IF YOU WOULD LIKE TO GET AN ACCOUNT, please write an
email to Administrator. User accounts are meant only to access repo
and report issues and/or generate pull requests.
This is a purpose-specific Git hosting for
BaseALT
projects. Thank you for your understanding!
Только зарегистрированные пользователи имеют доступ к сервису!
Для получения аккаунта, обратитесь к администратору.
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: b6f3fc005a2c ("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: 76fdb3a9e13a ('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: c4cfcf6f4297 ("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 648e921888ad ("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: 648e921888ad ("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>
