASoC world, we can get rid of the manual snd_pcm_suspend
calls. This will be done in the later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple
sdo lines if supported. Audio needs to be striped across number of sdo
lines for simultaneous playbacks of higher resolutions to work.
This needs to be implemented only for an Audio Output Converter and only
if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter
is 1.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Platforms having multiple SORs and hdmi/dp sinks require higher
bandwidth to support simultaneous playbacks of higher resolution.
If hda controller supports multiple SDO lines, STRIPE can be used
to indicate how many of the SDO lines the stream should be striped
across.
During stream start stripe control bits are programmed to use given
number of sdo lines and the same is cleared during stream stop.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Controllers and codecs can support striping of audio out across
multiple SDO lines. The number of supported SDO lines can be
specific to chip. GCAP register can be read to know the maximum
supported SDO lines.
snd_hdac_get_stream_stripe_ctl() is exposed to program stripe bits
on controller and codec side.
stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines, etc.,
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/usb/mixer.c: In function 'parse_audio_feature_unit':
sound/usb/mixer.c:1838:28: warning:
variable 'first_ch_bits' set but not used [-Wunused-but-set-variable]
It never used since 2.6
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To enable SIE(Stream Interrupt Enable) in snd_hdac_stream_start(), we
should set both mask and value to be "1 << azx_dev->index" for register
update, the mask was 0, here fix it.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. for azx_int_enable(), we should set both mask and value to be
"AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable
controller CIE and GIE.
We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL,
here try to correct all of them.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
snd_ctl_add() could fail, so let's check its return value and return its
error code upstream upon failure.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix checks if snd_card_register() fails, and if so logs the error
via dev_err() consistent with other patches.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_i2c_sendbytes could fail. The fix checks its return value: if it
fails, issues an error message and returns with its error code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ctl_add() could fail, so let's check its status and issue an error
message if it indeed fails.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no reason for us to do that while we initialize dac_mute to
1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE
bit anyway.
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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-----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=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>
Commit 4fb7f4df49 ("ASoC: simple-card: use cpu/codec pointer on
simple_dai_props") updated {cpu,codec}_dai to be pointers in struct
simple_dai_props but didn't update these locations to dereference the
pointers.
This patch fixup it for non DT simple-card use case.
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
eukrea-tlv320.c machine driver runs on non-DT platforms
and include <asm/mach-types.h> header file in order to be able
to use some machine_is_eukrea_xxx() macros.
Building it for ARM64 causes the following build error:
sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory
Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320
driver when ARM64 is selected.
This is needed in preparation for the i.MX8M support.
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Shawn Guo <shawnguo@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of Huawei laptops come with a LED in the micmute key. This patch
enables the use of micmute LED for these devices:
1. Matebook X (19e5:3200), (19e5:3201)
2. Matebook X Pro (19e5:3204)
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch solves bug 200501 'Only 2 of 4 speakers playing sound.'
It enables the front speakers on Huawei Matebook X Pro laptops.
These laptops come with Dolby Atmos sound system and these pins
configuration enables the front speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200501
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
stream 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/core/pcm.c:140 snd_pcm_control_ioctl() warn: potential spectre issue 'pcm->streams' [r] (local cap)
Fix this by sanitizing stream before using it to index pcm->streams
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
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info.mode and info.port are indirectly controlled by user-space,
hence leading to a potential exploitation of the Spectre variant 1
vulnerability.
These issues were detected with the help of Smatch:
sound/synth/emux/emux_hwdep.c:72 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[i]->ctrls' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[info.port]->ctrls' [w] (local cap)
Fix this by sanitizing both info.mode and info.port before using them
to index emu->portptrs[i]->ctrls, emu->portptrs[info.port]->ctrls and
emu->portptrs.
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
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current rsnd is using RSND_REG_xxx for register naming,
and using RSND_REG_##f style macro for read/write.
The biggest reason why it uses this style is that
we can avoid non-existing register access.
But, its demerit is sequential register access code will
be very ugly.
Current rsnd driver is well tested, so, let's remove RSND_REG_
from rsnd_reg, and cleanup sequential register access code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fireface 800 is a flagship model of RME GmbH for audio and music units
on IEEE 1394 bus, shipped 2004. This model consists of four chips:
- TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus
- TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus
- Xilinx Spartan-3 FPGA XC3S400
- Xilinx High-Performance CPLD XC9572XL
This commit adds support Fireface 800. In this time, the support is
restricted to its MIDI functionality, thus this commit adds some
condition statements to avoid touching streaming functionality.
Unlike Fireface 400, Fireface 800 has no functionality to suppress
asynchronous transactions for MIDI messages except for unregister of
listen address in controller side, thus the feature is available as is.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Content of asynchronous transaction for MIDI messages differs between
Fireface 400 and 800.
This commit adds a model-specific handler for the transaction and adds
arrangement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400 and 800 have the same mechanism to decide address to which
asynchronous transactions are sent for MIDI messages, however they use
different registers for controllers to notify higher 4 byte of the
address.
This commit adds a model-specific parameter to represent the address.
Additionally, it corrects some comments. I note that these two models have
a difference to enable/disable the transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, a register
to receive asynchronous transactions for MIDI messages is the same. For
Fireface 800, minor register is used.
This commit declares macros for the transactions and obsoletes
model-specific parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unlike Fireface 400, Fireface 800 have two pair of optical interface
for ADAT signal and S/PDIF signal. ADAT signals for the interface
are handled for sampling clock source separately.
This commit modifies a parser for clock configuration to distinguish
these two ADAT signals.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for a register of clock configuration to
obsolete model-specific operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for the registers to obsolete model-specific
operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, status
registers for clock synchronization is common.
This commit moves some macros for them to header file.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-scu-card didn't care about codec_conf
for multi DPCM case. This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge simple-card and simple-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on simple-scu-card.
It is same logic with simple-card, thus easy merging.
This is prepare for merging simple card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-scu-card.c is supporting "convert-rate/channels" which is
used for DPCM.
But, sound card might have multi codecs, and each codec might need
each convert-rate/channels.
This patch supports each codec's convert-rate/channles support.
top node convert-rate/channels will overwrite settings if exist.
It can't support each codec's convert-rate/channels if sound card had
multi codecs without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links.
If sound card is caring only DPCM, link count = dai count,
but, if non DPCM case, link count != dai count.
Now, we want to merge simple-card and simple-scu-card,
then, we need to care both link / dai count more carefly
This patch cares it, and prepare for merging simple card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is supporting dai-link support, but simple-scu-card
doesn't have it.
This patch support it. This is prepare for merging simple-card
and simple-scu-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building without CONFIG_PCI, we can (depending on the architecture)
get a link failure:
ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined!
Adding a compile-time check for PCI gets it to work correctly on
32-bit ARM.
Fixes: d99501b857 ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've excluded the display_power_control flag for Intel HSW and BDW
codecs as the HD-audio controllers of the corresponding platforms take
care of the display power as well. But the recent refactoring
separates the controller and the codec power accounting, so it's fine
to call the display PM even for HSW/BDW codecs. This is less
confusing since we can avoid this well-hidden condition.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The display power is in unbalance at removing the driver since it
misses the snd_hdac_display_power(OFF) call.
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent refactoring, snd_hdac_display_power() doesn't return
any error, hence it can be defined to return void.
This makes many error checks redundant and allows us to reduce them
gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an error occurs in azx_probe_continue(), we should release the
display power. However, the current code ignores it and releases the
display power only for HSW/BDW cases. Fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hdac_display_power() can be called even for a HDA controller
without DRM binding. The same is true for other helpers,
snd_hdac_i915_set_bclk() and snd_hdac_set_codec_wakeup().
So all superfluous AZX_DCAPS_I915_POWERWELL checks in hda_intel.c can
be dropped, and the definition of AZX_DCAPS_I915_POWERWELL itself can
be removed as well. This simplifies the code a lot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current HD-audio code manages the DRM audio power via too complex
redirections, and this seems even still unbalanced in a corner case as
Intel DRM CI has been intermittently reporting. This patch is a big
surgery for addressing the complexity and the possible unbalance.
Basically the patch changes the display PM in the following ways:
- Both HD-audio controller and codec drivers call a single helper,
snd_hdac_display_power(). (Formerly, the display power control from
a codec was done indirectly via link_power bus ops.)
- snd_hdac_display_power() receives the codec address index. For
turning on/off from the controller, pass HDA_CODEC_IDX_CONTROLLER.
- snd_hdac_display_power() doesn't manage refcounts any longer, but
keeps the power status in bitmap. If any of controller or codecs is
turned on, the function updates the DRM power state via get_power()
or put_power().
Also this refactor allows us more cleanup:
- The link_power bus ops is dropped, so there is no longer indirect
management, as mentioned in the above.
- hdac_device link_power_control flag is moved to hda_codec
display_power_control flag, as it's only for HDA legacy.
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=106525
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-scu-card driver is parsing codec position for DPCM
and consider DAI format. But, current operation is doing totally pointless,
because it should be called for each CPU/Codec pair.
Let's tidyup asoc_simple_card_parse_daifmt() timing.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge simple-card and simple-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on simple-card.
It is same logic with simple-scu-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_OF is disabled, of_graph_parse_endpoint() does not
initialize 'info', and gcc can see that:
sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_card_parse_graph_dai':
sound/soc/generic/simple-card-utils.c:284:13: error: 'info.port' may be used uninitialized in this function [-Werror=maybe-uninitialized]
It's probably best to check the return code anyway, and that also
takes care of the warning.
Fixes: b6f3fc005a ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Calling into the codec driver adds a dependency on that being reachable
from the module:
ERROR: "rt5663_sel_asrc_clk_src" [sound/soc/qcom/snd-soc-sdm845.ko]
undefined!
Add the corresponding select statement, as it is done in the other user
(Intel).
Fixes: f7485875a687 ("ASoC: sdm845: Add configuration for headset codec")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
From the da7219 spec, the button A, B, C and D are remapped to
0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE,
B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND.
Signed-off-by: Zhuohao Lee <zhuohao@chromium.org>
Signed-off-by: Max Chang <changmax@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the Point of View Mobii TAB-P1005W-232 v2.0 tablet, this
BYTCR device uses IN1 for its MIC and JD2 for jack-detect, rather then the
default IN3 and JD1.
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 a quirk for the Prowise PT301 tablet, this BYTCR tablet has no CHAN
package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which
is the default for BYTCR devices.
Also it uses IN1 for its MIC and JD2 for jack-detect, rather then the
default IN3 and JD1.
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 ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC
and output through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
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>
The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
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>
The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack
sensing and enable use of the internal microphone on this laptop
X542UN. However, it's ALC294 so create a new fixup named
ALC294_FIXUP_ASUS_MIC to avoid confusion.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make unified suspend / resume helpers and call them from both the
runtime- and the system-PM callbacks for simplifying code.
There are slight changes of call orders, but there shouldn't be any
functional difference after refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In an initial commit, 'SYNC_STATUS' register is referred to get
clock configuration, however this is wrong, according to my local
note at hand for reverse-engineering about packet dump. It should
be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config()
is correctly programmed.
This commit fixes 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>
Users reported a mute LED regression on Lenovo X1 Carbon, the root
cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE
to this machine, then the machine can't apply the fixup of
ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup
together.
Fixes: c4cfcf6f42 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Driver rewritten, assign copyright notice and change module author
as original one remains silent and I want to be notified about bugs.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sound capture and line bypass currently do not work as well as
some mixer controls. Fix that by building proper audio paths and
adjusting volume controls to match datasheet.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop "Common NI Values Table" and calculate LRCLK divider, then
add allowed rate constraints based on master clock frequency.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement set_bias_level to drive shutdown bit, so device is
put to sleep when unused.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will enable headset button for new Chrome platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend some structs to add the support for jack button changes.
Now snd_hda_jack_add_kctl() receives two more arguments: the jack type
and the jack keymaps. Both are optional, and when zero are passed,
the function behaves just like before.
For reporting button state changes, you'd need to update
jack->button_state bits accordingly, typically in the jack callback.
Then the value OR'ed with button_state and the jack plug state is
passed to snd_jack_report().
Note that currently the code assumes only the one-shot button events,
i.e. it tries to send the button release soon after sending the button
event. If a driver really supports the button release handling by
itself, we may need to introduce some flag to control this behavior in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing the callee to evaluate the associated jack information
and the unsolicited event data, add the new fields to
hda_jack_callback. They can be used, for example, to retrieve the
headset button state in the callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If it plugged headphone or headset into the jack, then
do the reboot, it will have a chance to cause headphone no sound.
It just need to run the headphone mode procedure after boot time.
The issue will be fixed.
It also suitable for ALC234 ALC274 and ALC294.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek codec ALC3277 is 100% compatible with the codec RT5660
in I2S mode. And on the Dell IoT platform, the codec is ALC3277,
and the HID of the codec in the BIOS is 10EC3277, so adding this
ID to the ACPI match table.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
For the FSL ASoC card, the full node names appear to be "ssi", "esai",
and "sai", so there's not any reason to use strstr and of_node_name_eq
can be used instead.
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD platform device acp_audio_dma can only be created by parent PCI
device driver (drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c). Pass struct
device of the parent to snd_pcm_lib_preallocate_pages() so
dma_alloc_coherent() can use correct dma_ops. Otherwise, it will
use default dma_ops which is nommu_dma_ops on x86_64 even when
IOMMU is enabled and set to non passthrough mode.
Though platform device inherits some dma related fields during its
creation in mfd_add_device(), we can't simply pass its struct device
to snd_pcm_lib_preallocate_pages() because dma_ops is not among the
inherited fields. Even it were, drivers/iommu/amd_iommu.c would
ignore it because get_device_id() doesn't handle platform device.
This change shouldn't give us any trouble even struct device of the
parent becomes null or represents some non PCI device in the future,
because get_dma_ops() correctly handles null struct device or uses
the default dma_ops if struct device doesn't have it set.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Gnawty model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3
for the mclk, just like the Clapper and Swanky models.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case of
the Gnawty that was breaking audio support since we were not using the
right clock in the cht_bsw_max98090_ti machine driver.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201787
Cc: stable@vger.kernel.org
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-and-tested-by: Jaime Pérez <19.jaime.91@gmail.com>
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>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
A couple of open coded iterating thru the child node names are converted
to use for_each_child_of_node() instead.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert soundbus uevent and sysfs OF node name and device type usage to
use printf specifier and helper functions instead of directly accessing
the name and type pointers. This will allow the eventual removal of the
pointers.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
Cc: "David S. Miller" <davem@davemloft.net>
Cc: sparclinux@vger.kernel.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues
with the input from external microphone. The issue can be fixed by
the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer AIO Veriton Z4660G with ALC286 codec has issue with the input
from external microphones connecting via 'Front Mic' jack. The fixup
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of
the headset and fix the audio input issue of external microphone.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer AIO Aspire C24-860 with ALC286 can't detect the headset
microphone. Just like another Acer AIO U27-880, it needs a different
pin value for 0x18 and the headset fixup to make headset mic work.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic
and internal mic not working either. It needs the similar quirk like
Sony laptops to fix headphone jack sensing and enables use of the
internal microphone.
Unfortunately jack sensing for the headset mic is still not working.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch added max98373_reset function to avoid amp software reset failure and code duplication.
Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed.
Chip revision ID is available when the amp is in the idle state which means software reset is completed well.
Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge audio-graph-card and audio-graph-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on audio-graph-card.
It is same logic with audio-graph-scu-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-scu-card didn't care about codec_conf
for multi DPCM case. This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge audio-graph-card and audio-graph-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on audio-graph-scu-card.
It is same logic with audio-graph-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links.
If sound card is caring only DPCM, link count = dai count,
but, if non DPCM case, link count != dai count.
Now, we want to merge audio-graph-card and audio-graph-scu-card,
then, we need to care both link / dai count more carefly
This patch cares it, and prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_get_dai_id() returns DAI ID, but it is based on
DT node's "endpoint" position.
Almost all cases 1 port has 1 endpoint, thus, it was no problem.
But in reality, port : endpoint = 1 : N, thus, counting endpoint
is BUG, it should based on "port" ID.
This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit c16015f36c ("ASoC: rsnd: add .get_id/.get_id_sub")
add new .get_id/.get_id_sub to indicate module ID/subID.
It is used for SSIU and CTU. In SSIU case, subID indicates BUSIF,
but register settings is based on SSIU ID.
OTOH, in CTU case, subID indicates CTU channel, and register settings
is based on it. This means regmap read/write function needs to care it.
This patch fixup this issue. It can't play MIXed sound without this
patch.
Fixes: c16015f36c ("ASoC: rsnd: add .get_id/.get_id_sub")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For TDM debug purpose, indicating Channel and Mode is very
useful. This patch indicate it if it has #define DEBUG
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tegra186 and Tegra194 contain the same codecs as earlier chips and can
be supported using the same patch function.
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recent devices support more than the 4 codecs that the AZX core will
probe by default. Probe up to 8 codecs to make sure all of them are
enumerated.
Suggested-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Clapper model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3
for the mclk, just like the Swanky model.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case of
the Clapper that was breaking audio support since we were not using the
right clock in the cht_bsw_max98090_ti machine driver.
Cc: stable@vger.kernel.org
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
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>
These boards are now fully ported to devicetree and make use of the
simple-card driver, so the platform specific machine driver can be
removed.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49ab ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Entry needed for ICL RVP w/ RT274
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We've got a regression report for some Thinkpad models (at least
T570s) which shows the too low speaker output volume. The bisection
leaded to the commit 61fcf8ece9 ("ALSA: hda/realtek - Enable Thinkpad
Dock device for ALC298 platform"), and it's basically adding the two
pin configurations for the dock, and looks harmless.
The real culprit seems, though, that the DAC assignment for the
speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14
to be coupled with DAC NID 0x03. When more pins are configured by the
commit above, the auto-parser changes the DAC assignment, and this
resulted in the regression.
As a workaround, just provide the fixed pin / DAC mapping table for
this Thinkpad fixup function. It's no generic solution, but the
problem itself is pretty much device-specific, so must be good
enough.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304
Fixes: 61fcf8ece9 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a series of patches for conversion to LEDs audio-mute
trigger. It's based on 4.20-rc3 to be an immutable branch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Signed-off-by: Young_X <YangX92@hotmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's similar to other AMD audio devices, it also supports D3, which can
save some power drain.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 67ec1072b0 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM
stream") fixes deadlock for non-atomic PCM stream. But, This patch
causes antother stuck.
If writer is RT thread and reader is a normal thread, the reader
thread will be difficult to get scheduled. It may not give chance to
release readlocks and writer gets stuck for a long time if they are
pinned to single cpu.
The deadlock described in the previous commit is because the linux
rwsem queues like a FIFO. So, we might need non-FIFO writelock, not
non-block one.
My suggestion is that the writer gives reader a chance to be scheduled
by using the minimum msleep() instaed of spinning without blocking by
writer. Also, The *_nonblock may be changed to *_nonfifo appropriately
to this concept.
In terms of performance, when trylock is failed, this minimum periodic
msleep will have the same performance as the tick-based
schedule()/wake_up_q().
[ Although this has a fairly high performance penalty, the relevant
code path became already rare due to the previous commit ("ALSA:
pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now
this unconditional msleep appears only when using linked streams,
and this must be a rare case. So we accept this as a quick
workaround until finding a more suitable one -- tiwai ]
Fixes: 67ec1072b0 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream")
Suggested-by: Wonmin Jung <wonmin.jung@lge.com>
Signed-off-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the PCM core calls snd_pcm_unlink() always unconditionally
at closing a stream. However, since snd_pcm_unlink() invokes the
global rwsem down, the lock can be easily contended. More badly, when
a thread runs in a high priority RT-FIFO, it may stall at spinning.
Basically the call of snd_pcm_unlink() is required only for the linked
streams that are already rare occasion. For normal use cases, this
code path is fairly superfluous.
As an optimization (and also as a workaround for the RT problem
above in normal situations without linked streams), this patch adds a
check before calling snd_pcm_unlink() and calls it only when needed.
Reported-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default HDA sound card is registered with shortname "tegra-hda".
Same driver is used across tegra platforms and it is necessary to
distinguish between platforms to use platform specific settings from
userspace. One such example is, hdmi port on different platforms use
different alsa pcm device ID. For hdmi playback to work it should
open correct pcm device depending on the platform.
This patch applies shortname from first compatible string provided
in root node of device tree. Userspace then can use this card name
to apply specific settings.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-----BEGIN PGP SIGNATURE-----
iQFSBAABCAA8FiEEq68RxlopcLEwq+PEeb4+QwBBGIYFAlv7H/seHHRvcnZhbGRz
QGxpbnV4LWZvdW5kYXRpb24ub3JnAAoJEHm+PkMAQRiGhasH/AvNibkzTAPQR7jE
r/R7NIaFUdPpaHYpLpoYIdSWkqpeWtN3q4eldJKWcpFCaE83EzDVu1PRw79rt/lm
JIYbZ2SZWbTB7d2TLEiEF2bFXr+OF8MVc1tSAsP5xTMvV6mIz7+i2xr9R0D4YrKY
ejd+ExF3HimuRf557Uey1MGsRIhedHsHjHbWjvR1MuKzvUv9nAL+1E2KJbfPXacp
LGXElCYcPXoKw+GxMS7C28idgLoEJoS/eVieRiMGsGlgEe/MuaNMU+GRLMde9lI0
c/G/4uumJTHVY7npOjuNt1+2FoLVt8oLlhPcB06R7FIwJN+TupLQifqBkIe2saT6
R6ExonQ=
=+/RR
-----END PGP SIGNATURE-----
Merge v4.20-rc4 into drm-next
Requested by Boris Brezillon for some vc4 fixes that are needed for future vc4 work.
Signed-off-by: Dave Airlie <airlied@redhat.com>
Now all relevant platform drivers are providing the LED audio trigger,
we can switch the mute LED control with the LED trigger, finally.
For the mic-mute LED trigger, a common fixup function,
snd_hda_gen_fixup_micmute_led(), is provided to be called for the
corresponding quirk entries. This sets up the capture sync hook with
ledtrig_audio_set() call appropriately.
For the mute LED trigger, which is done currently only for
thinkpad_acpi, the call is replaced with ledtrig_audio_set() as well.
Overall, the beauty of the new implementation is that the whole ugly
bindings with request_symbol() are dropped, and also that it provides
more flexibility to users.
One potential behavior change by this patch is that the mute LED enum
may be created on machines that actually have no LED device. In the
former code, we did test-call and abort binding if the test failed.
But with the LED-trigger binding, this test isn't possible, and the
actual check is done in the LED class device side. So it's the
downside of simpleness.
Also, note that the HD-audio codec driver doesn't select CONFIG_LEDS
and co by itself. It's supposed to be selected by the platform
drivers instead.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introducing a module param for wakeup_delay in order to
align with modeswitch_delay parameter. With this change, both
wakeup_delay and modeswitch_delay parameters can be passed
as module parameters.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On startup, applications such as PulseAudio or CRAS enable playback or
capture on all PCM devices to verify that configurations are correct,
and close them immediately. For DMICs, this can result in the clock
being turned off very quickly, which may not compatible with internal
state machine transition requirements.
This patch add a mode-switch delay which will prevent the clock from
being turned off without complying with manufacturer timing
specifications. While the DMIC clock may be controlled at a lower level,
be it with hardware or firmware, applying the delay during the
STOP_TRIGGER phase ensures that there is no race condition, e.g. with
the hardware/firmware turning off the clock earlier
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a callback for init ops on dai_link to create and setup jack.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add board specific dapm widgets so these widgets can be used
in the route.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the dismod is specified in the DT node, use the specified custom value
to configure the drive on state of the inactive TX slots.
If the dismod is not present or booted in legacy mode, the dismod is set
to low as it was the original behavior.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When McASP is master and the PDIR for the clock pins are configured as
outputs before the clocking is configured it will output whatever clock
is generated at the moment internally.
The clock will switch to the correct rate only when the we start the clock
generators.
To avoid this we must only set the pin as output after the clock is
configured and enabled.
AXR pins configured as outputs behaves somehow interesting as well:
when McASP is not enabled and the pin is selected as output it will not
honor the DISMOD settings for the inactive state, but will pull the pin
down.
Add a new bitfield and mark the pins there which needs to be output and
set the pins only at the time when they will behave correctly.
On stream stop configure the pins back to input which makes them to obey
the global pin configuration regarding to pull up/down.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow the guideline from the TRM:
Before starting, clear the respective transmitter and receiver status
registers
To avoid stale state stored in the status registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If mclk/sclk is already running, FW responds with IPC reply MCLK/SCLK
already running. Add these to the IPC reply lookup table.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add more meaning to the IPC replies for easy debugging. Replace the switch
case with a lookup table to lookup for the IPC replies and print in human
readable form.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide
useful string for the vendor and product names too. In order to share
the UCM with WD15, here we keep the profile_name same as the WD15.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current rsnd dvc.c is using flags to avoid duplicating register for
MIXer case. OTOH, commit e894efef9a ("ASoC: core: add support to card
rebind") allows to rebind sound card without rebinding all drivers.
Because of above patch and dvc.c flags, it can't re-register kctrl if
only sound card was rebinded, because dvc is keeping old flags.
(Of course it will be no problem if rsnd driver also be rebinded,
but it is not purpose of above patch).
This patch checks current card registered kctrl when registering.
In MIXer case, it can avoid duplicate register if card already has same
kctrl. In rebind case, it can re-register kctrl because card registered
kctl had been removed when unbinding.
This patch is updated version of commit b918f1bc7f ("ASoC: rsnd: DVC
kctrl sets once")
Reported-by: Nguyen Viet Dung <dung.nguyen.aj@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Nguyen Viet Dung <dung.nguyen.aj@renesas.com>
Cc: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lots of fixes here, the majority of which are driver specific but
there's a couple of core things and one notable driver specific one:
- A core fix for a DAPM regression introduced during the component
refactoring, we'd lost the code that forced a reevaluation of the
DAPM graph after probe (which we suppress during init to save lots
of recalcuation) and have now restored it.
- A core fix for error handling using the newly added
for_each_rtd_codec_dai_rollback() macro.
- A fix for the names of widgets in the newly introduced pcm3060
driver, merged as a fix so we don't have a release with legacy names.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlv9WBQTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0KeSB/9mrVVRe8hVg2ErfCEY8BaPGIdtNoBT
IIRh/woXwbZEsY3owRuF5FeQxUbsRWLQL8fnzLFE7b6ltlRiPKahIGG0rIE0Xqcz
jkvqR0Kvck1MLNjj8qyd72Eqv0uQmW5qoezY9eFJiWms5uIlbh+WR3KPFu4WkcnD
9DHoYYq4ZU3EIqgt92QJ2uOWPwrSfueBcz6+bdYy11mSMuUtHz4HaAV8/XGFEjFy
9uPeNmS4nTFxGXR5PzpxTP1mG7nYixq3J07iwXGTH/X3LDANmJCPgMgQQRnY6h1k
KrtnZfcNljJxyYuNPV/A3ShG98Snbym17cq2zN1o9ZBUlPyxfGdRwrTd
=NoTQ
-----END PGP SIGNATURE-----
Merge tag 'asoc-v4.20-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.20
Lots of fixes here, the majority of which are driver specific but
there's a couple of core things and one notable driver specific one:
- A core fix for a DAPM regression introduced during the component
refactoring, we'd lost the code that forced a reevaluation of the
DAPM graph after probe (which we suppress during init to save lots
of recalcuation) and have now restored it.
- A core fix for error handling using the newly added
for_each_rtd_codec_dai_rollback() macro.
- A fix for the names of widgets in the newly introduced pcm3060
driver, merged as a fix so we don't have a release with legacy names.
This patch will enable ALC300.
[ It's almost equivalent with other ALC269-compatible ones, and
apparently has no loopback mixer -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device makes a loud buzzing sound when a headphone is inserted while
playing audio at full volume through the speaker.
Fixes: bbf8ff6b1d ("ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers")
Signed-off-by: Girija Kumar Kasinadhuni <gkumar@neverware.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The platform_device_register_full() function doesn't return NULL, it
returns error pointers.
Fixes: 7894a7e7ea ("ASoC: amd: create ACP3x PCM platform device")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have several Lenovo laptops with the codec alc285, when playing
sound via headphone, we can hear click/pop noise in the headphone,
if we let the headphone share the DAC of NID 0x2 with the speaker,
the noise disappears.
The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon.
I have tried to set preferred_dacs and override_conn, but neither of
them worked. Thanks for Kailang, he told me to invalidate the NID 0x3
through override_wcaps.
BugLink: https://bugs.launchpad.net/bugs/1805079
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both snd_ctl_add() and snd_ctl_replace() process the things in a
fairly similar way, and indeed the most of the codes can be unified.
This patch is a refactoring to consolidate the both functions to call
a single helper with an extra "mode" argument. There should be no
functional difference, except for one additional sanity check applied
now to snd_ctl_replace() (which was rather overlooking, IMO), too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for adding a user control element has some window opened
for race against the concurrent removal of a user element. This was
caught by syzkaller, hitting a KASAN use-after-free error.
This patch addresses the bug by wrapping the whole procedure to add a
user control element with the card->controls_rwsem, instead of only
around the increment of card->user_ctl_count.
This required a slight code refactoring, too. The function
snd_ctl_add() is split to two parts: a core function to add the
control element and a part calling it. The former is called from the
function for adding a user control element inside the controls_rwsem.
One change to be noted is that snd_ctl_notify() for adding a control
element gets called inside the controls_rwsem as well while it was
called outside the rwsem. But this should be OK, as snd_ctl_notify()
takes another (finer) rwlock instead of rwsem, and the call of
snd_ctl_notify() inside rwsem is already done in another code path.
Reported-by: syzbot+dc09047bce3820621ba2@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of sparc cs4231 driver code. Since
runtime->dma_area is managed by the PCM core helper, we shouldn't
release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of wss driver code. Since runtime->dma_area
is managed by the PCM core helper, we shouldn't release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the
MS-B120, in order for the headset mic to be properly detected.
They both use a single 3-way jack for both mic and headset with an
ALC283 codec, with the same pins used.
Cc: stable@vger.kernel.org
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on the built-in audio
of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to
the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Cc: <stable@vger.kernel.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_ac97_put_spsa() gets the bit shift value from the
associated private_value, but it extracts too much; the current code
extracts 8 bit values in bits 8-15, but this is a combination of two
nibbles (bits 8-11 and bits 12-15) for left and right shifts.
Due to the incorrect bits extraction, the actual shift may go beyond
the 32bit value, as spotted recently by UBSAN check:
UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7
shift exponent 68 is too large for 32-bit type 'int'
This patch fixes the shift value extraction by masking the properly
with 0x0f instead of 0xff.
Reported-and-tested-by: Meelis Roos <mroos@linux.ee>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, ALSA firewire-tascam driver queues events to notify
change of state of control surface to userspace via ALSA hwdep
interface.
This commit implements actual notification of the events. The events are
not governed by real time, thus no need to care underrun.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commits, ALSA firewire-tascam driver will allow userspace
applications to receive notifications about changes of device state,
transferred in tx isochronous packet. At present, all of drivers in ALSA
firewire stack have mechanism to notify change of status of packet
streaming, thus it needs to distinguish these two types of notification.
This commit is a preparation for the above.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Units of TASCAM FireWire series transfer image of states of the unit in
tx isochronous packets. Demultiplexing of the states from the packets
is done in software interrupt context regardless of any process context.
In a view of userspace applications, it needs to have notification
mechanism to catch change of the states.
This commit implements a queue to store events for the notification. The
image of states includes fluctuating data such as level of gain/volume
for physical input/output and position of knobs. Therefore the events
are queued corresponding to some control features only.
Furthermore, the queued events are planned to be consumed by userspace
applications via ALSA hwdep interface. This commit suppresses event
queueing when no applications open the hwdep interface.
However, the queue is maintained in an optimistic scenario, thus without
any care against overrrun. This is reasonable because target events are
useless just to handle PCM frames. It starts queueing when an usespace
application opens hwdep interface, thus it's expected to read the queued
events steadily.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a previous commit, ALSA firewire-tascam driver stores state image
from tx isochronous packets. This image includes states of knob, fader,
button of control surface, level of gain/volume of each physical
inputs/outputs, and so on. It's useful for userspace applications to
read whole of the image.
This commit adds a unique ioctl command for ALSA hwdep interface for the
purpose. For actual meaning of each bits in this image, please refer to
discussion in alsa-devel[1].
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140785.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Units of TASCAM FireWire series multiplex PCM frames and state of
control surface into the same tx isochronous packets. One isochronous
packet includes a part of the state in a quadlet data. An image of the
state consists of 64 quadlet data.
This commit demultiplexes the state from tx isochronous packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without under of overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The latency number is in usec for the pm_qos. Correct the calculation to
give us the time in usec
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit f986907c92 ("ASoC: audio-graph-card: add widgets and routing for
external amplifier support") added new function
asoc_graph_card_outdrv_event(), but the inserted position breaks
define area. This patch tidyup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
1 "simple" is enough on Kconfig help
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-scu-card driver is parsing codec position for DPCM
and consider DAI format. But, current operation is doing totally pointless,
because 1) asoc_simple_card_parse_daifmt() will be called not only for 1st
codec on current implementation, and it will be used as fixed format
2) it should be called for each CPU/Codec pair.
Let's tidyup asoc_simple_card_parse_daifmt() timing.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-scu-card.c is supporting "convert-rate/channels" which is
used for DPCM.
But, sound card might have multi codecs, and each codec might need
each convert-rate/channels.
This patch supports each codec's convert-rate/channles support.
top node convert-rate/channels will overwrite settings if exist.
It can't support each codec's convert-rate/channels if sound card had
multi codecs without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-scu-card.c is supporting "prefix" which is used to avoid
DAI naming conflict when CPU/Codec matching.
But, sound card might have multi sub-devices, and each codec might need
each prefix.
Now, ASoC is supporting snd_soc_of_parse_node_prefix(), let's support
it on audio-graph-scu-card, too. It is keeping existing DT style.
It can't support each codec's prefix if sound card had multi sub-devices
without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-scu-card.c is supporting "prefix" which is used to avoid
DAI naming conflict when CPU/Codec matching.
But, sound card might have multi sub-devices, and each codec might need
each prefix.
Now, ASoC is supporting snd_soc_of_parse_node_prefix(), let's support
it on audio-graph-scu-card, too. It is keeping existing DT style.
It can't support each codec's prefix if sound card had multi sub-devices
without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has snd_soc_of_parse_audio_prefix() to get codec_conf
settings from DT which is used to avoid DAI naming conflict when
CPU/Codec matching.
Currently, it is parsing from "top node",
but, we want to parse from "each sub node" if sound card had multi
cpus/codecs.
This patch adds new snd_soc_of_parse_node_prefix() to allow parsing
settings from selected node.
It is keeping existing snd_soc_of_parse_audio_prefix() by using macro.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amplifier may have assosicated regulator, so add a widget for it
and appropriate route.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the Allwinner A64 SoCs, the audio codec has a built-in headphone
amplifier. This amplifier has a power supply separate from the rest of
the analog audio circuitry, labeled cpvdd.
This patch adds a DAPM widget for this supply, and ties it to the
headphone amp widget.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_of_parse_routing() had "option" parameter
to consider error handling, but it is very pointless parameter.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils has asoc_simple_card_parse_convert() to parse
convert channel/rate for be_hw_params_fixup.
But, it is parsing from top of node.
If sound card had multi subnode, we need to parse it from each sub node.
This patch tidyup asoc_simple_card_parse_convert() to allow parsing
settings from each node.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If simple-card-utils accept NULL pointer on asoc_simple_card_xxx(),
each driver code will be more simple.
Let's accept NULL pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_simple_card_clk_register() is used but only 1 user,
and very pointless code. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ssi.c only is using rsnd_ssi_is_dma_mode().
Let's move it as static function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_parse_connect_ssiu_compatible() is doing
- using rsnd_ssiu_id(), but we use it via rsnd_mod_id()
- we can break loop if rsnd_dai_connect() was called
This patch fixup these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to configure bit clock for secondary MI2S
TX interface.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change slot_width for quaternary TDM port to 16 and
update bclk rate for TDM and MI2S interfaces
accordingly.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason we have different mechanisms for passing data to
machine drivers. Use the solution used by Atom/SST and SOF instead of
using drv_data as done by Skylake.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The changes for HDaudio overlooked the fact that the machine drivers
used for Chromebooks rely on the dmic number information passed as
pdata.
Add dmic_num field to standard interface and use standard interface
instead of SKL-specific one.
Also clean-up pdata definition to remove fields that are no longer
used.
Fixes: 842bb5135f ('ASoC: Intel: use standard interface for Hdaudio machine driver')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The machine_quirk may return NULL which means the acpi entries should be
skipped and search for next matched entry is needed, here add return
check here and continue for NULL case.
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: Mark Brown <broonie@kernel.org>
The Skylake driver currently has a set of problems supporting
load/unload modules. We need to make the HDaudio codec support
optional to help narrow down the issues.
Support for HDaudio codecs also leads to a Kconfig issue. We want the
hdac_hda codec to be compilable independently of Skylake (e.g. with
ALL_CODECS) but when Skylake is selected as built-in the hdac_hda
codec needs to use the same option due a a code dependency
Solve both problems by adding a user-selectable boolean Kconfig,
select HDAC_HDA as needed and make the HDaudio codec support in the
Skylake driver optional. Tests on a Chell Chromebook device without
HDaudio show no regression for speaker and HDMI playback.
This is submitted as an RFC to allow for comments and more validation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec can support any variation of bclk/fs master/slave configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove directly accessing device_node.type pointer and use the accessors
instead. This will eventually allow removing the type pointer.
Replace the open coded iterating over child nodes with
for_each_child_of_node() while we're here.
Cc: Johannes Berg <johannes@sipsolutions.net>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: linuxppc-dev@lists.ozlabs.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the pincfg assignment for the AE-5, which was
previously using the Recon3D pincfg's by mistake.
Fixes: d06feaf02f ("ALSA: hda/ca0132 - Add pincfg for AE-5")
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new PCI subsys ID for the ZxR, as found and tested by
other users. Without a way to know if any Z's use it as well, it keeps
the quirk of QUIRK_SBZ and goes through the HDA subsys test function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct declaration is not indented correctly. Fix this by replacing
spaces with a tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default implementation of regulator_set_load returns
REGULATOR_MODE_NORMAL, which is positive. [This was a bug which is
being fixed but the change is valid anyway -- bronie]
rt5663_i2c_probe should only do error handling when return value of
regulator_set_load is negative.
In this case, rt5663_i2c_probe should return error.
Also, consolidate err_irq into err_enable.
Fix the missing goto for temporary regmap and rt5663->regmap.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented too much by one level, fix this by
removing the extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented incorrectly. Fix this by adding in
the missing tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The goto statement is indented too much by one level, fix this by
removing the extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return statement is indented too much by one level, fix this by
removing an extraneous tab.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to remove the node name pointer from struct device_node,
convert printf users to use the %pOFn format specifier.
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Olivier Moysan <olivier.moysan@st.com>
Cc: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Cc: Maxime Coquelin <mcoquelin.stm32@gmail.com>
Cc: Alexandre Torgue <alexandre.torgue@st.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-stm32@st-md-mailman.stormreply.com
Cc: linux-arm-kernel@lists.infradead.org
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add regulator support to turn on cpvdd and avdd in probe.
If a regulator is not given from device tree, a dummy regulator will be
used.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AK4118A is a digital audio transceiver supporting 8 input channels
at 192kHz and with 24bits resolution.
It converts the S/PDIF signal to I2S format and is configurable over I2C.
This driver introduce a minimal support of the AK4118, like selecting the
input channel, reading input frequency and detecting some errors.
Datasheet is available here:
https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf
Signed-off-by: Adrien Charruel <adrien.charruel@devialet.com>
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the current device datasheet (TI Lit # SLAS831D, revised
March 2018) the value written to the device's PAGE register to trigger
a complete register reset should be 0xfe, not 0xff. So go ahead and
update to the correct value.
Reported-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Tested-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/amd/raven/acp3x-pcm-dma.c: In function 'acp3x_dma_hw_params':
sound/soc/amd/raven/acp3x-pcm-dma.c:333:25: warning:
variable 'dma_buffer' set but not used [-Wunused-but-set-variable]
It never used since introduction in commit
8de1b5ed03 ("ASoC: amd: add acp3x system resume pm op")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license
compliance management.
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixed build errors.
- Implicit declaration of pci_enable_msi() & pci_disable_msi()
api's for openrisc architecture.
- type defaults to 'int' in declaration of 'module_pci_driver'
Enabled build for x86 architecture.
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch updates license to SPDX-License-Identifier
instead of verbose license text.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio map are possible in wrong state before card->instantiated has
been set to true. Imaging the following examples:
time 1: at the beginning
in:-1 in:-1 in:-1 in:-1
out:-1 out:-1 out:-1 out:-1
SIGGEN A B Spk
time 2: after someone called snd_soc_dapm_new_widgets()
(e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c)
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN A B Spk
time 3: routes added
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN -----> A -----> B ---> Spk
In the end, the path should be powered on but it did not. At time 3,
"in" of SIGGEN and "out" of Spk did not propagate to their neighbors
because snd_soc_dapm_add_path() will not invalidate the paths if
the card has not instantiated (i.e. card->instantiated is false).
To correct the state of audio map, recalculate the whole map forcely.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The machine driver fails to probe in next-20181113 with:
[ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered
[ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517
...
[ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040
[ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040
[ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22
[ 3.715148] omap-abe-twl6040: probe of sound failed with error -22
Bisect pointed to a merge commit:
first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master'
and a diff between a working kernel does not reveal anything which would
explain the change in behavior.
Further investigation showed that on the second try of loading fails
because the dai_link->platform is no longer NULL and it might be pointing
to uninitialized memory.
The fix is to move the snd_soc_dai_link and snd_soc_card inside of the
abe_twl6040 struct, which is dynamically allocated every time the driver
probes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when snd_pcm_hw_constraint_integer fails there is
a memory leak of i2s_data on the error return path. Fix this by
kfree'ing i2s_data before returning.
Detected by CoverityScan, CID#1475479 ("Resource leak")
Fixes: 0b87d6bcd6 ("ASoC: amd: add acp3x pcm driver dma ops")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAC output may be differential (default) or single-ended.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the initial commit [1], I added differential output of the codec as
separate `+` and `-` widgets:
OUTL+
OUTR+
OUTL-
OUTR-
Later, in the commit [2], I added a device tree property to configure the
output as single-ended or differential. Having this property, the `+` and
`-` separation in widgets seems for me confusing. There are no functional
benefits in such separation, so I find reasonable to get rid of it:
OUTL
OUTR
The new naming is more friendly for sound cards, and is better aligned with
other codec drivers in kernel.
Renaming the output widgets now should not be a problem from the backwards-
compatibility perspective, as the driver for PCM3060 is added into the
mainline very recently, and did not yet appear in any releases.
[1] commit 6ee47d4a8d ("ASoC: pcm3060: Add codec driver")
[2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended
output")
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
sizeof() when applied to a pointer typed expression gives the
size of the pointer, not that of the pointed data.
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Frontend dai_link id is used for closing ADM sessions.
During concurrent usecase when one session is closed,
it closes other ADM session associated with other usecase
too. Dai_link->id should always point to Frontend dai id.
Set cpu_dai id as dai_link id to fix the issue.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it
to the driver to avoid adding it to every dts.
Fixes: eda85d1fee ("ASoC: sun8i-codec: Add ADC support for a33")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x drivers can be built by selecting necessary kernel config option.
The patch enables build support of the same.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When system wide suspend happens, ACP will be powered off.
When system resumes, all the runtime configuration data for
ACP needs to be programmed again.
Added 'resume'pm call back to ACP pm ops.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x I2S (CPU DAI) can act in normal I2S and TDM modes.
Added support for TDM mode.
Desired mode can be selected from ASoC machine driver.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x has a i2s controller block for playback and capture.
This patch adds ACP3x i2s DAI operations.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP3x has a DMA controller to access system memory.
This controller transfers data from/to system memory
to/from the ACP internal FIFO.
The patch adds PCM driver DMA operations.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Whenever audio data equal to the I2S FIFO watermark level are
produced/consumed, interrupt is generated.
Acknowledge the interrupt.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM platform driver binds to the platform device created by ACP3x PCI
device. PCM driver registers ALSA DMA and CPU DAI components with ASoC
framework.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP 3x IP has I2S controller device as one of IP blocks.
Create a platform device for it, so that the PCM platform driver
can be bound to this device. Pass PCI resources like MMIO, irq
to the platform device.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <vishnuvardhanrao.ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP 3.0 is a PCI audio device. This patch adds PCI driver to bind
to this device and get PCI resources.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Sanju R Mehta <sanju.mehta@amd.com>
Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP 3.x is a new audio block in raven. Added register header
of the same.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On certain platforms, Display HDMI HDA codec was not going to sleep state
after the use when links are powered down after turning off the display
power. As per the HW recommendation, links are powered down before turning
off the display power to ensure that the codec goes to sleep state.
This patch was updated from an earlier version submitted upstream [1]
which conflicted with the changes merged for HDaudio codec support
with the Intel DSP.
[1] https://patchwork.kernel.org/patch/10540213/
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Icelake device id. Also, Icelake's pin2port mapping table is
complicated. So we use a mapping table to do the pin2port mapping.
Signed-off-by: Bard liao <bard.liao@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Factor out the duplicated initialization statements from
wm_adsp1_init() and wm_adsp2_init() into new function
wm_adsp_common_init().
The entire content of wm_adsp1_init() is the common code
but it is convenient to retain this exported function
to hide what we currently treat as common init (which might
change in the future) and also make clear the difference
between an ADSP1 entry point and common code.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Stack memory isn't DMA-safe so it isn't safe to use either
regmap_raw_read or regmap_bulk_read to read into stack memory.
The two functions to read the scratch registers were using
stack memory and regmap_raw_read. It's not worth allocating
memory just for this trivial read, and it isn't time-critical.
A simple regmap_read for each register is sufficient.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds missing prepare_sleve_config that is needed for
setup the DMA slave channel for I2S.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initializing to -EINVAL is not correct as the variables are unsigned and
if buffer_size is 0 then they are not used anyway.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver will not probe if the pdata is not provided or created.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The mcbsp.c was copied a while back from arch/arm/plat-omap/mcbsp.c and it
contained a mix of McBSP and McBSP sidetone functions.
Create new file structure with the following split:
omap-mcbsp.c - McBSP related functions
omap-mcbsp-st.c - McBSP sidetone functionality
omap-mcbsp-priv.h - Private header for internal use
omap-mcbsp.h - Header for user drivers
I have tried to do the code move with minimal code change, cleanup patches
can be based on the new structure.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check if the McBSP have FIFO in the omap_mcbsp_set_threshold() and
omap_mcbsp_dai_delay() delay function to skip calling the lower layer if
it is not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We either start/stop TX or RX, never both. Move the tx/rx direction
selection within the functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The parameter name of dev_id is leftover from the old times when we passed
numeric ID as data for the interrupt handlers.
The mcbsp_rx and mcbsp_tx is misleading as they are pointers to the mcbsp
struct.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is configured runtime so no need to initialize it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the omap_mcbsp_dma_reg_params() a bit more intuitive to read for the
first glance by using SNDRV_PCM_STREAM_PLAYBACK/CAPTURE and to group the
outermost if case by stream direction.
While there, fix the outdated comment for the function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to call pci_iounmap() instead of iounmap() for the regions
obtained via pci_iomap() call for some archs that need special
treatment.
Fixes: aa31704fd8 ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for
the mute LED control over mic3 pin. Added the corresponding quirk
entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All the recent support of Creative boards and onboard audio depend on
PCI, but they can't be trimmed easily even if you build without
CONFIG_PCI, since the quirk is detected dynamically and the code has
many branches with the flag check like spec->quirk type or
spec->use_alt_functions.
This patch makes these checks static for CONFIG_PCI=n case so that the
compiler optimizes out. The access to flags are replaced with macros
that are replaced with a static value for CONFIG_PCI=n.
The macros look slightly ugly for avoiding compiler warnings wrt
unused variables, and some additional default-case handlings for
another compiler warnings, but the rest are very straightforward
changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit c0ea089dba ("ASoC: rsnd: rsnd_mod_name() handles both name and
ID") merged "name" and "ID" on rsnd_mod_name() to handle sub-ID
(= for CTU/BUSIF).
Then, it decided to share static char to avoid pointless memory.
But, it doesn't work correctry in below case, because last called
name will be used.
dev_xxx(dev, "%s is connected to %s\n",
rsnd_mod_name(mod_a), /* ssiu[00] */
rsnd_mod_name(mod_b)); /* ssi[0] */
->
rcar_sound ec500000.sound: ssi[0] is connected to ssi[0]
~~~~~~ ~~~~~~
We still don't want to have pointless memory, so let's use ring buffer.
16byte x 5 is very enough for this purpose.
dev_xxx(dev, "%s is connected to %s\n",
rsnd_mod_name(mod_a), /* ssiu[00] */
rsnd_mod_name(mod_b)); /* ssi[0] */
->
rcar_sound ec500000.sound: ssiu[00] is connected to ssi[0]
~~~~~~~~ ~~~~~~
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
drvdata is actually sun8i_codec, not snd_soc_card, so it crashes
when calling snd_soc_card_get_drvdata().
Drop card and scodec vars anyway since we don't need to
disable/unprepare clocks - it's already done by calling
runtime_suspend()
Drop clk_disable_unprepare() calls for the same reason.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM OSS layer may allocate a few temporary buffers, one for the core
read/write and another for the conversions via plugins. Currently
both are allocated via vmalloc(). But as the allocation size is
equivalent with the PCM period size, the required size might be quite
small, depending on the application.
This patch replaces these vmalloc() calls with kvzalloc() for covering
small period sizes better. Also, we use "z"-alloc variant here for
addressing the possible uninitialized access reported by syzkaller.
Reported-by: syzbot+1cb36954e127c98dd037@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gen2 has BUSIF0-3, Gen3 has BUSIF0-7 on some SSIU.
Current driver is assuming it is using BUSIF0 as default.
Thus, SSI is attaching SSIU (with BUSIF0) by using rsnd_ssiu_attach().
But, TDM split mode also needs other BUSIF to use it.
This patch adds missing SSIU BUSIFx support.
BUSIF is handled by SSIU instead of SSI anymore.
Thus, its settings no longer needed on SSI node on DT.
This patch removes its settings from Document, but driver is still
keeping compatibility. Thus, old DT style is still working.
But, to avoid confusing, it doesn't indicate old compatibility things on
Document. New SoC should have SSIU on DT from this patch.
1) old style DT is still supported (= no rcar_sound,ssiu node on DT)
2) If ssiu is not indicated on playback/capture,
BUSIF0 will be used as default
playback = <&ssi3>; /* ssiu30 will be selected */
3) you can select own ssiu
playback = <&ssi32 &ssi3>; /* ssiu32 will be selected */
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
converted rate/chan are handled each rated module, but
it will be used other module too.
For examle, converted channel is currently used for CTU,
but, it will be used for TDM Split mode, too.
This patch move/merge SRC/CTU hw_param under core.c
and handles converted rate/chan under rsnd_dai_stream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current driver is supporting HDMI output, and its information
are handled under ssi.c. But, it is stream information.
Let's move it from ssi.c to core.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rdai->playback/rdai->capture are defined as io_playback/io_capture
on __rsnd_dai_probe(). Let's use it instead of original one.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver has rsnd_runtime_is_ssi_xxx() functions,
but it is not only related to SSI, thus, it is misunderstandable.
This patch renames it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DTC commit df536831d02c ("checks: add graph binding checks")
is checking endpoint bidirectional, and it is upstreamed to linux by
commit 50aafd6089 ("scripts/dtc: Update to upstream version
v1.4.6-21-g84e414b0b5bc").
Let's remove own bidirectional check
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current driver is checking situation that can not happen.
This patch removes over-kill check
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DTC commit df536831d02c ("checks: add graph binding checks")
is checking endpoint bidirectional, and it is upstreamed to linux by
commit 50aafd6089 ("scripts/dtc: Update to upstream version
v1.4.6-21-g84e414b0b5bc").
Let's remove own bidirectional check
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DTC commit df536831d02c ("checks: add graph binding checks")
is checking endpoint bidirectional, and it is upstreamed to linux by
commit 50aafd6089 ("scripts/dtc: Update to upstream version
v1.4.6-21-g84e414b0b5bc").
Let's remove own bidirectional check
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason the RVP/LeafHill SSDT exposes an INT34C3 ID which is
used on other boards to point to the TDF8532 amplifier. Yay BIOS.
Add a DMI-quirk to ignore this ID and check for other valid machine
driver descriptors.
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add firmware/topology information for APL RVP
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Bard liao <bard.liao@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
GFP_ATOMIC is not required on any Intel drivers, use GFP_KERNEL
instead. A first cleanup was merged in April but missed a number
occurrences and new ones were added by copy/paste inertia.
While we are at it, make checkpatch happy with a sizeof(*msg)
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
s/skylaye/skylake
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't rely on internal Atom/SST-specific data structures, use
generic interface to let other drivers use the same machine drivers
as is, e.g. SOF to support BYT-CR devices
Tested-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't rely on internal Skylake-specific data structures, use
generic interface to let other drivers use the same machine driver
as is, e.g. SOF to support HDaudio codecs and HDMI outputs.
Tested on LeafHill CRB board, no regression seen with this change.
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Enabling both the old AC97_BUS code and the new AC97_BUS_COMPAT causes
problems because both modules provide an exported snd_ac97_reset()
function.
I had tried to fix the problem of having both coexist earlier, but
my patch only prevented them from being built-in. This is because
of a special Kconfig feature that lets a symbol have a dependency
on another one being disabled, but still allow both to be loadable
modules.
Changing the dependency to =n avoids that problem, now we can only
build the new driver if the old one is completely disabled.
If we could figure out a way to let rename one of the reset
functions and have each driver link to exactly the old or
the compat code, that would also work, but I could not find if
that's possible.
Fixes: bec5ecdf41 ("ASoC: pxa: avoid AC97_BUS build warning")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio support is done via simple-audio-card via DT, the custom driver
is no longer needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"DACn Zero Flag" and "ADCn Overflow Flag" are read-only status
register. These information are needed for driver not for user.
Let's remove these from snd_kcontrol_new.
There is no detail explanation for these register on
datasheet, but, judging from its behavior, it seems these register
needs clock to read. Otherwise, I2C returns error.
Therefor, amixer command will be failed under non working timing.
Without this patch, user often fail amixer command.
This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_component_update_bits() will only update the mask bits,
so remove the redundant snd_soc_component_read32().
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason the dapm widgets are incorrectly defined from the start,
Not sure how we ended up with such thing. Fix them now!
Without this fix the backend dais are always powered up even if there
is no active stream.
Reported-by: Jimmy Cheng-Yi Chiang <cychiang@google.com>
Reported-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
q6asm routing gets added multiple times as part of dai probe.
Move this to q6routing routes which has those widgets defined, this also
fixes the issue where these are added each time at dai probe.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The register map for CM6206 is known and we can define what
the values written to the different registers actually mean.
I tested to print the hex values before/after this change,
there is no functional change.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit c647f806b8 ("ALSA: hda - Allow multiple ADCs for
mic mute LED controls") we allow enabling the mic mute LED with
multiple ADCs. The commit changed the function return value to be
zero or a negative error, while this change was overlooked in the
thinkpad_acpi helper code where it still expects a positive return
value for success. This eventually leads to a NULL dereference on a
system that has only a mic mute LED.
This patch corrects the return value check in the corresponding code
as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621
Fixes: c647f806b8 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SD line mask for MI2S starts from BIT 0 instead of BIT 1.
Fix all bit mask for MI2S SD lines.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the
mclk instead of pmc_plt_clk_3.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case
of the Swanky that was breaking audio support since we were not using
the right clock in the cht_bsw_max98090_ti machine driver.
Cc: stable@vger.kernel.org
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-and-tested-by: Dean Wallace <duffydack73@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver is using "%s[%d]" for mod name and ID,
but, this ID portion might confusable.
For example currently, CTU ID is 0 to 7, but using 00 to 13
(= 00, 01, 02, 03, 10, 11, 12, 13) is very best matching to datasheet.
In the future, we will support BUSIFn, but it will be more complicated
numbering. To avoid future confusable code, this patch modify
rsnd_mod_name() to return understandable name.
To avoid using pointless memory, it uses static char and snprintf,
thus, rsnd_mod_name() user should use it immediately, and shouldn't keep
its pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ID for CTU and SSIU are confusable.
1 CTU has 4 sub nodes. This means, CTU0 has CTU01 - CTU03, CTU1 has
CTU10 - CTU13. SSIU is more confusable. Gen2 SSIU has BUSIF0-3, Gen3
SSIU has BUSIF0-7, but not for all SSIU.
In rsnd driver, each mod drivers are assuming rsnd_mod_id() returns
main device ID (In CTU case CTU0-1, SSIU case SSIU0-9), not serial
number.
This patch adds new .id/.id_sub to handling more detail ID.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each mod needs to have .get_status, but current driver is handling it
under rsnd_mod, instead of rsnd_mod_ops.
It is not any make sence. This patch moves it to rsnd_mod_ops, and
tidyup its parameter order to align to other callback functions.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
hw constraint for LEFT_J/I2S are same, but had duplicated code.
This code cleanup these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to
receive or transmit the most significant part of the 32bit sample.
We can not trick the system by using the DRR2/DXR2 register offset and use
32bit element size since these (and other McBSP) registers are 16bit ones.
We would need support for port_window in cyclic mode from the sDMA driver,
but it is not straight forward as we would need to have the sDMA frame to
cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA
period to be able to receive periodic interrupts.
Since the 32bit samples are not working (DMA timeout), just remove it in
case we have McBSP with reg_size of 2.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The regulator_desc structure can be const as it is only passed as the
second argument of devm_regulator_register and the corresponding
parameter is declared as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_dai_ops structures are only stored in the ops field of a
snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures
const as well.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_dai_ops structures are only stored in the ops field of a
snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures
const as well.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
WARN_ON() already contains an unlikely(), so it's not necessary to use
unlikely.
Signed-off-by: Yangtao Li <tiny.windzz@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Randconfig testing revealed a very old bug, with gcc-8:
sound/soc/intel/atom/sst/sst_loader.c: In function 'sst_load_fw':
sound/soc/intel/atom/sst/sst_loader.c:357:5: error: 'fw' may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (fw == NULL) {
^
sound/soc/intel/atom/sst/sst_loader.c:354:25: note: 'fw' was declared here
const struct firmware *fw;
We must check the return code of request_firmware() before we look at the
pointer result that may be uninitialized when the function fails.
Fixes: 9012c9544e ("ASoC: Intel: mrfld - Add DSP load and management")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
gcc notices that without either the ac97 bus or the pdata, we never
initialize the regmap pointer, which leads to an uninitialized variable
access:
sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe':
sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized]
Since that configuration is invalid, it's better to return an error
here. I tried to avoid adding complexity to the conditions, and turned
the #ifdef into a regular if(IS_ENABLED()) check for readability.
This in turn requires moving some header file declarations out of
an #ifdef.
The same code is used in three drivers, all of which I'm changing
the same way.
Fixes: 2ed1a8e0ce ("ASoC: wm9712: add ac97 new bus support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_ops structure can be const as it is only stored in the
ops field of a snd_soc_dai_link structure and this field is const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, the comparison of div < 0 is always false because div is
an unsigned int. Fix this by making div an int.
Detected by CoverityScan, CID#1475309 ("Unsigned compared against 0")
Fixes: 8307b2afd3 "(ASoC: stm32: sai: set sai as mclk clock provider")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover
of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace
it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some
configurations:
sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe':
sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init'
Fixes: 42371f327d ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack.
-----BEGIN PGP SIGNATURE-----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=lrUL
-----END PGP SIGNATURE-----
Merge tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack"
* tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
ALSA: dice: fix to wait for releases of all ALSA character devices
Ever since commit 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime
pm as "no callbacks"") the runtime suspend/resume hooks are no longer
used. Inline them into the system suspend hooks.
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20181024154825.18185-2-ville.syrjala@linux.intel.com
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Acked-by: Takashi Iwai <tiwai@suse.de>
Commit 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime pm as
"no callbacks"") broke runtime PM with lpe audio. We can no longer
runtime suspend the GPU since the sysfs power/control for the
lpe-audio device no longer exists and the device is considered
always active. We can fix this by not marking the device as
active.
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Fixes: 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"")
Signed-off-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20181024154825.18185-1-ville.syrjala@linux.intel.com
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Acked-by: Takashi Iwai <tiwai@suse.de>
commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set under
non-atomic") fixuped clock start timing. But it exchanged clock start
checker from ssi->usrcnt to ssi->rate.
Current rsnd_ssi_master_clk_start() is called from .prepare,
but some player (for example GStreamer) might calls it many times.
In such case, the checker might returns error even though it was not
error. It should check ssi->usrcnt instead of ssi->rate.
This patch fixup it. Without this patch, GStreamer can't switch
48kHz / 44.1kHz.
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In a former commit, PCM constraint based on LCM of SYT_INTERVAL was
obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in
some cases that max/min values of size of buffer/period is not
multiples of one of values of SYT_INTERVAL. For example, pulseaudio
always fail to configure PCM substream.
This commit changes strategy for the PCM rule. Although the buggy rules
had a single dependency (rate from period, period from rate, rate from
buffer, buffer from rate), a revised rule has double dependencies
(period from period/rate, buffer from buffer/rate). A step of value is
calculated with table of SYT_INTERVAL and list of available rates. This
prevents interval template which brings -EINVAL to a call of
snd_interval_refine().
Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Creative Audigy SE (SB0570) card currently exhibits an audible pop
whenever playback is stopped or resumed, or during silent periods of an
audio stream. Initialise the IZD bit to the 0 to eliminate these pops.
The Infinite Zero Detection (IZD) feature on the DAC causes the output
to be shunted to Vcap after 2048 samples of silence. This discharges the
AC coupling capacitor through the output and causes the aforementioned
pop/click noise.
The behaviour of the IZD bit is described on page 15 of the WM8768GEDS
datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples
will cause all outputs to be connected directly to VCAP. This also
happens if 2048 consecutive zero input samples are applied to all 6
channels, and IZD=0. It will be removed as soon as any channel receives
a non-zero input". I believe the second sentence might be referring to
IZD=1 instead of IZD=0 given the observed behaviour of the card.
This change should make the DAC initialisation consistent with
Creative's Windows driver, as this popping persists when initialising
the card in Linux and soft rebooting into Windows, but is not present on
a cold boot to Windows.
Signed-off-by: Alex Stanoev <alex@astanoev.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a development period for Linux kernel v4.20, drivers in ALSA firewire
stack were changed to wait for releases of all ALSA character devices at
.remove callback of bus driver. However, ALSA dice driver is partly out
of this change. This bug can bring fault to user process which holds
the last of character device in unplugging.
This commit fixes the driver to wait in the callback.
Fixes: 61ccc6f6b2 ('ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates.
Here are highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060
-----BEGIN PGP SIGNATURE-----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=3LrT
-----END PGP SIGNATURE-----
Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates. Here are
highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060"
* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
ASoC: stm32: sai: fix master clock naming
ASoC: stm32: add clock dependency for sai
ALSA: hda/ca0132 - Actually fix microphone issue
ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
ASoC: wm2000: Remove wm2000_read helper function
ASoC: cs42l51: fix mclk support
ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
ASoC: cs42l51: add mclk support
ASoC: stm32: sai: set sai as mclk clock provider
ASoC: dt-bindings: add mclk support to cs42l51
ASoC: dt-bindings: add mclk provider support to stm32 sai
ASoC: soc-core: fix trivial checkpatch issues
ASoC: dapm: Add support for hw_free on CODEC to CODEC links
ASoC: Intel: kbl_da7219_max98927: minor white space clean up
ALSA: i2c/cs8427: Fix int to char conversion
ALSA: doc: Brush up the old writing-an-alsa-driver
ASoC: rsnd: tidyup SSICR::SWSP for TDM
ASoC: rsnd: enable TDM settings for SSI parent
ASoC: pcm3168a: add hw constraint for capture channel
...
sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer
sizeof when applied to a pointer typed expression gives the size of
the pointer
Generated by: scripts/coccinelle/misc/noderef.cocci
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
CC: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A couple of fixes for build bot issues in the STM32 SAI driver.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAlvPB6ETHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0AolB/9nKMIGJqCZWWLh+Rq1dAc1k3+cOkuU
4Culg5XaTn9uS9Tu5DJKotk8SC4zvblbC4JiQPVisuUm1EY/ADwsu92DeC3xB3w4
iEyep1H1UzM8um/uuPe0lMWalNBFCsNRih0ZETxLh1ZK2hJlUtCYHz0n5ewmq/6R
QVdkAtm8oMll1LhTu9x95+ozYp33IxFpDVjSDXDZcWbgg93wBZ6dUMVLeREOK+Vl
h2H4TVgN/frJyBdKDBItCLVyL9e5ygw6UioEi4vgy08q1Kw01RKmRA86dScgBXaY
NVCi8OAVqQKoO0j5ChISnJPX3SmVV/5f6K+0x1to5bqfGOwOmoqCbOSB
=jBs9
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.0-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional fixes for the next release
A couple of fixes for build bot issues in the STM32 SAI driver.
Takashi Sakamoto