| Age | Commit message (Collapse) | Author | Files | Lines |
|
For the HP Pavilion 15-CK0xx, with audio subsystem ID 0x103c:0x841c,
adding a line in patch_realtek.c to apply the ALC269_FIXUP_HP_MUTE_LED_MIC3
fix activates the mute key LED.
Signed-off-by: Carlos M <carlos.marr.pz@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210531202026.35427-1-carlos.marr.pz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Previously this fix was applied only to Bullseye variant laptops,
and should be applied to Cyborg and Warlock variants.
Fixes: 45b14fe200ba ("ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210531163754.136736-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This commit takes ALSA fireface driver to perform sequence replay for
media clock recovery.
The protocol specific to RME Fireface series is not compliant to
IEC 61883-1/6 since it has no CIP header, therefore presentation time
is not used for media clock recovery. The sequence of the number of data
blocks per packet is important.
I note that the device skips an isochronous cycle corresponding to an
empty packet or a NODATA packet in blocking transmission method of
IEC 61883-1/6. For sequence replay, the cycle is handled as receiving an
empty packet. Furthermore, it doesn't start packet transmission till
receiving any packet.
The sequence replay is tested with below models:
* Fireface 400
* Fireface 800
* Fireface 802
I note that it is better to initialize Fireface 400 in advance by
initialization transaction implemented in snd-fireface-ctl-service of
snd-firewire-ctl-services project. You can see whether initialized or
not by HOST LED on the device. Unless, the device often stops packet
transmission even if session starts.
I guess the sequence replay also works well with below models:
* Fireface UFX
* Fireface UCX
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This commit takes ALSA firewire-tascam driver to perform sequence replay
for media clock recovery.
The protocol specific to Tascam FireWire series is not compliant to
IEC 61883-1/6 in terms of syt field of CIP. The protocol doesn't use
presentation time in received CIP for playback timing. The sequence of
the number of data blocks per packet is important for media clock
recovery.
Although the devices in Tascam FireWire series transfer packets
regardless of receiving packets, the tx packets includes no events
in the beginning of streaming. It takes so long to multiplex any event
into the packet after receiving the sequence of packets. As long as I
experienced, it takes several thousands of isochronous cycle. Furthermore,
just after changing sampling transmission frequency, it stops multiplexing
event at once, then starts multiplexing again.
The sequence replay is tested with below models:
* FW-1884
* FW-1804
* FW-1082
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This commit takes ALSA firewire-digi00x driver to perform sequence replay
for media clock recovery.
All of models in Digidesign digi00x family don't transfer isochronous
packets till receiving isochronous packets. The on-the-fly mode is used
for the purpose. They don't interpret presentation time expressed in syt
field of received CIP, therefore the sequence of the number of data blocks
per packet is important for media clock recovery.
The sequence replay is tested with below models:
* Digidesign Digi 002
* Digidesign Digi 002 Rack
* Digidesign Digi 003
* Digidesign Digi 003 Rack
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This commit takes ALSA oxfw driver to perform sequence replay for media
clock recovery. Unfortunately, OXFW970 ASIC and its firmware has a quirk
called jumbo payload which skips several isochronous cycles for packet
transmission, thus the sequence replay is just adopted to OXFW971 ASIC.
As well as Fireworks, OXFW ASICs also ignores presentation time against
the way in IEC 61883-1/6.
The sequence replay is tested with below models:
* Tascam FireOne
* Stanton Magnetics SCS.1m
* Apogee Duet FireWire
For below models, the sequence replay is tested to be disabled:
* Griffin FireWave
* Behringer F-Control Audio 202
* Loud Technology Tapco Link.FireWire 4x6
* Loud Technology Mackie Onyx Satellite
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Echo Digital Audio Corporation had US patent US7599388B2 titled as
'System and method for high-bandwidth serial bus data transfer'. In the
patent, dual-banked shared memory is used to deliver data between
serial bus transmission and processor in FIFO way. The patent seems to be
used for Fireworks board module. The mechanism is not compliant to
synchronization based on presentation time expressed in syt field
of CIP header. Fireworks board module takes care of the sequence of
the number of data blocks per packet and just ignores the value of syt
field.
This commit takes fireworks driver to performs sequence replay for media
clock recovery. As long as I tested, Audiofire 2 and 4 have a quirk to
skip an isochronous cycle several thousands after starting packet
transmission.
The sequence replay is tested with below models:
* Loud Technology Mackie 400f
* Echo Audio Audiofire 12 (DSP model)
* Echo Audio Audiofire 12 (FPGA model)
* Echo Audio Audiofire 8 (DSP model)
* Echo Audio Audiofire 8 (FPGA model)
* Echo Audio Audiofire Pre8
* Echo Audio Audiofire 4
* Echo Audio Audiofire 2
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In the design of Fireworks board module, the device does't adjust its
media clock voluntarily by the sequence of presentation time expressed in
syt field of CIP header of received packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The recent commit to drop the HDA-specific mute-LED control,
e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the
audio LED trigger"), caused a regression on the mixer element read for
"Capture Switch" when it's built from bind controls. The function
create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly
via snd_hda_gen_add_kctl() instead of add_control(). Although the
commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in
add_control() as default, this code path fell out from the radar. As
a result, now the driver gives -EPERM error because of the lack of the
proper access bit at reading "Capture Switch" element value.
Fix the regression by setting the access bit properly.
Fixes: e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634
Link: https://lore.kernel.org/r/20210531180633.27831-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We need the tty/serial fixes in here as well.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
We need the driver core fixes in here as well.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
Add HD Audio PCI ID for Intel AlderLake-M. Add rules to
snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with
PCH-DMIC or Soundwire codecs, and legacy driver for the rest.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Function 'snd_usb_endpoint_suspend' is declared twice, so remove the
repeated declaration.
Signed-off-by: Shaokun Zhang <zhangshaokun@hisilicon.com>
Link: https://lore.kernel.org/r/1622278926-63857-1-git-send-email-zhangshaokun@hisilicon.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Pull sound fixes from Takashi Iwai:
"A slightly high volume at this time due to pending ASoC fixes.
While there are a few generic simple-card fixes for regressions, most
of the changes are device-specific fixes: ASoC Intel SOF, codec
clocks, other codec / platform fixes as well as usual HD-audio and
USB-audio"
* tag 'sound-5.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (37 commits)
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook Fury 17 G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook Fury 15 G8
ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Zbook G8
ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
ALSA: usb-audio: scarlett2: snd_scarlett_gen2_controls_create() can be static
ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i
ALSA: hda/realtek: Headphone volume is controlled by Front mixer
ALSA: usb-audio: scarlett2: Improve driver startup messages
ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci
ALSA: usb-audio: fix control-request direction
ASoC: qcom: lpass-cpu: Use optional clk APIs
ASoC: cs35l33: fix an error code in probe()
ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware
ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency
ASoC: cs42l52: Minor tidy up of error paths
ASoC: cs35l32: Add missing regmap use_single config
ASoC: cs35l34: Add missing regmap use_single config
ASoC: cs42l73: Add missing regmap use_single config
ASoC: cs53l30: Add missing regmap use_single config
...
|
|
Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR,
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20210526121828.8460-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Drivers of ALSA firewire stack can process packets for IT/IR context in
process context when the process operates ALSA PCM character device by
calling ioctl(2) with some requests. The ioctl requests are:
* SNDRV_PCM_IOCTL_HWSYNC
* SNDRV_PCM_IOCTL_SYNC_PTR
* SNDRV_PCM_IOCTL_REWIND
* SNDRV_PCM_IOCTL_FORWARD
* SNDRV_PCM_IOCTL_WRITEI_FRAMES
* SNDRV_PCM_IOCTL_READI_FRAMES
* SNDRV_PCM_IOCTL_WRITEN_FRAMES
* SNDRV_PCM_IOCTL_READN_FRAMES
This means that general application can process PCM frames apart from
hardware IRQ invocation, even if they are programmed by either IRQ-based
scheduling model or Timer-based scheduling model.
This commit add support for Timer-based scheduling model by allowing
PCM runtime to suppress both process wakeup per period and scheduling
hardware IRQ.
SNDRV_PCM_INFO_BATCH is obsoleted since ALSA IEC 61883-1/6 packet streaming
engine can report the number of transferred PCM frames within PCM period
boundary. The granularity equals to SYT_INTERVAL in blocking transmission.
In non-blocking transmission, it doesn't equal to SYT_INTERVAL but doesn't
exceed.
This patch is tested with PulseAudio, and --sched-model option of axfer
with fix against the issue reported at:
* https://lore.kernel.org/alsa-devel/687f9871-7484-1370-04d1-9c968e86f72b@linux.intel.com/#r
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527123253.174315-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Models in below series start transmission of packet after receiving the
sequence of packets:
* Digidesign Digi00x family
* RME Fireface series
Additionally, models in Tascam FireWire series start multiplexing PCM
frames into packets enough after receiving packets. It's required to
transfer packets on-the-fly for the above models according to nominal
sampling transfer frequency before starting sequence replay.
This commit allows drivers to decide whether the engine transfers packet
on-the-fly or not.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
ALSA IEC 61883-1/6 packet streaming engine uses pre-computed parameters
ideal for nominal sampling transfer frequency (STF) to transfer packets
to device since it was added 2011. As a result of user experience for a
decade, it is clear that the sequence is not suitable to some actual
devices. It takes the devices to generate noise, and causes any type of
discontinuity in the series of packet transferred from the device. It's
required for the engine to transfer packets according to effective STF.
The effective STF is given by media clock recovered by the sequence of
packet transferred from the target device. In the previous commit, the
sequence is already cached. The media clock recovery can be achieved by
analyzing the sequence.
In technological world, many ideas are proposed for media clock recovery.
However, the small part of them could be actually adopted in our case
since floating point arithmetic is not mostly available in Linux kernel
land.
This commit adopts the simple way from them; sequence replay, which means
that the sequence of parameters from incoming packet is used as is to
transfer outgoing packets. The media clock is not computed internally,
but the sequence of outgoing packet superficially looks to be generated by
the media clock.
The association between source and destination is decided when starting
AMDTP domain. When the target device supports a pair of isochronous packet
streams, the tx stream is source and the rx stream is destination. When it
supports two pair of streams, each of tx stream is associated to
corresponding rx stream in its order. When it supports less number of tx
streams than rx streams, the fist tx stream is selected for all of rx
streams. When it supports more tx streams than rx streams, the first tx
packet is associated to the rx stream.
As I noted in previous commit, the sequence of parameters from incoming
packet is different between devices, time to time. It is worse idea to
replay the sequence of parameters from a device for the sequence of
packet to the other devices even if they are in the same category of
device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In design of audio and music unit in IEEE 1394 bus, feedback of
effective sampling transfer frequency (STF) is delivered by packets
transferred from device. The devices supported by ALSA firewire stack
are categorized to three groups regarding to it.
* Group 1:
* Echo Audio Fireworks board module
* Oxford Semiconductor OXFW971 ASIC
* Digidesign Digi00x family
* Tascam FireWire series
* RME Fireface series
* Group 2:
* BridgeCo. DM1000/DM1100/DM1500 ASICs for BeBoB solution
* TC Applied Technologies DICE ASICs
* Group 3:
* Mark of the Unicord FireWire series
In group 1, the effective STF is determined by the sequence of the number
of events per packet. In group 2, the sequence of presentation timestamp
expressed in syt field of CIP header is interpreted as well. In group 3,
the presentation timestamp is expressed in source packet header (SPH) of
each data block.
I note that some models doesn't take care of effective STF with large
internal buffer. It's reasonable to name it as group 0:
* Group 0
* Oxford Semiconductor OXFW970 ASIC
The effective STF is known to be slightly different from nominal STF for
all of devices, and to be different between the devices. Furthermore, the
effective STF is known to be shifted for long-period transmission. This
makes it hard for software to satisfy the effective STF when processing
packets to the device.
The effective STF is deterministic as a result of analyzing the batch of
packet transferred from the device. For the analysis, caching the sequence
of parameter in the packet is required.
This commit adds an option so that AMDTP domain structure takes AMDTP
stream structure to cache the sequence of parameters in packet transferred
from the device. The parameters are offset ticks of syt field against the
cycle to receive the packet and the number of data blocks per packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
./sound/soc/fsl/imx-rpmsg.c:140:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Zou Wei <zou_wei@huawei.com>
Message-Id: <1622113652-56646-1-git-send-email-zou_wei@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after
the DAPM widgets were added, and these powers were set by the JD settings
"RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function,
these powers were ignored to prevent them controlled by DAPM.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Message-Id: <15fced51977b458798ca4eebf03dafb9@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
rsnd_ssiu_busif_err_irq_ctrl() has very similar duplicated code.
This patch merge and tidyup the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87sg28lwxw.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
rsnd_ssiu_busif_err_status_clear() has very similar duplicated code.
This patch merge and tidyup the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87tumolwy3.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
BUSIF is SSIU feature, but its related code is
implemented at ssi.c today.
This patch moves it to ssiu.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87v974lwy9.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Renesas Sound uses many modules (SSI/SSIU/SRC/CTU/MIX/DVC/DMA),
and supports complex connections/path.
Thus each modules needs to save its status to correctly control it.
This status is updated when by .trigger, and .hw_params/.hw_free.
Renesas Sound is protecting modules by using lock when .trigger,
but it was not enough to protecting each modules "status" if it was
used from many paths.
1) .hw_params/.hw_free update status
2) another doesn't update status, but overwrites by same value
This patch do
1) protects .hw_params/.hw_free by lock
2) do nothing if no status update
Without this patch, protected mod->status (= .trigger) might be
overwrote by non protected mod->status (= .hw_params / .hw_free),
and in such case, CTU/MIX/DVC/SSIU/SSI which are used from
many paths might get damage.
If above issue happens, Renesas Sound will be hung (= silence)
and never be recoverd.
I could reproduce this issue by continue playing very short sound
with loop very long term (3-4 hours) through 2 inputs (= MIXer).
For updating rsnd_status_update(), this patch removes rsnd_dai_call()
debug message. Because we already have debugfs support, and is not
good match to new code.
Reported-by: Linh Phung T. Y <linh.phung.jy@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87wnrklwyh.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current rsnd supports #define DEBUG, but it is not helpful
if issue happen after 4-5 hours.
This patch adds debugfs support for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87y2c0lwyn.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current rsnd is using dev_dbg() if irq error happen,
but it makes debug very difficult if some strange things happen.
This patch uses dev_info() for it, and rename the macro name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <87zgwglwyv.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current rsnd_dai_call() doesn't indicate error message,
thus it is very difficult to know the issue
when strange things happen.
This patch indicates error for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <871r9snbji.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current ssi.c checks BUSIF when TDM mode, but it should be checked
everytime.
This patch do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <8735u8nbjr.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
SSIU is not needed if SSI was PIO mode.
This patch ignores such case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <874keonbkg.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
runtime might be NULL. Let's ignore such case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <875yz4nbkt.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
commit d6956a7dde6fb ("ASoC: rsnd: add null CLOCKIN support")
added null_clk, but it is using local static valuable.
It will be leaked if rsnd driver was removed.
This patch moves it to priv, and call unregister when removing.
Fixes: d6956a7dde6fb ("ASoC: rsnd: add null CLOCKIN support")
Link: https://lore.kernel.org/r/87tumsoe2p.wl-kuninori.morimoto.gx@renesas.com
Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Message-Id: <877djknbl5.wl-kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Don't populate the const array dsp_dma_stream_ids the stack but instead
make it static. Makes the object code smaller by 21 bytes.
Before:
text data bss dec hex filename
189012 70376 192 259580 3f5fc ./sound/pci/hda/patch_ca0132.o
After:
text data bss dec hex filename
188927 70440 192 259559 3f5e7 ./sound/pci/hda/patch_ca0132.o
(gcc version 10.3.0)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210526160616.3764119-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Sparse throws the following warning:
sound/pci/lx6464es/lx_core.c:677:34: error: self-comparison always
evaluates to false
This comparison and error message make no sense, let's remove them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Sparse throws the following warning:
sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always
evaluates to false
This is likely a 16+ year old confusion between vp2 and vp.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Remove superfluous "break", as there is a "return" before them.
Signed-off-by: zuoqilin <zuoqilin@yulong.com>
Link: https://lore.kernel.org/r/20210527030445.1201-1-zuoqilin1@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The HP ZBook Studio 17.3 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-4-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The HP ZBook Fury 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-3-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The HP ZBook Studio 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-2-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs
ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the
fixup, the mute/micmute LEDs work good.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Message-Id: <20210524114239.7960-1-yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Revise variable names and comments in
cs35l35 and cs35l36 amp drivers.
Signed-off-by: David Rhodes <drhodes@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Message-Id: <20210525194439.2232908-1-drhodes@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Message-Id: <20210524114017.18672-1-yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Message-Id: <20210524114753.39544-1-yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Message-Id: <20210524114503.26460-1-yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces
bug plock/pop noise over line out (green jack on the back) which can be
fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine.
Convert the existing entry for the same SSID to chain to apply this fixup
as well.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210524203726.2278-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The sparse tool complains as follows:
sound/soc/fsl/imx-card.c:121:27: warning:
symbol 'ak4458_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:138:31: warning:
symbol 'ak4458_tdm_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:149:27: warning:
symbol 'ak4497_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:166:27: warning:
symbol 'ak5558_fs_mul' was not declared. Should it be static?
sound/soc/fsl/imx-card.c:180:31: warning:
symbol 'ak5558_tdm_fs_mul' was not declared. Should it be static?
Those symbols are not used outside of imx-card.c, so marks
them static.
Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Message-Id: <20210524133553.2366502-1-weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@sirena.org.uk>
|
|
Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Message-Id: <20210524115131.46288-1-yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@sirena.org.uk>
|
|
Jack must be connected before reporting button events and
if the jack is disconnected button release must be reported
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-Id: <20210525090822.64577-4-tanureal@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@sirena.org.uk>
|
|
Add support for 2.4MHz clock source
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-Id: <20210525090822.64577-3-tanureal@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@sirena.org.uk>
|