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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
...
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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.
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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>
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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>
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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>
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The commit 0d6defc7e0e4 ("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 0d6defc7e0e4 ("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>
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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>
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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>
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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: 24c8d14192cc ("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>
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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>
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Current topology doesn't add prefix of component to new kcontrol.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: 5549ea647997 ("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>
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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>
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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>
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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: 2a9e92d371db ("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>
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link" from Cezary Rojewski <cezary.rojewski@intel.com>:
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
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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>
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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>
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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>
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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>
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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: b56be800f1292 ("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>
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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>
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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>
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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: 88eb404ccc3e ("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>
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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>
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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>
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MSI GL63 laptop requires the similar quirk like other MSI models,
ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID
for the device, hence we need to take the codec SSID (1462:1275)
instead.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: 6b8d6e5518e2 ("[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>
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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: 0401e8548eac ("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>
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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>
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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>
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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>
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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>
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[ This is essentially the same fix as commit ae769d355664, 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
f2ecf903ef06: "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>
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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>
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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
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The SPDX-License-Identifier shall not be suffixed with anything further.
This makes ./scripts/spdxcheck.py complain:
sound/soc/codecs/mt6660.c: 1:36 Invalid token: //
Clean up SPDX-License-Identifier line to make spdxcheck.py happy.
Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Add SPDX License Identifier to all .gitignore files.
Signed-off-by: Masahiro Yamada <masahiroy@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Some .gitignore files have comments like "Generated files",
"Ignore generated files" at the header part, but they are
too obvious.
Signed-off-by: Masahiro Yamada <masahiroy@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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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
...
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endpoint descriptor
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>
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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: 8c69729b4439 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd785619a ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb21 ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457d0 ("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>
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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: e1e8c1fdce8b ("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>
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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.
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Bossart <pierre-louis.bossart@linux.intel.com>:
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
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Add support for Google Volteer device. As per new unified soundwire machine
driver, add rt5682-sdw helper function, which configures codec to Link0.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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RT5682 is in SoundWire mode on link0.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This machine driver provides support for different configurations:
RT700, RT711, RT1308 (1x and 2x, I2S or SoundWire mode), and RT715
CometLake, Icelake, TigerLake.
PDM digital microphones
HDMI
To avoid introducing one driver per configuration, this common machine
driver relies on platform-specific information, tables and quirks to
dynamically create the relevant dailinks.
Unlike a lot of machine drivers, we use different DAI links for
SoundWire capture and playback since the Cadence PDIs can do capture
OR playback, not both simultaneously.
For each configuration, the card component string is updated so that UCM
can select the relevant parts.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Update topology and reflect change to unified machine driver for SoundWire.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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