Topology manifest v4 is still part of the ABI. Move its data structures
into the uapi header file.
No functional change.
Signed-off-by: Guenter Roeck <groeck@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit dc31e741db ("ASoC: topology: ABI - Add the types for BE
DAI") introduced sound topology files version 5. Initially, this
change made the topology code incompatible with v4 topology files.
Backwards compatibility with v4 configuration files was
subsequently added with commit 288b8da7e9 ("ASoC: topology:
Support topology file of ABI v4").
Unfortunately, backwards compatibility was never fully implemented.
First, the manifest size in (Skylake) v4 configuration files is set
to 0, which causes manifest_new_ver() to bail out with error messages
similar to the following.
snd_soc_skl 0000:00:1f.3: ASoC: invalid manifest size
snd_soc_skl 0000:00:1f.3: tplg component load failed-22
snd_soc_skl 0000:00:1f.3: Failed to init topology!
snd_soc_skl 0000:00:1f.3: ASoC: failed to probe component -22
skl_n88l25_m98357a skl_n88l25_m98357a: ASoC: failed to instantiate card -22
skl_n88l25_m98357a: probe of skl_n88l25_m98357a failed with error -22
After this problem is fixed, the following error message is seen instead.
snd_soc_skl 0000:00:1f.3: ASoC: old version of manifest
snd_soc_skl 0000:00:1f.3: Invalid descriptor token 1093938482
snd_soc_skl 0000:00:1f.3: ASoC: failed to load widget media0_in cpr 0
snd_soc_skl 0000:00:1f.3: tPlg component load failed-22
This message is seen because backwards compatibility for loading widgets
was never implemented.
The lack of audio support when running the upstream kernel on recent
Chromebooks has been reported in various forums, and can be traced back
to this problem. Attempts to fix the problem, usually by providing v5
configuration files, were only partially successful.
Let's implement backward compatibility properly to solve the problem
for good.
Signed-off-by: Guenter Roeck <groeck@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/pci/hda/local.h seems to be an earlier version of
sound/hda/local.h; it was added at the same time but doesn't seem to
have ever been used (within the git history). Most of its macros
depend on a hdac_read_parm() function which is not defined anywhere.
Signed-off-by: Ben Hutchings <ben.hutchings@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently there is no support for Tempo Semiconductor's TSCS454 CODEC.
Add support for it.
Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On APL, commit fd0f237572
("ASoC: Intel: bxt: Move codec sysclk config to codec_init function")
fixed an issue related to jack detection.
The MCLK for DA7219 does not change in this platform, but is
currently being configured everytime as part of the platform_clock
event handler for DAPM. The upshot of this is that we have
unnecessary calls to this function, and it also means that if
a stream hasn't yet been started, DA7219 driver does not have the
correct MCLK rates programmed and so the HP detection feature does
not operate as expected.
The same fix is needed on KBL.
This patch rectifies this issue by moving the sysclk call to
codec_init function so it's only called once at initialisation.
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>
The simple-card driver currently accepts a clock node in the cpu dai
sub-node and only uses it as an alternative to the
'system-clock-frequency' property to get the current frequency.
This patch adds another use of the passed clock node. If mclk-fs is
specified, the clocks in cpu and codec dai sub-nodes will be set to
the calculated rate (stream rate * mclk_fs) in hw_params.
This allows platforms to pass tuneable clocks as phandle that will
automatically be set to the right rates.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
When 'kzalloc()' fails in 'snd_hda_attach_pcm_stream()', a new pcm instance is
created without setting its operators via 'snd_pcm_set_ops()'. Following
operations on the new pcm instance can trigger kernel null pointer dereferences
and cause kernel oops.
This bug was found with my work on building a gray-box fault-injection tool for
linux-kernel-module binaries. A kernel null pointer dereference was confirmed
from line 'substream->ops->open()' in function 'snd_pcm_open_substream()' in
file 'sound/core/pcm_native.c'.
This patch fixes the bug by calling 'snd_device_free()' in the error handling
path of 'kzalloc()', which removes the new pcm instance from the snd card before
returns with an error code.
Signed-off-by: Bo Chen <chenbo@pdx.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
match_string() returns the index of an array for a matching string,
which can be used instead of open coded variant.
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
match_string() returns the index of an array for a matching string,
which can be used instead of open coded variant.
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.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
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
match_string() returns the index of an array for a matching string,
which can be used instead of open coded variant.
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
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
match_string() returns the index of an array for a matching string,
which can be used instead of open coded variant.
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
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Many X86 devices using a BYT SoC + RT5651 codec are cheap devices with
generic DMI strings, causing snd_soc_set_dmi_name() to fail to set a
long_name, making it impossible for userspace to have a correct UCM
profile which knowns which input is connected to the internal mic,
which input is connected to the hsmic (for correct jack-based switching)
and which inputs are unused.
Our quirks already specify which inputs the internal and headset mic
are connected to.
This commit sets a long_name based on the quirks so that userspace can
have UCM profiles doing the right thing based on the long_name.
Note that if we ever encounter the need for a special UCM profile for
some device we can add a quirk to set a specific long_name for the
device,
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add path for hosltess lpbk from ADDA Capture to ADDA Playback
add path for hostless phone call between ADDA DAI and PCM DAI
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The new mt6797-afe driver uses some functions in a common file, which
works for a built-in driver but fails for a loadable module:
ERROR: "mtk_afe_pcm_free" [sound/soc/mediatek/mt6797/snd-soc-mt6797-afe.ko] undefined!
ERROR: "mtk_afe_add_sub_dai_control" [sound/soc/mediatek/mt6797/snd-soc-mt6797-afe.ko] undefined!
ERROR: "mtk_afe_pcm_new" [sound/soc/mediatek/mt6797/snd-soc-mt6797-afe.ko] undefined!
ERROR: "mtk_afe_combine_sub_dai" [sound/soc/mediatek/mt6797/snd-soc-mt6797-afe.ko] undefined!
ERROR: "mtk_afe_pcm_ops" [sound/soc/mediatek/mt6797/snd-soc-mt6797-afe.ko] undefined!
This exports the five symbols above for modules.
Fixes: b3c702f56b ("ASoC: mt6797: combine DAI to register component")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_GPIOLIB is disabled, this codec fails to build
because gpio/consumer.h is not included implicitly.
sound/soc/codecs/pcm1789.c: In function 'pcm1789_common_init':
sound/soc/codecs/pcm1789.c:247:19: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_gpio_request_one'? [-Werror=implicit-function-declaration]
pcm1789->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_HIGH);
^~~~~~~~~~~~~~~~~~~~~~~
Fixes: 4ae340d1be ("ASoC: codecs: Add support for PCM1789")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
We want the loop to exit when "to" is set to zero, but in the current
code it's set to -1. Also I tweaked the indenting so it doesn't look
like we're passing "--to" to xenbus_read_unsigned().
Fixes: cc3196ae19 ("ALSA: xen-front: Introduce Xen para-virtualized sound frontend driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the call to hw_params for a component fails, the error code is held
by the variable '__ret' but the error message displays the value held by
the variable 'ret'. Fix the return code shown when hw_params fails for
a component.
Fixes: b8135864d4 ("ASoC: snd_soc_component_driver has snd_pcm_ops")
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have several Lenovo AIOs like M810z, M820z and M920z, they have
the same design for mic-mute hotkey and led and they use the same
codec with the same pin configuration, so use the pin conf table to
apply fix to all of them.
Fixes: 29693efcea ("ALSA: hda - Fix micmute hotkey problem for a lenovo AIO machine")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new helper file for these fixups due to requiring a huge number
of coefs being set to get the top speakers to work, as well as
setting pin 0x17 for the top speakers and the correct input source
of 0x17 for volume control
[ Note: this is a revised work based on Tom's fixup patch with the
replacement of the full COEF tables provided by Realtek.
Also, the fixup function has a proper HDA_FIXUP_ACT_* handling now.
The credit for the new COEF table goes to Kailang -- tiwai ]
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189331
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Tom Briden <tom@decompile.me.uk>
Tested-by: Tom Briden <tom@decompile.me.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Atmel I2S controller embedded into
sama5d2x SoCs.
Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds TX and RX TDM mixers for 40 TDM ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
All the mixer controls are pretty much same from all the afe ports.
Make these as proper macros for 2 reasons.
1> To avoid any typos in adding new mixer controls for each port.
2> Easy to edit from single place, easy to add new ports
This also prepares the routing driver to accomdate 40 tdm dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to 40 TDM ports supported in AFE.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use common q6afe_dai_prepare() for MI2S dais, this will remove
some code duplication. Also make the if statement to switch to
make the code look neater.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to tdm ports in AFE.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should set BE symmetric constraint on FE substream.
in case one BE is used by two FE1/FE2,
the first BE runtime will use FE1's substream->runtime.
hence the FE1's will be constrained by BE symmetry property.
Though, second FE2 call dpcm_apply_symmetry,
the be_substream->runtime == FE1's substream->runtime.
The FE2's substream->runtime will not be constrained
by BE's symmetry property.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case, one BE is used by two FE1/FE2
FE1--->BE-->
|
FE2----]
when FE1/FE2 call dpcm_be_dai_hw_free() together
the BE users will be 2 (> 1), hence cannot be hw_free
the be state will leave at, ex. SND_SOC_DPCM_STATE_STOP
later FE1/FE2 call dpcm_be_dai_shutdown(),
will be skip due to wrong state.
leaving the BE not being hw_free and shutdown.
The BE dai will be hw_free later when calling
dpcm_be_dai_shutdown() if still in invalid state.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
channel 1: SYSMEM<->ACP
channel 2: ACP<->I2S
Instead of waiting on period interrupt of ch 2 and then starting
dma on ch1, we make ch1 dma as circular.
This removes dependency of period granularity on hw pointer.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Tested-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the ARCHOS 80 Cesium 8" windows tablet, this device mostly
works with the default settings, except that it has only one speaker.
So add a quirk with the default settings + the mono-speaker flag.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds the mute LED control for HP Spectre x360 Kabylake
model. The mute LED is controlled via VREF bits on NID 0x1b, so we
need a new fixup function.
Note that this doesn't fix the other issues like the missing speaker
output on the machine. They will be addressed by later patches.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189331
Signed-off-by: Tom Briden <tom@decompile.me.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as
this serves merely as an intermediate buffer that is copied to each
URB transfer buffer. This works well in general on x86, but on some
archs this may result in cache coherency issues when mmap is used.
OTOH, it works also on such arch unless mmap is used.
This patch is a step for mitigating the inconvenience; a new module
option "use_vmalloc" is provided so that user can choose to allocate
the DMA coherent buffer instead of the existing vmalloc buffer.
The drawback is that it'd be the standard dma_alloc_coherent() calls
and the system would require contiguous pages on non-x86 archs.
Note that it's a global option and not dynamically switchable since
the buffer is pre-allocated at the probe time. In theory, it's
possible to be switchable, but it'd be trickier and racier.
As default use_vmalloc option is set to true, so that the old behavior
is kept. For allowing the coherent mmap on ARM or MIPS, pass
use_vmalloc=0 option explicitly.
Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing a humming sound when active on the Intel
NUC5i7RY, add it to the blacklist.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199607
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
gcc-8 warns that pcm_instance->name is not necessarily terminated correctly
if the input is more than 80 characters long or lacks a termination byte
itself:
In function 'strncpy',
inlined from 'cfg_device' at sound/xen/xen_snd_front_cfg.c:399:3,
inlined from 'xen_snd_front_cfg_card' at sound/xen/xen_snd_front_cfg.c:509:9:
include/linux/string.h:254:9: error: '__builtin_strncpy' specified bound 80 equals destination size [-Werror=stringop-truncation]
return __builtin_strncpy(p, q, size);
Using strlcpy() instead of strncpy() makes this a bit safer.
Fixes: fd3b36045c ("ALSA: xen-front: Read sound driver configuration from Xen store")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the device management by replacing the lowlevel device object
allocation with the card->private_data. Nowadays there is almost no
advantage by the lowlevel device, and with card->private_data, the
code becomes cleaner.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid if ((err = ...) style and expand to multiple lines instead.
No change in the end result, but just the beautification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that we can avoid the extra goto lines.
Also beautify the code to follow the standard codex.
No functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stream direction in open and close callbacks can be retrieved from
substream->direction, hence we don't have to stick with the unique PCM
ops hard-coded for each direction. Rewrite the common open/close
callback functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The negative error return from the call to to_sndif_format is being
assigned to an unsigned 8 bit integer and hence the check for a negative
value is always going to be false. Fix this by using ret as the error
return and hence the negative error can be detected and assign
the u8 sndif_format to ret if there is no error.
Detected by CoverityScan, CID#1469385 ("Unsigned compared against 0")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamoccchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error for a -ve value in ret is redundant as all previous
assignments to ret have an associated -ve check and hence it
is impossible for ret to be less that zero at the point of the
check. Remove this redundant error check.
Detected by CoveritScan, CID#1469407 ("Logically Dead code")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the S_<FOO> symbolic permissions to their octal equivalents as
using octal and not symbolic permissions is preferred by many as more
readable.
see: https://lkml.org/lkml/2016/8/2/1945
Done with automated conversion via:
$ ./scripts/checkpatch.pl -f --types=SYMBOLIC_PERMS --fix-inplace <files...>
Miscellanea:
o Wrapped one multi-line call to a single line
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At present, all of models produced by TC Electronic except for Konnekt Live
are supported with hard-coded their stream formats. Studio Konnekt 48 is
sore model to support dual streams for both directions. The second stream
has no MIDI conformant data channel in its data block. But current
implementation transfers the second stream with MIDI conformant data
channel.
This commit fixes this issue.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This series contains two omap1 ams-delta GPIO clean-up patches to get
started with removal of hard-coded GPIO numbers from drivers. And then
the removal of board mach includes from drivers. The second patch mostly
touches the ams-delta audio driver but is included here because of the
removal of the latch gpios and is acked by Mark Brown.
And there are two more am437x related PM patches to save and restore
control module and timer registers for RTC only suspend mode. Looks like
the patch title for the timer changes is a bit misleading, not all the
timer code is yet living under drivers/clocksource. But I had already
pushed out the branch before I noticed this.
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Merge tag 'omap-for-v4.18/soc-late-signed' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap into next/soc
Late omap soc changes for v4.18 merge window
This series contains two omap1 ams-delta GPIO clean-up patches to get
started with removal of hard-coded GPIO numbers from drivers. And then
the removal of board mach includes from drivers. The second patch mostly
touches the ams-delta audio driver but is included here because of the
removal of the latch gpios and is acked by Mark Brown.
And there are two more am437x related PM patches to save and restore
control module and timer registers for RTC only suspend mode. Looks like
the patch title for the timer changes is a bit misleading, not all the
timer code is yet living under drivers/clocksource. But I had already
pushed out the branch before I noticed this.
* tag 'omap-for-v4.18/soc-late-signed' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap:
OMAP: CLK: CLKSRC: Add suspend resume hooks
ARM: AM43XX: Add functions to save/restore am43xx control registers
ASoC: ams_delta: use GPIO lookup table
ARM: OMAP1: ams-delta: add GPIO lookup tables
Signed-off-by: Olof Johansson <olof@lixom.net>
In MediaTek SoC chip we have multiple DAI,
such as I2S, ADDA, PCM, etc.
Organize each DAI in to one sub dai,
with its dai driver, controls, widgets, routes.
add mtk_afe_combine_sub_dai() to combine
dai driver from each DAI.
add mtk_afe_add_sub_dai_control() to register
the control, widget, routes to component.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The newly introduced driver causes a harmless Kconfig warning when
compile-testing random configurations:
WARNING: unmet direct dependencies detected for SND_SDMA_SOC
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && DMA_OMAP [=n]
Selected by [y]:
- SND_OMAP_SOC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (ARCH_OMAP [=y] && DMA_OMAP [=n] || ARM [=y] && COMPILE_TEST [=y])
By simply allow build testing without DMA_OMAP, we can shut up that warning.
Fixes: dde637f2da ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before commit 3b5b899ca6 ("ALSA: hda: Make use of core codec functions
to sync power state"), hda_set_power_state() returned the response to
the Get Power State verb, a 32-bit unsigned integer whose expected value
is 0x233 after transitioning a codec to D3, and 0x0 after transitioning
it to D0.
The response value is significant because hda_codec_runtime_suspend()
does not clear the codec's bit in the codec_powered bitmask unless the
AC_PWRST_CLK_STOP_OK bit (0x200) is set in the response value. That in
turn prevents the HDA controller from runtime suspending because
azx_runtime_idle() checks that the codec_powered bitmask is zero.
Since commit 3b5b899ca6, hda_set_power_state() only returns 0x0 or
0x1, thereby breaking runtime PM for any HDA controller. That's because
an inline function introduced by the commit returns a bool instead of a
32-bit unsigned int. The change was likely erroneous and resulted from
copying and pasting snd_hda_check_power_state(), which is immediately
preceding the newly introduced inline function. Fix it.
Link: https://bugs.freedesktop.org/show_bug.cgi?id=106597
Fixes: 3b5b899ca6 ("ALSA: hda: Make use of core codec functions to sync power state")
Cc: Alex Deucher <alexander.deucher@amd.com>
Cc: Abhijeet Kumar <abhijeet.kumar@intel.com>
Reported-and-tested-by: Gunnar Krüger <taijian@posteo.de>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop pci_device() macro that just leads to chip->pci->dev, and pass it
directly to request_firmware(). It was introduced for allowing the
external alsa-driver kernel module builds. Since it was discontinued
years ago, we should clean it up now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the superfluous #ifndef checks that had been put just for
allowing building the alsa-driver kernel modules externally.
Since the external build was discontinued years ago, let's clean up
the old kludges.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now as the Amstrad Delta board provides GPIO lookup tables, switch from
GPIO numbers to GPIO descriptors and use the table to locate required
GPIO pins.
The card uses two pins, one for jack and the other for voice modem
codec DAI control.
For jack pin, remove hardcoded GPIO number and use GPIO descriptor
based variant of jack GPIO initialization.
For modem_codec pin, declare static variable for storing its GPIO
descriptor, obtain it on card initialization and replace obsolete
ams_delta_latch2_write() with gpiod_set_value(). For that to work,
don't request the modem_codec pin from the board init code anymore.
If the modem_codec GPIO lookup fails, skip initialization of
functionality of the card which depends on its availability.
Pin naming used by the driver should be followed while respective GPIO
lookup table is initialized by a board init code.
Created and tested against linux-4.17-rc3, on top of patch 1/6 "ARM:
OMAP1: ams-delta: add GPIO lookup tables"
Signed-off-by: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Tony Lindgren <tony@atomide.com>
Power-saving is causing plops on audio start/stop on ASRock H81M-HDS
machines, add these to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on Gigabyte
P55A-UD3 and Gigabyte Z87-D3HP machines, add these to the power_save
blacklist.
Note these 2 boards both use 1458:a002 as subsystem ids, so they share
a single entry.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing a plop and silences the first 2 seconds
(give or take) of audio, silencing notifications sounds on Medion /
Clevo W35xSS_370SS laptops.
Add the Clevo W35xSS_370SS to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1581607
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing a humming sound when active on the Intel
NUC7i3BNB, add it to the blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1520902
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The nau8824 codec can detect whether a headset or plain headphones is
inserted (as well as button presses on the headset) as such the jack_type
passed to snd_soc_card_jack_new() should include SND_JACK_MICROPHONE.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
some monitors care about the parity bit in the sub-frame of I2S,
but the cdn-dp always set this bit to "1", so these monitors
do not have sound output if use i2s, use spdif can fix this issue.
Signed-off-by: Chris Zhong <zyw@rock-chips.com>
Signed-off-by: Lin Huang <hl@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HP Z2 G4 requires the same workaround as other HP machines that have
no mic-pin detection.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FOr platforms that use the simple-card driver, the codec cannot be selected
through 'select' magic in Kconfig. So turn this into a real config option.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is needed when the codec is instanciated from from a device tree.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There's no need to read the register again prior to writing it, we did
that in the beginning of the function.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pxa-ssp driver currently assumes that .set_fmt() is called before
.set_clkdiv(), .set_pll() etc.
Commit a8bd0ee558 ("ASoC: raumfeld: Use static DAI format setup") broke
support for Raumfeld hardware (and possible other PXA based ones) because
it effectively changed the order of these calls. Also, as the call to
.set_fmt() is now done at probe time, the port clock is not yet enabled.
To fix this, strip all hardware register access code from the .set_fmt()
callback and memorize the desired value, so we can use it from the
.hw_params() callback. Also make the .set_fmt() callback less destructive
by reading all registers that it writes to in the beginning and only
masking out the bits that it possibly fiddles with.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
match_string() returns the index of an array for a matching string,
which can be used intead of open coded variant.
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support to DB820c machine driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6afe backend dais driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add support to MI2S mixers required to select path between
ASM stream and AFE ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to SLIMBus related mixers to control mux between
ASM stream and AFE port.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6 routing driver which configures route
between ASM and AFE module using ADM apis.
This driver uses dapm widgets to setup the matrix between AFE ports and
ASM streams.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to open, write and media format commands
in the q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to memory map and unmap regions commands in
q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds basic support to Q6 ASM (Audio Stream Manager) module on
Q6DSP. ASM supports up to 8 concurrent streams. each stream can be setup
as playback/capture. ASM provides top control functions like
Pause/flush/resume for playback and record. ASM can Create/destroy encoder,
decoder and also provides POPP dynamic services.
This patch adds support to basic features to allow hdmi playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to Q6ADM (Audio Device Manager) module in
q6dsp. ADM performs routing between audio streams and AFE ports.
It does Rate matching for streams going to devices driven by
different clocks, it handles volume ramping, Mixing with channel
and bit-width. ADM creates and destroys dynamic COPP services
for device-related audio processing as needed.
This patch adds basic support to ADM.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
With in ACP, There are three I2S controllers can be
configured/enabled ( I2S SP, I2S MICSP, I2S BT).
Default enabled I2S controller instance is I2S SP.
This patch provides required changes to support I2S BT
controller Instance.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
System clock on the platform is 25Mhz and not 24Mhz.
PLL_OUT for da7219 codec to use DA7219_PLL_FREQ_OUT_98304
as it is for 48KHz SR.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
hw_param can be called multiple times and thus we can have
more clk enable. The clk may not get diabled due to refcounting.
startup/shutdown ensures single clk enable/disable call.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
rtd structure freed early may result in kernel panic in dma close
call back. moved releasing memory for rtd structure to the end of
dma close callback.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added sram bank variable to audio_substream_data structure.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added pte offset variable in audio_substream_data structure.
Added Stoney related PTE offset macros in acp header file.
Modified hw_params callback to assign the pte offset value
based on asic_type.
PTE Offset macros used to calculate no of PTE entries
need to be programmed when memory allocated for audio buffer.
Depending upon allocated audio buffer size, PTE offset values
will change.
Compared to CZ, Stoney has SRAM memory limitation i.e 48k
It is required to define separate PTE Offset macros for
Stoney.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to LPASS Bit clock, LPASS Digital
core clock and OSR clock. These clocks are required for both
MI2S and PCM setup.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to 4 MI2S ports on LPASS.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to 6 SLIMBus AFE ports, which are used as
backend dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
AFE module sits right at the other end of cpu where the codec/audio
devices are connected.
AFE provides abstraced interfaces to both hardware and virtual devices.
Each AFE tx/rx port can be configured to connect to one of the hardware
devices like codec, hdmi, slimbus, i2s and so on. AFE services include
starting, stopping, and if needed, any configurations of the ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Log the correct error code in case the .open() call to a component fails.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistake in snprintf literal string
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HID is assumed to be made of TI PCI ID (0x104C) + part number, so all
four 104C5121, 104C5122, 104C5141 104C5142 are valid.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no platform code that uses i2c module table.
Remove it altogether and adjust ->probe() to be ->probe_new().
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The "entry" pointer is always non-NULL so this test for out of bounds
won't work.
Fixes: f1f0f330b1 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make C-header and SPDX-License-Identifier header uniform.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistakes in SND_SOC_BYTES literal strings
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
