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 e3743f4311 ("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
There are a couple of statements that are not indented correctly,
add in the missing tab and break the lines to address a checkpatch
warning.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200327141429.269191-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
This patchset provides the support for SoundWire support on Intel
CometLake, IcelLake and TigerLake RVP platforms and form-factor
devices to be released 'soon'.
The bulk of the code is about detecting a valid SoundWire
configuration from ACPI, and implementing the interfaces suggested in
'[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces'
for interrupts, PCI wakes and clock-stop configurations.
Since that SoundWire series will not be in 5.7, the build support for
SOF w/ SoundWire is not provided for now, and fall-back functions will
be used. This code is tested on a daily basis in the SOF tree and is
not expected to change in significant ways.
Changes since v2:
Corrected error in ACPI table (thanks Amadeusz)
Added patch 11 to add reset cycle required on some SoundWire platforms
Bard Liao (1):
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt
handlers
Pierre-Louis Bossart (8):
ASoC: soc-acpi: expand description of _ADR-based devices
ASoC: SOF: Intel: add SoundWire configuration interface
ASoC: SOF: IPC: dai-intel: move ALH declarations in header file
ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks
ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect
ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop
quirks
ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing
capabilities
Rander Wang (2):
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
include/sound/soc-acpi.h | 39 +-
include/sound/sof/dai-intel.h | 18 +-
.../intel/common/soc-acpi-intel-cml-match.c | 87 +++-
.../intel/common/soc-acpi-intel-icl-match.c | 97 ++++-
.../intel/common/soc-acpi-intel-tgl-match.c | 49 ++-
sound/soc/sof/intel/hda-ctrl.c | 25 +-
sound/soc/sof/intel/hda-dsp.c | 2 +
sound/soc/sof/intel/hda-loader.c | 31 ++
sound/soc/sof/intel/hda.c | 400 ++++++++++++++++++
sound/soc/sof/intel/hda.h | 66 +++
10 files changed, 750 insertions(+), 64 deletions(-)
--
2.20.1
The SoundWire mode doesn't need the DAI clocks.
Therefore, the DAI clock registry moves to I2S mode case.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200327073849.18291-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If pci device is in D0, wakeen interrupt will be
aggregated at cAVS level as interrupt. This commit
check the wakeen status and process it in irq thread
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When a SoundWire link is in clock stop state, a Slave device may wake
up the Master for some events such as jack detection. The WAKEEN
interrupt will be triggered and processed by the audio pci device.
If audio device is in D3, the interrupt will be routed to PME, or
aggregated at cAVS level as interrupt when audio device is in D0. This
patch only supports D3 case, where the audio pci device will be
resumed by a PME event and the WAKEEN interrupt will be processed
after audio pci device is powered up and ROM is initialized
successfully.
The WAKEEN handling is only enabled after the first boot due to
dependencies on a shim_lock mutex being initialized.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have a single irq handler for SOF interrupts. We can further merge
SoundWire ones to completely remove MSI interrupts handling issues
leading to timeouts.
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/20200325215027.28716-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Doing this avoid conflicts and errors reported on the bus.
The interrupts are only re-enabled on resume after the firmware is
downloaded, so the behavior is not fully symmetric
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For now we have a limited number of machine driver configurations, and
we can detect them based on the link configuration returned after
checking hardware and firmware (BIOS) configurations.
The link configuration is checked with a link_mask as well as a list
of _ADR descriptors for each link.
There is a chance that in extreme cases where the BIOS contains too
much information we would need to detect which Slave devices actually
report as 'attached'. This would be more accurate than static
table-based solutions, but it also introduces timing dependencies
since we don't know when those devices might become attached, so will
only be only be looked at if we see limitations with static methods
and the usual quirks based e.g. on DMI information.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These callbacks are invoked when a matching hw_params/hw_free() DAI
operation takes place, and will result in IPC operations with the SOF
firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the SoundWire core supports the multi-step initialization,
call the relevant APIs.
The actual hardware enablement can be done in two places, ideally we'd
want to startup the SoundWire IP as soon as possible (while still
taking power rail dependencies into account)
However when suspend/resume is implemented, the DSP device will be
resumed first, and only when the DSP firmware is downloaded/booted
would the SoundWire child devices be resumed, so there are only
marginal benefits in starting the IP earlier for the first probe.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For SoundWire, we need to know if endpoints needs to be 'aggregated'
(MIPI parlance, meaning logically grouped), e.g. when two speaker
amplifiers need to be handled as a single logical output.
We don't necessarily have the information at the firmware (BIOS)
level, so add a notion of endpoints and specify if a device/endpoint
is part of a group, with a position.
This may be expanded in future solutions, for now only provide a group
and position information.
Since we modify the header file, change all existing upstream tables
as well to avoid breaking compilation/bisect.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The system in question uses ALC285, and it uses GPIO 0x04 to control its
mute LED.
The mic mute LED can be controlled by GPIO 0x01, however the system uses
DMIC so we should use that to control mic mute LED.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200327044626.29582-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When two (or more) amplifiers are on the same link, the integrator may:
a) assign dedicated slots for each of the amplifiers.
b) provide the same configuration to all amplifiers, and rely on
additional controls/processing in the amplifier to generate different
outputs.
case a) was the initial direction for SoundWire and is required for
amplifiers with limited capabilities, but to deal with orientation or
'posture' changes it's easier to implement case b) when the amplifier
can deal with multiple channels.
This patchset suggest the use of the set_tdm_slot() API to define
which of the channels will be consumed by what amplifiers. This maps
well with SoundWire's 'ChannelEnable' registers. The notion of
slot_width is however irrelevant here and ignored, and SoundWire ports
are typically single direction, so only one of the two masks shall be
used.
Pierre-Louis Bossart (2):
ASoC: rt1308-sdw: add set_tdm_slot() support
ASoC: rt1308-sdw: use slot and rx_mask to configure stream
sound/soc/codecs/rt1308-sdw.c | 38 +++++++++++++++++++++++++++++++----
sound/soc/codecs/rt1308-sdw.h | 2 ++
2 files changed, 36 insertions(+), 4 deletions(-)
--
2.20.1
The AC'97 based PXA machines currently don't build reliably as they don't
ensure that an AC'97 bus is built, causing at least eseries_pxa_defconfig
to fail to build. Add selects to fix this.
Reported-by: KernelCI <bot@kernelci.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200326180116.21375-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
regmap needs to be selected by users which for machine drivers that select
AC'97 CODEC drivers means that we need to also select regmap to ensure that
the CODEC driver will build if nothing else enables regmap as is likely for
such systems.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200326151053.40806-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
If the DAI was configured with a set_tdm_slots() call, use the information.
A platform or machine driver may configure each amplifier to extract
different bitSlots from the frame, or extract the same data and use
processing to generate the relevant output. The latter case is easier
to handle in case of orientation changes.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325212905.28145-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Further align HDA init sequence to the legacy non-DSP HDA driver by
calling snd_hdac_set_codec_wakeup() during the chip init sequence.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The misc clock gating (MISCBDCGE) is disabled for controller reset and
reenabled once reset is complete.
Fix the case when error happens during reset, and clock gating is
left disabled. The clock gating should be reenabled also in this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the VirtIO case the sof_pcm_open() function isn't called on the
host during guest streaming, which then leaves "work" structures
uninitialised. However it is then used to handle position update
messages from the DSP. Move their initialisation to immediately after
allocation of the containing structure.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use for_each_pcm_streams() to enumerate streams in sof_dai_load()
instead of doing that manually.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Improve the DSP power state logs with the state names
instead of values.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Update tgl mach table with: Maxim98373 Amp and ALC5682 hp codec.
Both of the codecs are on I2S bus.
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch does the below:
1. Adds the driver data and updates quirk info for TGL
with Max98373 speaker amp and ALC5682 headset codec.
2. Added max98373 speaker related code to common file for re-use.
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add "Spk Switch" and associated widget, route to max98360a
speaker amp for power saving, also remove the speaker_amp_init()
callback with complete separated tables for max98373 and max98360a.
Signed-off-by: Bhat, Uday M <uday.m.bhat@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Without the dynamic flag to allow runtime routing, the card cannot
probe on chromebooks because SOF is constantly waiting for the link.
Adding flag back to allow upstream kernels to work on rt5682 based
chromebooks since SOF can now ignore the hard coded front end.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The USB-audio driver may call snd_card_register() multiple times as
its probe function is per USB interface while some USB-audio devices
may provide multiple interfaces to assign different streams although
they belong to the same device. This works in most cases but the
registration is racy, hence it may miss the device recognition,
e.g. PA doesn't see certain devices when hotplugged.
The recent addition of the delayed registration quirk allows to sync
the registration at the last known interface, and the previous commit
added a new module option to allow the dynamic setup for that
purpose.
Now, this patch tries to find out and notifies for such devices that
require the delayed registration. It shows a message like:
Found post-registration device assignment: 1234abcd:02
If you hit this message, you can pass delayed_register module option
like:
snd_usb_audio.delayed_register=1234abcd:02
by just copying the last shown entry. If this works, it can be added
statically in the quirk list, registration_quirks[] found at the end
of sound/usb/quirks.c.
Link: https://lore.kernel.org/r/20200325103322.2508-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new option for specifying the quirk for delayed registration of
the certain device. A list of devices can be passed in a form
ID:IFACE,ID:IFACE,ID:IFACE,....
where ID is the 32bit hex number combo of vendor and device IDs and
IFACE is the interface number to trigger the register.
When a matching device is probed, the card registration is delayed
until the given interface is probed. It's needed for syncing the
registration until the last interface when multiple interfaces are
provided for the same card.
Link: https://lore.kernel.org/r/20200325103322.2508-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A slight refactoring of the registration quirk code. Now it uses the
table lookup for easy additions in future. Also the return type was
changed to bool, and got a few more comments.
Link: https://lore.kernel.org/r/20200325103322.2508-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Overlapping header include additions in macsec.c
A bug fix in 'net' overlapping with the removal of 'version'
string in ena_netdev.c
Overlapping test additions in selftests Makefile
Overlapping PCI ID table adjustments in iwlwifi driver.
Signed-off-by: David S. Miller <davem@davemloft.net>
The signed 1 bit bitfields should be unsigned, so make them unsigned.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200325132913.110115-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't right and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Fixes: a40acc6bfc ("ASoC: Intel: bdw-rt5650: change cpu_dai and platform components for SOF")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325131611.545-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't right and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Fixes: 4865bde187 ("ASoC: Intel: bdw-rt5677: change cpu_dai and platform components for SOF")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325131611.545-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't correct and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Fixes: 64df6afa0d ("ASoC: Intel: broadwell: change cpu_dai and platform components for SOF")
Reported-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325131611.545-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
The new macro set has a consistent namespace and uses C99 initializers
instead of the grufty C89 ones.
Get rid the of the local macro wrappers for consistency.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Borislav Petkov <bp@suse.de>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lkml.kernel.org/r/20200320131510.594671507@linutronix.de
Before the JZ4770, the playback and capture sampling rates had to match.
The JZ4770 supports independent sampling rates for both.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20200306222931.39664-6-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The change of offset for the {rx,tx}_threshold fields in the conf
register predates the JZ4780, and was first introduced in the JZ4760.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20200306222931.39664-5-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/wm8974.c:200:38: warning:
wm8974_aux_boost_controls defined but not used [-Wunused-const-variable=]
sound/soc/codecs/wm8974.c:204:38: warning:
wm8974_mic_boost_controls defined but not used [-Wunused-const-variable=]
commit 8a123ee2a4 ("ASoC: WM8974 DAPM cleanups")
left behind this, remove them.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200324070615.16248-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the tas2562 datasheet,the bits[5:1] represents the amp_level value.
So to set the amp_level value correctly,the shift value should be set to 1.
Signed-off-by: Jonghwan Choi <charlie.jh@kakaocorp.com>
Acked-by: Dan Murphy <dmurphy@ti.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200319140043.GA6688@jhbirdchoi-MS-7B79
Signed-off-by: Mark Brown <broonie@kernel.org>
Hello,
This small series adds audio route for built-in microphone on NVIDIA Tegra
boards that use WM8903 CODEC. In particular this is needed in order to unmute
internal microphone on Acer A500 tablet device. I'm planning to send out the
device tree for the A500 for 5.8, so will be nice to get the microphone
sorted out. Please review and apply, thanks in advance.
Dmitry Osipenko (2):
dt-bindings: sound: tegra-wm8903: Document built-in microphone audio
source
ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone
.../sound/nvidia,tegra-audio-wm8903.txt | 1 +
sound/soc/tegra/tegra_wm8903.c | 18 ++++++++++++++++++
2 files changed, 19 insertions(+)
--
2.25.1
The patch adds a property for DMIC clock rate (hz) and changes the
default to the common optimize DMIC clock rate.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200323082547.7898-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/20200321114022.8545-1-lukas.bulwahn@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The enable-GPIO needs to be toggled on a DAPM event in order to turn
microphone ON/OFF, otherwise microphone won't work.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20200320205504.30466-3-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A new small helper to get the current state of the device registration
for the given object. It'll be used for USB-audio driver to check the
delayed device registrations.
Link: https://lore.kernel.org/r/20200323170643.19181-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now CPU/Codec DAIs are alias for dais.
Thus, we can directly use for_each_rtd_dais() macro
for soc_dai_pcm_new().
This patch merge CPU/Codec for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/87r1xsolen.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/87sgi8olet.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/87tv2ooley.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/87v9n4olf4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC is currently categorizing CPU/Codec DAIs,
and it works well.
But modern devices require more complex connections,
for example Codec to Codec, etc, and future devices will
enable to more complex connections.
Because of these background, CPU/Codec DAIs categorizing is
no longer good much to modern device.
Currently, rtd has both CPU/Codec DAIs pointer.
rtd->cpu_dais = [][][][][][][][][]
rtd->codec_dais = [][][][][][][][][]
This patch merges these into DAIs pointer.
rtd->dais = [][][][][][][][][][][][][][][][][][]
^cpu_dais ^codec_dais
|--- num_cpus ---|--- num_codecs --|
Then, we can merge for_each_rtd_cpu/codec_dais() from this patch.
- for_each_rtd_cpu_dais() {
- ...
- }
- for_each_rtd_codec_dais() {
- ...
- }
+ for_each_rtd_dais() {
+ ...
+ }
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.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/87wo7kolfa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
I have a system which has an EVGA X99 Classified motherboard. The pin
assignments for the HD Audio controller are not correct under Linux.
Windows 10 works fine and informs me that it's using the Recon3Di
driver, and on Linux, `cat
/sys/class/sound/card0/device/subsystem_{vendor,device}` yields
0x3842
0x1038
This patch adds a corresponding entry to the quirk list.
Signed-off-by: Geoffrey Allott <geoffrey@allott.email>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/a6cd56b678c00ce2db3685e4278919f2584f8244.camel@allott.email
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patchset corrects a rebind issue on STM32 SPDIFRX and I2S drivers.
The same correction has already been applied for SAI driver:
0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
The following problems have been seen on STM32 drivers.
1) DMA channel is not requested:
With the sound card rebind the simplified call sequence is:
probe
snd_soc_register_component
snd_soc_try_rebind_card
snd_soc_instantiate_card
devm_snd_dmaengine_pcm_register
The problem occurs because the pcm must be registered,
before snd_soc_instantiate_card() is called.
Modify the driver, to change the call sequence as follows:
probe
devm_snd_dmaengine_pcm_register
snd_soc_register_component
snd_soc_try_rebind_card
2) DMA channel is not released:
dma_release_channel() is not called when
devm_dmaengine_pcm_release() is executed.
This occurs because SND_DMAENGINE_PCM_DRV_NAME component,
has already been released through devm_component_release().
devm_dmaengine_pcm_release() should be called before
devm_component_release() to avoid this problem.
Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from the driver, to have the right sequence.
Olivier Moysan (3):
ASoC: stm32: spdifrx: fix regmap status check
ASoC: stm32: spdifrx: manage rebind issue
ASoC: stm32: i2s: manage rebind issue
sound/soc/stm/stm32_i2s.c | 40 ++++++++++++++++------
sound/soc/stm/stm32_spdifrx.c | 64 +++++++++++++++++++----------------
2 files changed, 63 insertions(+), 41 deletions(-)
--
2.17.1
Recent addition of SoundWire stream state-machine checks in linux-next
have shown an existing issue with handling soundwire streams in codec drivers.
In general soundwire stream prepare/enable/disable can be called from either
codec/machine/controller driver. However calling it in codec driver means
that if multiple instances(Left/Right speakers) of the same codec is
connected to the same stream then it will endup calling stream
prepare/enable/disable more than once. This will mess up the stream
state-machine checks in the soundwire core.
Moving this stream handling to machine driver would fix this issue
and also allow board/platform specfic power sequencing.
Changes since v1:
- removed false error check while setting sruntime.
Srinivas Kandagatla (2):
ASoC: qcom: sdm845: handle soundwire stream
ASoC: codecs: wsa881x: remove soundwire stream handling
sound/soc/codecs/wsa881x.c | 44 +------------------------
sound/soc/qcom/Kconfig | 2 +-
sound/soc/qcom/sdm845.c | 67 ++++++++++++++++++++++++++++++++++++++
3 files changed, 69 insertions(+), 44 deletions(-)
--
2.21.0
In existing setup WSA881x codec handles soundwire stream,
however DB845c and other machines based on SDM845c have 2
instances for WSA881x codec. This will force soundwire stream
to be prepared/enabled twice or multiple times.
Handling SoundWire Stream in machine driver would fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200317151233.8763-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
There could be multiple instances of this codec on any platform,
so handling stream directly in this codec driver can lead to
multiple calls to prepare/enable/disable on the same SoundWire stream.
Move this stream handling to machine driver to fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
Corrections:
- Call snd_dmaengine_pcm_register() before snd_soc_register_component().
- Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from I2S driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200318144125.9163-4-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
Corrections:
- Call snd_dmaengine_pcm_register() before snd_soc_register_component().
- Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from SPDFIRX driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200318144125.9163-3-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
