Since the addition of commit 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow
playback/capture if supported"), meson-axg cards which have codec-to-codec
links fail to init and Oops:
Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
Internal error: Oops: 96000044 [#1] PREEMPT SMP
CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
pc : invalidate_paths_ep+0x30/0xe0
lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
Call trace:
invalidate_paths_ep+0x30/0xe0
snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
dpcm_path_get+0x38/0xd0
dpcm_fe_dai_open+0x70/0x920
snd_pcm_open_substream+0x564/0x840
snd_pcm_open+0xfc/0x228
snd_pcm_capture_open+0x4c/0x78
snd_open+0xac/0x1a8
...
While initiliazing the links, ASoC treats the codec-to-codec links of this
card type as a DPCM backend. This error eventually leads to the Oops.
Most of the card driver code is shared between DPCM backends and
codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on
codec-to-codec links, leading to this problem. This commit fixes that.
Fixes: 0a8f1117a6 ("ASoC: meson: axg-card: add basic codec-to-codec link support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need
yet more quirks for the proper control names.
This patch provides the mapping table for those boards, correcting the
FU names for volume and mute controls as well as the terminal names
for jack controls. It also improves build_connector_control() not to
add the directional suffix blindly if the string is given from the
mapping table.
With this patch applied, the new UCM profiles will be effective.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3c6fd1f07e ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
This patch reverts the corresponding entry as a temporary solution.
Although Zenith II and co will see get the empty HD-audio bus again,
it'd be merely resource wastes and won't affect the functionality,
so it's no end of the world. We'll need to address this later,
e.g. by either switching to DMI string matching or using PCI ID &
SSID pairs.
Fixes: 3c6fd1f07e ("ALSA: hda: Add driver blacklist")
Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping
spec->init_amp detection") changed the way to assign spec->init_amp
field that specifies the way to initialize the amp. Along with the
change, the commit also replaced a few fixups that set spec->init_amp
in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather
aligning to the other fixups, and not supposed to change the actual
behavior.
However, this change turned out to cause a regression on FSC S7020,
which hit exactly the above. The reason was that there is still one
place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE
call, namely in alc_ssid_check().
This patch fixes the regression by adding the proper spec->init_amp
override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED.
Fixes: 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping spec->init_amp detection")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329
Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many Focusrite devices supports a limited set of sample rates per
altsetting. These includes audio interfaces with ADAT ports:
- Scarlett 18i6, 18i8 1st gen, 18i20 1st gen;
- Scarlett 18i8 2nd gen, 18i20 2nd gen;
- Scarlett 18i8 3rd gen, 18i20 3rd gen;
- Clarett 2Pre USB, 4Pre USB, 8Pre USB.
Maximum rate is exposed in the last 4 bytes of Format Type descriptor
which has a non-standard bLength = 10.
Tested-by: Alexey Skobkin <skobkin-ru@ya.ru>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Major regressions were detected by SOF CI on CherryTrail and Broadwell:
[ 25.705750] SSP2-Codec: ASoC: no backend playback stream
[ 27.923378] SSP2-Codec: ASoC: no users playback at close - state
This is root-caused to the introduction of the DAI capability checks
with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a
requirement for all DAIs to report at least a non-zero min_channels
field.
For some reason the SSP structures used for SKL+ did provide this
information but legacy platforms didn't.
Fixes: 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
One significant regression fix is for HD-audio buffer preallocation.
In 5.6 it was set to non-prompt for x86 and forced to 0, but this
turned out to be problematic for some applications, hence it gets
reverted. Distros would need to restore CONFIG_SND_HDA_PREALLOC_SIZE
value to the earlier values they've used in the past.
Other than that, we've received quite a few small fixes for HD-audio
and USB-audio. Most of them are for dealing with the broken TRX40
mobos and the runtime PM without HD-audio codecs.
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Merge tag 'sound-5.7-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"One significant regression fix is for HD-audio buffer preallocation.
In 5.6 it was set to non-prompt for x86 and forced to 0, but this
turned out to be problematic for some applications, hence it gets
reverted. Distros would need to restore CONFIG_SND_HDA_PREALLOC_SIZE
value to the earlier values they've used in the past.
Other than that, we've received quite a few small fixes for HD-audio
and USB-audio. Most of them are for dealing with the broken TRX40
mobos and the runtime PM without HD-audio codecs"
* tag 'sound-5.7-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: call runtime_allow() for all hda controllers
ALSA: hda: Allow setting preallocation again for x86
ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported
ALSA: hda: Skip controller resume if not needed
ALSA: hda: Keep the controller initialization even if no codecs found
ALSA: hda: Release resources at error in delayed probe
ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops
ALSA: hda: Don't release card at firmware loading error
ALSA: usb-audio: Check mapping at creating connector controls, too
ALSA: usb-audio: Don't create jack controls for PCM terminals
ALSA: usb-audio: Don't override ignore_ctl_error value from the map
ALSA: usb-audio: Filter error from connector kctl ops, too
ALSA: hda/realtek - Enable the headset mic on Asus FX505DT
ALSA: ctxfi: Remove unnecessary cast in kfree
This fixes two issues in the snd-soc-rcar driver blocking multichannel
HDMI audio out: The parent SSI in a multi-SSI configuration is not
correctly set up and started, and the SSI->HDMI channel mapping is
wrong.
With these patches, the following device tree snippet can be used on an
r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on
HDMI0:
rsnd_port1: port@1 {
rsnd_endpoint1: endpoint {
remote-endpoint = <&dw_hdmi0_snd_in>;
dai-format = "i2s";
bitclock-master = <&rsnd_endpoint1>;
frame-master = <&rsnd_endpoint1>;
playback = <&ssi0 &ssi1 &ssi2 &ssi9>;
};
};
With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1)
and 8ch audio output should work.
Matthias Blankertz (2):
ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode
ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode
sound/soc/sh/rcar/ssi.c | 8 ++++----
sound/soc/sh/rcar/ssiu.c | 2 +-
2 files changed, 5 insertions(+), 5 deletions(-)
base-commit: 7111951b8d
--
2.26.0
If we don't find any pcm, pcm will point at address at an offset from
the the list head and not a meaningful structure. Fix this by returning
correct pcm if found and NULL if not. Found with coccinelle.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HDMI?_SEL register maps up to four stereo SSI data lanes onto the
sdata[0..3] inputs of the HDMI output block. The upper half of the
register contains four blocks of 4 bits, with the most significant
controlling the sdata3 line and the least significant the sdata0 line.
The shift calculation has an off-by-one error, causing the parent SSI to
be mapped to sdata3, the first multi-SSI child to sdata0 and so forth.
As the parent SSI transmits the stereo L/R channels, and the HDMI core
expects it on the sdata0 line, this causes no audio to be output when
playing stereo audio on a multichannel capable HDMI out, and
multichannel audio has permutated channels.
Fix the shift calculation to map the parent SSI to sdata0, the first
child to sdata1 etc.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The parent SSI of a multi-SSI setup must be fully setup, started and
stopped since it is also part of the playback/capture setup. So only
skip the SSI (as per commit 203cdf51f2 ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 597b046f0d ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On Baytrail/Cherrytrail, the Atom/SST driver fails miserably:
[ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01
[ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed
[ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed
Commit b56be800f1 ("ASoC: soc-pcm: call
snd_soc_dai_startup()/shutdown() once") was the initial problematic
commit.
Commit 1ba616bd1a ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
was an attempt to fix things but it does not work on Baytrail,
reverting all changes seems necessary for now.
Fixes: 1ba616bd1a ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason, the MI2S DAIs do not have channels_min/max defined.
This means that snd_soc_dai_stream_valid() returns false,
i.e. the DAIs have neither valid playback nor capture stream.
It's quite surprising that this ever worked correctly,
but in 5.7-rc1 this is now failing badly: :)
Commit 0e9cf4c452 ("ASoC: pcm: check if cpu-dai supports a given stream")
introduced a check for snd_soc_dai_stream_valid() before calling
hw_params(), which means that the q6i2s_hw_params() function
was never called, eventually resulting in:
qcom-q6afe aprsvc:q6afe:4:4: no line is assigned
... even though "qcom,sd-lines" is set in the device tree.
Commit 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
now even avoids creating PCM devices if the stream is not supported,
which means that it is failing even earlier with e.g.:
Primary MI2S: ASoC: no backend playback stream
Avoid all that trouble by adding channels_min/max for the MI2S DAIs.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, PCM devices for DPCM are only created based on the
dpcm_playback/capture parameters of the DAI link, without considering
if the CPU/FE DAI is actually capable of playback/capture.
Normally the dpcm_playback/capture parameter should match the
capabilities of the CPU DAI. However, there is no way to set that
parameter from the device tree (e.g. with simple-audio-card or
qcom sound cards). dpcm_playback/capture are always both set to 1.
This causes problems when the CPU DAI does only support playback
or capture. Attemting to open that PCM device with an unsupported
stream type then results in a null pointer dereference:
Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
Internal error: Oops: 96000044 [#1] PREEMPT SMP
CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
pc : invalidate_paths_ep+0x30/0xe0
lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
Call trace:
invalidate_paths_ep+0x30/0xe0
snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
dpcm_path_get+0x38/0xd0
dpcm_fe_dai_open+0x70/0x920
snd_pcm_open_substream+0x564/0x840
snd_pcm_open+0xfc/0x228
snd_pcm_capture_open+0x4c/0x78
snd_open+0xac/0x1a8
...
... because the DAI playback/capture_widget is not set in that case.
We could add checks there to fix the problem (maybe we should
anyway), but much easier is to not expose the device as
playback/capture in the first place. Attemting to use that
device would always fail later anyway.
Add checks for snd_soc_dai_stream_valid() to the DPCM case
to avoid exposing playback/capture if it is not supported.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
As mentioned slightly out of patch context in the code, there
is no reset routine for the chip. On boards where the chip is
supplied by a fixed regulator, it might not even be resetted
during (e.g. watchdog) reboot and can be in any state.
If the device is probed with VAG enabled, the driver's probe
routine will generate a loud pop sound when ANA_POWER is
being programmed. Avoid this by properly disabling just the
VAG bit and waiting the required power down time.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Fabio Estevam <festivem@gmail.com>
Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Before the pci_driver->probe() is called, the pci subsystem calls
runtime_forbid() and runtime_get_sync() on this pci dev, so only call
runtime_put_autosuspend() is not enough to enable the runtime_pm on
this device.
For controllers with vgaswitcheroo feature, the pci/quirks.c will call
runtime_allow() for this dev, then the controllers could enter
rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like
Intel hda controllers, the runtime_pm is not enabled because the
runtime_allow() is not called.
Since it is no harm calling runtime_allow() twice, here let hda
driver call runtime_allow() for all controllers. Then the runtime_pm
is enabled on all controllers after the put_autosuspend() is called.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
WSA881x works in PDM mode so the wordlength is fixed, which also makes
the only field "WordLength" in DPN_BlockCtrl1 register a read-only.
Writing to this register will throw up errors with Qualcomm Controller.
So use ro_blockctrl1_reg flag to mark this field as read-only so that
core will not write to this register.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
All error paths in qcom_snd_parse_of() prints more specific error
messages, so silence the one in apq8096_platform_probe() and
sdm845_snd_platform_probe() to avoid spamming the kernel log.
Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Link: https://lore.kernel.org/r/20200406003229.2354631-1-bjorn.andersson@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the error message into snd_soc_dapm_new_dai from
dapm_connect_dai_pair, since the two copies are almost identical and
are the only callers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181209.30130-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following gcc warning:
sound/soc/intel/common/soc-acpi-intel-cml-match.c:116:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
^~~~~~~~~~~~
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Jason Yan <yanaijie@huawei.com>
Link: https://lore.kernel.org/r/20200410081117.21319-2-yanaijie@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following gcc warning:
sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
^~~~~~~~~~~~
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Jason Yan <yanaijie@huawei.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200410081117.21319-1-yanaijie@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If probe fails after enabling the regulators regulator_put is called for
each supply without having them disabled before. This produces some
warnings like
WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c
[<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14)
[<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4)
[<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4)
[<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c)
[<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38)
[<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38)
[<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c)
[<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c)
[<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c)
[<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60)
[<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc)
[<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0)
[<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0)
[<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108)
[<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88)
[<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250)
[<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244)
[<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540)
[<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8)
[<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20)
Fixes: 3fd6e7d9a1 (ASoC: tas571x: New driver for TI TAS571x power amplifiers)
Signed-off-by: Philipp Puschmann <p.puschmann@pironex.de>
Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Refactoring was done to factor out the linking of DAI widgets into
a helper function, dapm_add_valid_dai_widget. However when this was
done, a regression was introduced for CODEC to CODEC links. It was
over looked that the playback and capture variables persisted across
all CODEC DAIs being processed, which ensured that the special DAI
widget that is added for CODEC to CODEC links was only created once.
This bug causes kernel panics during DAPM shutdown.
To stick with the spirit of the original refactoring whilst fixing the
issue, variables to hold the DAI widgets are added to snd_soc_dai_link.
Furthermore the dapm_add_valid_dai_widget function is renamed to
dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC
to CODEC case and its primary job is to add routes connecting two DAI
widgets, making the original name quite misleading.
Fixes: 6c4b13b51a ("ASoC: Add dapm_add_valid_dai_widget helper")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181209.30130-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 450312b640 ("ASoC: soc-core: remove DAI suspend/resume")
removed the DAI side suspend/resume hooks and switched entirely to
component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was
not updated.
Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while
changing dai to component which allows to keep the struct
snd_soc_component_driver const.
This fixes build errors:
sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’:
sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’
dai_drv->suspend = s3c2412_i2s_suspend;
Reported-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 450312b640 ("ASoC: soc-core: remove DAI suspend/resume")
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200413124548.28197-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit c31427d0d2 ("ALSA: hda: No preallocation on x86
platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default
to zero for x86, as the preallocation should work almost all cases.
However, this expectation was too naive; some applications try to
allocate as the max buffer size as possible, and it leads to the
memory exhaustion. More badly, the commit changed the kconfig no
longer adjustable for x86, so you can't fix it statically (although it
can be still adjusted via procfs).
So, practically seen, it's more recommended to set a reasonable limit
for x86, too. This patch follows to that experience, and changes the
default to 2048 and allow the kconfig adjustable again.
Fixes: c31427d0d2 ("ALSA: hda: No preallocation on x86 platforms")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223
Link: https://lore.kernel.org/r/20200413201919.24241-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HD-audio controller does system-suspend and resume operations by
directly calling its helpers __azx_runtime_suspend() and
__azx_runtime_resume(). However, in general, we don't have to resume
always the device fully at the system resume; typically, if a device
has been runtime-suspended, we can leave it to runtime resume.
Usually for achieving this, the driver would call
pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the
system suspend and resume ops. Unfortunately, this doesn't work for
the resume path in our case. For handling the jack detection at the
system resume, a child codec device may need the (literally) forcibly
resume even if it's been runtime-suspended, and for that, the
controller device must be also resumed even if it's been suspended.
This patch is an attempt to improve the situation. It replaces the
direct __azx_runtime_suspend()/_resume() calls with with
pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight
trick as we've done for the codec side. More exactly:
- azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and
azx_runtime_resume(), so that it can be properly executed from the
system-suspend/resume path
- The WAKEEN handling depends on the card's power state now; it's set
and cleared only for the runtime-suspend
- azx_resume() checks whether any codec may need the forcible resume
beforehand. If the forcible resume is required, it does temporary
PM refcount up/down for actually triggering the runtime resume.
- A new helper function, hda_codec_need_resume(), is introduced for
checking whether the codec needs a forcible runtime-resume, and the
existing code is rewritten with that.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when the HD-audio controller driver doesn't detect any
codecs, it tries to abort the probe. But this abort happens at the
delayed probe, i.e. the primary probe call already returned success,
hence the driver is never unbound until user does so explicitly.
As a result, it may leave the HD-audio device in the running state
without the runtime PM. More badly, if the device is a HD-audio bus
that is tied with a GPU, GPU cannot reach to the full power down and
consumes unnecessarily much power.
This patch changes the logic after no-codec situation; it continues
probing without the further codec initialization but keep the
controller driver running normally.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Tested-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-hda-intel driver handles the most of its probe task in the delayed
work (either via workqueue or via firmware loader). When an error
happens in the later delayed probe, we can't deregister the device
itself because the probe callback already returned success and the
device was bound. So, for now, we set hda->init_failed flag and make
the rest untouched until the device gets really unbound.
However, this leaves the device up running, keeping the resources
without any use that prevents other operations.
In this patch, we release the resources at first when a probe error
happens in the delayed probe stage, but keeps the top-level object, so
that the PM and other ops can still refer to the object itself.
Also for simplicity, snd_hda_intel object is allocated via devm, so
that we can get rid of the explicit kfree calls.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the error path of the firmware loading error, the driver tries to
release the card object and set NULL to drvdata. This may be referred
badly at the possible PM action, as the driver itself is still bound
and the PM callbacks read the card object.
Instead, we continue the probing as if it were no option set. This is
often a better choice than the forced abort, too.
Fixes: 5cb543dba9 ("ALSA: hda - Deferred probing with request_firmware_nowait()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the mapping check to build_connector_control() so that the device
specific quirk can provide the node to skip for the badly behaving
connector controls. As an example, ALC1220-VB-based codec implements
the skip entry for the broken SPDIF connector detection.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mapping table may contain also ignore_ctl_error flag for devices
that are known to behave wild. Since this flag always writes the
card's own ignore_ctl_error flag, it overrides the value already set
by the module option, so it doesn't follow user's expectation.
Let's fix the code not to clear the flag that has been set by user.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ignore_ctl_error option should filter the error at kctl accesses,
but there was an overlook: mixer_ctl_connector_get() returns an error
from the request.
This patch covers the forgotten code path and apply filter_error()
properly. The locking error is still returned since this is a fatal
error that has to be reported even with ignore_ctl_error option.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Asus FX505DT with Realtek ALC233, the headset mic is connected
to pin 0x19, with default 0x411111f0.
Enable headset mic by reconfiguring the pin to an external mic
associated with the headphone on 0x21. Mic jack detection was also
found to be working.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131
Signed-off-by: Adam Barber <barberadam995@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes gathered since the previous update.
* ALSA core:
- Regression fix for OSS PCM emulation
* ASoC:
- Trivial fixes in reg bit mask ops, DAPM, DPCM and topology
- Lots of fixes for Intel-based devices
- Minor fixes for AMD, STM32, Qualcomm, Realtek
* Others
- Fixes for the bugs in mixer handling in HD-audio and ice1724
drivers that were caught by the recent kctl validator
- New quirks for HD-audio and USB-audio
Also this contains a fix for EDD firmware fix, which slipped
from anyone's hands.
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Merge tag 'sound-fix-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes gathered since the previous update.
ALSA core:
- Regression fix for OSS PCM emulation
ASoC:
- Trivial fixes in reg bit mask ops, DAPM, DPCM and topology
- Lots of fixes for Intel-based devices
- Minor fixes for AMD, STM32, Qualcomm, Realtek
Others:
- Fixes for the bugs in mixer handling in HD-audio and ice1724
drivers that were caught by the recent kctl validator
- New quirks for HD-audio and USB-audio
Also this contains a fix for EDD firmware fix, which slipped from
anyone's hands"
* tag 'sound-fix-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (35 commits)
ALSA: hda: Add driver blacklist
ALSA: usb-audio: Add mixer workaround for TRX40 and co
ALSA: hda/realtek - Add quirk for MSI GL63
ALSA: ice1724: Fix invalid access for enumerated ctl items
ALSA: hda: Fix potential access overflow in beep helper
ASoC: cs4270: pull reset GPIO low then high
ALSA: hda/realtek - Add HP new mute led supported for ALC236
ALSA: hda/realtek - Add supported new mute Led for HP
ASoC: rt5645: Add platform-data for Medion E1239T
ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
ASoC: stm32: sai: Add missing cleanup
ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Alpha S
ASoC: Intel: atom: Fix uninitialized variable compiler warning
ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked
ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
ASoC: SOF: Turn "firmware boot complete" message into a dbg message
ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
ALSA: pcm: oss: Fix regression by buffer overflow fix (again)
ALSA: pcm: oss: Fix regression by buffer overflow fix
edd: Use scnprintf() for avoiding potential buffer overflow
...
Function soc_tplg_dai_config can fail, check for and handle possible
failure.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-7-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Function pcm_new_ver can fail, so we should check it's return value and
handle possible error.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Function soc_tplg_add_route can propagate error code from callback, we
should check its return value and handle fail in correct way.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Functions soc_tplg_denum_create, soc_tplg_dmixer_create,
soc_tplg_dbytes_create can fail, so their return values should be
checked and error should be propagated.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Function soc_tplg_create_tlv can fail, so we should check if it succeded
or not and proceed appropriately.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kstrdup is an allocation function and it can fail, so its return value
should be checked and handled appropriately.
In order to check all cases, we need to modify set_stream_info to return
a value, so check that everything went correctly when doing kstrdup().
Later add proper checks and error handlers.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of fixes that have been accumilated since the merge window,
mainly relating to x86 platform support.
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Merge tag 'asoc-fix-v5.7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.7
A collection of fixes that have been accumilated since the merge window,
mainly relating to x86 platform support.
The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.
This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry. As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio. The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.
As a workaround, simply ignore this node by adding a usbmix_name_map
table entry. The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The access to Analog Capture Source control value implemented in
prodigy_hifi.c is wrong, as caught by the recently introduced sanity
check; it should be accessing value.enumerated.item[] instead of
value.integer.value[]. This patch corrects the wrong access pattern.
Fixes: 6b8d6e5518 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep control helper function blindly stores the values in two
stereo channels no matter whether the actual control is mono or
stereo. This is practically harmless, but it annoys the recently
introduced sanity check, resulting in an error when the checker is
enabled.
This patch corrects the behavior to store only on the defined array
member.
Fixes: 0401e8548e ("ALSA: hda - Move beep helper functions to hda_beep.c")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull the RST line low then high when initializing the driver,
in order to force a reset of the chip.
Previously, the line was not pulled low, which could result in
the chip registers not resetting to their default values on boot.
Signed-off-by: Mike Willard <mwillard@izotope.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200401205454.79792-1-mwillard@izotope.com
Signed-off-by: Mark Brown <broonie@kernel.org>
HP new platform has new mute led feature.
COEF index 0x34 bit 5 to control playback mute led.
COEF index 0x35 bit 2 and bit 3 to control Mic mute led.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Note Book supported new mute Led.
Hardware PIN was not enough to meet old LED rule.
JD2 to control playback mute led.
GPO3 to control capture mute led.
(ALC285 didn't control GPO3 via verb command)
This two PIN just could control by COEF registers.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Medion E1239T uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like other models following Intel's Brasswell's reference design.
This commit adds a DMI quirk pointing to the intel_braswell_platform_data
for this model.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402185257.3355-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MPMAN MPWIN895CL tablet almost fully works with out default settings.
The only problem is that it has only 1 speaker so any sounds only playing
on the right channel get lost.
Add a quirk for this model using the default settings + MONO_SPEAKER.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
converts some function calls to their non-devm equivalents. The
appropriate cleanup code was added to the remove function, but not
to the probe function. Add a call to snd_dmaengine_pcm_unregister
to compensate for the call to snd_dmaengine_pcm_register in case
of subsequent failure.
Fixes: commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:16d8) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com>
Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here are 3 SPDX patches for 5.7-rc1.
One fixes up the SPDX tag for a single driver, while the other two go
through the tree and add SPDX tags for all of the .gitignore files as
needed.
Nothing too complex, but you will get a merge conflict with your current
tree, that should be trivial to handle (one file modified by two things,
one file deleted.)
All 3 of these have been in linux-next for a while, with no reported
issues other than the merge conflict.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'spdx-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/spdx
Pull SPDX updates from Greg KH:
"Here are three SPDX patches for 5.7-rc1.
One fixes up the SPDX tag for a single driver, while the other two go
through the tree and add SPDX tags for all of the .gitignore files as
needed.
Nothing too complex, but you will get a merge conflict with your
current tree, that should be trivial to handle (one file modified by
two things, one file deleted.)
All three of these have been in linux-next for a while, with no
reported issues other than the merge conflict"
* tag 'spdx-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/spdx:
ASoC: MT6660: make spdxcheck.py happy
.gitignore: add SPDX License Identifier
.gitignore: remove too obvious comments
GCC 10 gives a "variable might be used uninitialized" warning for the
block variable in sst_prepare_and_post_msg().
This is a false-positive warning, but lets fix it anyways.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402185359.3424-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sst_fill_and_send_cmd_unlocked must be called with the drv->lock mutex
locked already. In the past there have been cases where this was not the
case, add a WARN_ON to check for drv->lock being locked.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402185359.3424-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some
places it is called with the drv->lock mutex already held.
So it must always be called with the mutex locked. This commit adds missing
locking in the sst_set_be_modules() code-path.
Fixes: 24c8d14192 ("ASoC: Intel: mrfld: add DSP core controls")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using a Canon Lake machine with the SOF driver causes dmesg to fill
up with a ton of these messages:
[ 275.902194] sof-audio-pci 0000:00:1f.3: firmware boot complete
[ 351.529358] sof-audio-pci 0000:00:1f.3: firmware boot complete
[ 560.049047] sof-audio-pci 0000:00:1f.3: firmware boot complete
etc.
Since the DSP is powered down when not in used this happens everytime
e.g. a notification plays, polluting dmesg.
Turn this messages into a debug message, matching what the code already
does for the ""booting DSP firmware" message.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402184948.3014-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card.
The MIDI controller part is standard but the PCM part is "vendor specific".
Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE.
Input is not working.
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ This is essentially the same fix as commit ae769d3556, but it's
adapted to the latest code for 5.7; hence it contains no Fixes or
other tags for avoid backport confusion -- tiwai ]
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications. The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.
This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.
Link: https://lore.kernel.org/r/20200403073818.27943-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became again a busy development cycle. There are a few ALSA
core updates (merely API cleanups and sparse fixes), while majority
of other changes are found in ASoC scene.
Here are some highlights:
* ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)
* ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140
* HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs
* USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus
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Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became again a busy development cycle. There are few ALSA core
updates (merely API cleanups and sparse fixes), with the majority of
other changes are found in ASoC scene.
Here are some highlights:
ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)
ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140
HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs
USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus"
* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
ALSA: hda/realtek - a fake key event is triggered by running shutup
ALSA: hda: default enable CA0132 DSP support
ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
ASoC: tlv320adcx140: Remove undocumented property
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: boards: add sof_sdw machine driver
ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
ASoC: rt5682: move DAI clock registry to I2S mode
ASoC: pxa: magician: convert to use i2c_new_client_device()
ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
...
The audio setup on the Lenovo Carbon X1 8th gen is the same as that on
the Lenovo Carbon X1 7th gen, as such it needs the same
ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk.
This fixes volume control of the speaker not working among other things.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196
Cc: stable@vger.kernel.org
Suggested-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If I2C is n but SoundWire is m, building fails:
sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class
module_i2c_driver(rt5682_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration
Guard this use #ifdef CONFIG_I2C.
Fixes: 5549ea6479 ("ASoC: rt5682: fix unmet dependencies")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull networking updates from David Miller:
"Highlights:
1) Fix the iwlwifi regression, from Johannes Berg.
2) Support BSS coloring and 802.11 encapsulation offloading in
hardware, from John Crispin.
3) Fix some potential Spectre issues in qtnfmac, from Sergey
Matyukevich.
4) Add TTL decrement action to openvswitch, from Matteo Croce.
5) Allow paralleization through flow_action setup by not taking the
RTNL mutex, from Vlad Buslov.
6) A lot of zero-length array to flexible-array conversions, from
Gustavo A. R. Silva.
7) Align XDP statistics names across several drivers for consistency,
from Lorenzo Bianconi.
8) Add various pieces of infrastructure for offloading conntrack, and
make use of it in mlx5 driver, from Paul Blakey.
9) Allow using listening sockets in BPF sockmap, from Jakub Sitnicki.
10) Lots of parallelization improvements during configuration changes
in mlxsw driver, from Ido Schimmel.
11) Add support to devlink for generic packet traps, which report
packets dropped during ACL processing. And use them in mlxsw
driver. From Jiri Pirko.
12) Support bcmgenet on ACPI, from Jeremy Linton.
13) Make BPF compatible with RT, from Thomas Gleixnet, Alexei
Starovoitov, and your's truly.
14) Support XDP meta-data in virtio_net, from Yuya Kusakabe.
15) Fix sysfs permissions when network devices change namespaces, from
Christian Brauner.
16) Add a flags element to ethtool_ops so that drivers can more simply
indicate which coalescing parameters they actually support, and
therefore the generic layer can validate the user's ethtool
request. Use this in all drivers, from Jakub Kicinski.
17) Offload FIFO qdisc in mlxsw, from Petr Machata.
18) Support UDP sockets in sockmap, from Lorenz Bauer.
19) Fix stretch ACK bugs in several TCP congestion control modules,
from Pengcheng Yang.
20) Support virtual functiosn in octeontx2 driver, from Tomasz
Duszynski.
21) Add region operations for devlink and use it in ice driver to dump
NVM contents, from Jacob Keller.
22) Add support for hw offload of MACSEC, from Antoine Tenart.
23) Add support for BPF programs that can be attached to LSM hooks,
from KP Singh.
24) Support for multiple paths, path managers, and counters in MPTCP.
From Peter Krystad, Paolo Abeni, Florian Westphal, Davide Caratti,
and others.
25) More progress on adding the netlink interface to ethtool, from
Michal Kubecek"
* git://git.kernel.org/pub/scm/linux/kernel/git/netdev/net-next: (2121 commits)
net: ipv6: rpl_iptunnel: Fix potential memory leak in rpl_do_srh_inline
cxgb4/chcr: nic-tls stats in ethtool
net: dsa: fix oops while probing Marvell DSA switches
net/bpfilter: remove superfluous testing message
net: macb: Fix handling of fixed-link node
net: dsa: ksz: Select KSZ protocol tag
netdevsim: dev: Fix memory leak in nsim_dev_take_snapshot_write
net: stmmac: add EHL 2.5Gbps PCI info and PCI ID
net: stmmac: add EHL PSE0 & PSE1 1Gbps PCI info and PCI ID
net: stmmac: create dwmac-intel.c to contain all Intel platform
net: dsa: bcm_sf2: Support specifying VLAN tag egress rule
net: dsa: bcm_sf2: Add support for matching VLAN TCI
net: dsa: bcm_sf2: Move writing of CFP_DATA(5) into slicing functions
net: dsa: bcm_sf2: Check earlier for FLOW_EXT and FLOW_MAC_EXT
net: dsa: bcm_sf2: Disable learning for ASP port
net: dsa: b53: Deny enslaving port 7 for 7278 into a bridge
net: dsa: b53: Prevent tagged VLAN on port 7 for 7278
net: dsa: b53: Restore VLAN entries upon (re)configuration
net: dsa: bcm_sf2: Fix overflow checks
hv_netvsc: Remove unnecessary round_up for recv_completion_cnt
...
Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.
Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.
According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:
The flag is being used in the sense explained in the previous audio
meeting -- the data transfer granularity isn't fine enough but aligned
to the period size (or less).
q6asm-dai reports the position as multiple of
prtd->pcm_count = snd_pcm_lib_period_bytes(substream)
so it indeed just a multiple of the period size.
Therefore adding the flag here seems appropriate and makes audio
work out of the box.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra
endpoint descriptor.
The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00
As the code in snd_usbmidi_get_ms_info() looks only at the
first extra descriptor to find USB_DT_CS_ENDPOINT the device
as such is recognized but there is neither input nor output
configured.
The patch iterates through the extra descriptors to find the
proper one. With this patch the device is correctly configured.
Signed-off-by: Andreas Steinmetz <ast@domdv.de>
Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
patch_realtek.c has historically failed to properly configure the PC
Beep Hidden Register for the ALC256 codec (among others). Depending on
your kernel version, symptoms of this misconfiguration can range from
chassis noise, picked up by a poorly-shielded PCBEEP trace, getting
amplified and played on your internal speaker and/or headphones to loud
feedback, which responds to the "Headphone Mic Boost" ALSA control,
getting played through your headphones. For details of the problem, see
the patch in this series titled "ALSA: hda/realtek - Set principled PC
Beep configuration for ALC256", which fixes the configuration.
These symptoms have been most noticed on the Dell XPS 13 9350 and 9360,
popular laptops that use the ALC256. As a result, several model-specific
fixups have been introduced to try and fix the problem, the most
egregious of which locks the "Headphone Mic Boost" control as a hack to
minimize noise from a feedback loop that shouldn't have been there in
the first place.
Now that the underlying issue has been fixed, remove all these fixups.
Remaining fixups needed by the XPS 13 are all picked up by existing pin
quirks.
This change should, for the XPS 13 9350/9360
- Significantly increase volume and audio quality on headphones
- Eliminate headphone popping on suspend/resume
- Allow "Headphone Mic Boost" to be set again, making the headphone
jack fully usable as a microphone jack too.
Fixes: 8c69729b44 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd78561 ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek PC Beep Hidden Register[1] is currently set by
patch_realtek.c in two different places:
In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to
non-beep input on 1Ah and no 1Ah loopback to either headphones or
speakers. (Although, curiously, the loopback amp is still enabled.) This
write was added fairly recently by commit e3743f431143 ("ALSA:
hda/realtek - Dell headphone has noise on unmute for ALC236") and is a
safe default. However, it happens in the wrong place:
alc_fill_eapd_coef() runs on module load and cold boot but not on S3
resume, meaning the register loses its value after suspend.
Conversely, in alc256_init(), the register is updated to unset bit 13
(disable speaker loopback) and set bit 5 (set non-beep input on 1Ah).
Although this write does run on S3 resume, it's not quite enough to fix
up the register's default value of 0x3717. What's missing is a set of
bit 14 to disable headphone loopback. Without that, we end up with a
feedback loop where the headphone jack is being driven by amplified
samples of itself[2].
This change eliminates the update in alc256_init() and replaces it with
the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is
supposed to be the codec's default value, so using it will make
debugging easier for Realtek.
Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs.
[1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst
[2] Setting the "Headphone Mic Boost" control from userspace changes
this feedback loop and has been a widely-shared workaround for headphone
noise on laptops like the Dell XPS 13 9350. This commit eliminates the
feedback loop and makes the workaround unnecessary.
Fixes: e1e8c1fdce ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull perf updates from Ingo Molnar:
"The main changes in this cycle were:
Kernel side changes:
- A couple of x86/cpu cleanups and changes were grandfathered in due
to patch dependencies. These clean up the set of CPU model/family
matching macros with a consistent namespace and C99 initializer
style.
- A bunch of updates to various low level PMU drivers:
* AMD Family 19h L3 uncore PMU
* Intel Tiger Lake uncore support
* misc fixes to LBR TOS sampling
- optprobe fixes
- perf/cgroup: optimize cgroup event sched-in processing
- misc cleanups and fixes
Tooling side changes are to:
- perf {annotate,expr,record,report,stat,test}
- perl scripting
- libapi, libperf and libtraceevent
- vendor events on Intel and S390, ARM cs-etm
- Intel PT updates
- Documentation changes and updates to core facilities
- misc cleanups, fixes and other enhancements"
* 'perf-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (89 commits)
cpufreq/intel_pstate: Fix wrong macro conversion
x86/cpu: Cleanup the now unused CPU match macros
hwrng: via_rng: Convert to new X86 CPU match macros
crypto: Convert to new CPU match macros
ASoC: Intel: Convert to new X86 CPU match macros
powercap/intel_rapl: Convert to new X86 CPU match macros
PCI: intel-mid: Convert to new X86 CPU match macros
mmc: sdhci-acpi: Convert to new X86 CPU match macros
intel_idle: Convert to new X86 CPU match macros
extcon: axp288: Convert to new X86 CPU match macros
thermal: Convert to new X86 CPU match macros
hwmon: Convert to new X86 CPU match macros
platform/x86: Convert to new CPU match macros
EDAC: Convert to new X86 CPU match macros
cpufreq: Convert to new X86 CPU match macros
ACPI: Convert to new X86 CPU match macros
x86/platform: Convert to new CPU match macros
x86/kernel: Convert to new CPU match macros
x86/kvm: Convert to new CPU match macros
x86/perf/events: Convert to new CPU match macros
...
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Cezary Rojewski (4):
ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link
sound/soc/intel/boards/bdw-rt5650.c | 1 -
sound/soc/intel/boards/bdw-rt5677.c | 1 -
sound/soc/intel/boards/broadwell.c | 1 -
sound/soc/intel/boards/haswell.c | 1 -
4 files changed, 4 deletions(-)
--
2.17.1
The addition of a single flag to track the DAI status prevents the DAI
startup sequence from being called on capture if the DAI is already
used for playback.
Fix by extending the existing code with one flag per direction.
Fixes: b56be800f1 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once")
Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200330160602.10180-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If regwshift is 32 and the selected architecture compiles '<<' operator
for signed int literal into rotating shift, '1<<regwshift' became 1 and
it makes regwmask to 0x0.
The literal is set to unsigned long to get intended regwmask.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/001001d60665$db7af3e0$9270dba0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Reported-by: Dominik Brodowski <linux@dominikbrodowski.net>
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200319204947.18963-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This avoids residual bit form previous format when the format is changed.
Hence, the resultant format is not an invalid one.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200328093921.32211-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The NULL check can be done gracefully without cast. It fixes a
compile warning like:
sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
Fixes: 88eb404ccc ("ASoC: brcm: Add DSL/PON SoC audio driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200330135645.9707-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a very big update for the core since Morimoto-san has been
rather busy continuing his refactorings to clean up a lot of the cruft
that we have accumilated over the years. We've also gained several new
drivers, including initial (but still not complete) parts of the Intel
SoundWire support.
- Lots of refactorings to modernize the code from Morimoto-san.
- Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven.
- Continued refactoring and fixing of the Intel support.
- Soundwire and more advanced clocking support for Realtek RT5682.
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and
TLV320ADCX140.
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Merge tag 'asoc-v5.7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.7
This is a very big update for the core since Morimoto-san has been
rather busy continuing his refactorings to clean up a lot of the cruft
that we have accumilated over the years. We've also gained several new
drivers, including initial (but still not complete) parts of the Intel
SoundWire support.
- Lots of refactorings to modernize the code from Morimoto-san.
- Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven.
- Continued refactoring and fixing of the Intel support.
- Soundwire and more advanced clocking support for Realtek RT5682.
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and
TLV320ADCX140.
On the Lenovo X1C7 machines, after we plug the headset, the rt_resume()
and rt_suspend() of the codec driver will be called periodically, the
driver can't stay in the rt_suspend state even users doen't use the
sound card.
Through debugging, I found when running rt_suspend(), it will call
alc225_shutup(), in this function, it will change 3k pull down control
by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a
fake key event and that event will resume the codec, when codec
suspend agin, it will trigger the fake key event one more time, this
process will repeat.
If disable the key event before changing the pull down control, it
will not trigger fake key event. It also needs to restore the pull
down control and re-enable the key event, otherwise the system can't
get key event when codec is in rt_suspend state.
Also move some functions ahead of alc225_shutup(), this can save the
function declaration.
Fixes: 76f7dec08f (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1)
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200329082018.20486-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as
well. Disabled DSP support leads to a hanging alsa system and no sound
output on the card otherwise. Tested on:
06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01)
Signed-off-by: Rouven Czerwinski <rouven@czerwinskis.de>
Link: https://lore.kernel.org/r/20200329053710.4276-1-r.czerwinski@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To handle multiple hardware combinations, this patchset suggests a
single machine driver which will create and initialize dailinks
dynamically. This allows us to support new configurations easily, as
shown with the TigerLake rt5682 example.
Each configuration updates the card component string, and UCM can test
for the presence of components to configure them as needed.
Since we use a single the machine driver name, all previous ACPI
tables need to be updated. That should have no impact since the
machine drivers listed at the time were not upstreamed and are no
longer maintained.
Naveen Manohar (2):
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper
function
Pierre-Louis Bossart (1):
ASoC: Intel: boards: add sof_sdw machine driver
Rander Wang (1):
ASoC: Intel: soc-acpi: update topology and driver name for SoundWire
platforms
sound/soc/intel/boards/Kconfig | 24 +
sound/soc/intel/boards/Makefile | 8 +-
sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++
sound/soc/intel/boards/sof_sdw_common.h | 114 +++
sound/soc/intel/boards/sof_sdw_dmic.c | 42 +
sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++
sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++
sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++
sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++
sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++
sound/soc/intel/boards/sof_sdw_rt715.c | 42 +
.../intel/common/soc-acpi-intel-cml-match.c | 24 +-
.../intel/common/soc-acpi-intel-icl-match.c | 6 +-
.../intel/common/soc-acpi-intel-tgl-match.c | 30 +-
14 files changed, 1896 insertions(+), 11 deletions(-)
create mode 100644 sound/soc/intel/boards/sof_sdw.c
create mode 100644 sound/soc/intel/boards/sof_sdw_common.h
create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c
create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c
--
2.20.1
