The extcon driver takes the DAPM mutex from within the interrupt thread
in several places, which makes it possible to get into a situation where
the interrupt thread is blocked waiting on the DAPM mutex whilst a DAPM
sequence is running which is attempting to configure the FLL. In this
case the FLL completion can't be completed as as the IRQ handler is
ONE_SHOT, which cause the FLL lock to use the full time out (250mS) and
report that the process timed out.
It is not really practical to make the extcon driver not take the DAPM
mutex from within the interrupt thread, at least not without extensive
modification. So this patch fixes the issue by switching the wait for
the FLL lock to polling. A few fast polls are done first as the FLL
should lock quickly for a good quality reference clock, (indeed it hits
on the first poll on my system) and it will poll every 20mS after that
until it times out.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
If there don't exist dynamic sink or source widget, it will failed to
add dynamic path.
"AIF3ADCDAT" is snd_soc_dapm_aif_out, can't be dynamic sink widget. So
change the audio route to fix this issue.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no use of snd_soc_unregister_card in remove function
as devm_snd_soc_register_card in probe function automatically
handles it. So, remove use of snd_soc_unregister_card and with
this change remove arndale_audio_remove as it is now redundant.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The topology code merged in the v4.2 merge window introduced a new ABI
which was believed to be suitable for use but subsequently additional
work by the developers of this feature have revealed some problems that
need to be addressed. In order to allow this to be done without having
to support the initial ABI add Kconfig to disable the build and also add
some #error statements to the UAPI header so users can't use them.
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Merge tag 'asoc-v4.2-disable-topology' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Disable topology support for v4.2
The topology code merged in the v4.2 merge window introduced a new ABI
which was believed to be suitable for use but subsequently additional
work by the developers of this feature have revealed some problems that
need to be addressed. In order to allow this to be done without having
to support the initial ABI add Kconfig to disable the build and also add
some #error statements to the UAPI header so users can't use them.
The wrong register was used to set the gain of ref loop, when changing
the FLL output on an active FLL. This patch corrects the offset of the
gain register.
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The fix for deadlock in PM in commit [1ee23fe07e: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the topology code to be compiled out so that users who don't need
topology don't need to havve the code compiled in, saving them some
memory.
Some more configuration could be added to remove some of the hooks into
the core data structures but that is probably best done with some
refactoring to use functions to do the updates of the data structures
rather than ifdefing in the code as we'd need to do at the minute.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't set .read_flag_mask for adav803, it's for adav801 only.
Fixes: 0c2d696456 ("ASoC: adav80x: Split SPI and I2C code into different modules")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Dell laptop causes the white noise by login screen and headphone,
and the fixup function ALC292_FIXUP_DISABLE_AAMIX can eliminate this
noise.
Codec: Realtek ALC3235
Vendor Id: 0x10ec0293
Subsystem Id: 0x102806db
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1484334
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power for line out was not turned on when line out is enabled.
So we add "LOUT amp" widget to turn on the power for line out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Enforce correct device sequencing when configuring a new
audio route when there is an existing active audio route(s).
This patch fixed recording noise issue while playback is active.
We have some registers which require the device to be in full shutdown
or to enter full shutdown before the register settings will take effect.
Currently the driver is not shutting down the device when a new audio
route is created. If a new audio route is made active while there is
already an active audio route, then the required register sequencing is
violated. A hardware shutdown toggle when creating a new audio route
corrects the sequencing error. The device must remain in hardware
shutdown for 40ms to allow the internal hardware core to fully shutdown.
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Acked-by: Anish Kumar <anish.kumar@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are a couple of small driver specific fixes here but the
overwhelming bulk of these changes are fixes to the topology ABI that
has been newly introduced in v4.2. Once this makes it into a release we
will have to firm this up but for now getting enhancements in before
they've made it into a release is the most expedient thing.
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Merge tag 'asoc-fix-v4.2-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.2
There are a couple of small driver specific fixes here but the
overwhelming bulk of these changes are fixes to the topology ABI that
has been newly introduced in v4.2. Once this makes it into a release we
will have to firm this up but for now getting enhancements in before
they've made it into a release is the most expedient thing.
Currently the TLV topology structure is targeted at only supporting the
DB scale data. This patch extends support for the other TLV types so they
can be easily added at a later stage.
TLV structure is moved to common topology control header since it's a
common field for controls and can be processed in a general way.
Users must set a proper access flag for a control since it's used to
decide if the TLV field is valid and if a TLV callback is needed.
Removed the following fields from topology TLV struct:
- size/count: type can decide the size.
- numid: not needed to initialize TLV for kcontrol.
- data: replaced by the type specific struct.
Added TLV structure to generic control header and removed TLV structure
from mixer control.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A lot of small fixes here, a few to the core:
- Fix for binding DAPM stream widgets on devices with prefixes assigned
to them
- Minor fixes for the newly added topology interfaces
- Locking and memory leak fixes for DAPM
- Driver specific fixes
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Merge tag 'asoc-fix-v4.2-rc3' into asoc-fix-topology
ASoC: Fixes for v4.2
A lot of small fixes here, a few to the core:
- Fix for binding DAPM stream widgets on devices with prefixes assigned
to them
- Minor fixes for the newly added topology interfaces
- Locking and memory leak fixes for DAPM
- Driver specific fixes
Some widgets may need sorting within, So add this support in topology.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The fixup ALC292_FIXUP_DISABLE_AAMIX can fix the white noise of
the headphone on this Dell machine.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on from it.
Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.
As long as I investigated, AudioFire2/AudioFire4/AudioFirePre8 have a
quirk to transfer a first packet with 0x02 in its dbc field. This causes
ALSA Fireworks driver to detect discontinuity. In this case, firmware
version 5.7.0, 5.7.3 and 5.8.0 are used.
Payload CIP CIP
quadlets header1 header2
02 00050002 90ffffff <-
42 0005000a 90013000
42 00050012 90014400
42 0005001a 90015800
02 0005001a 90ffffff
42 00050022 90019000
42 0005002a 9001a400
42 00050032 9001b800
02 00050032 90ffffff
42 0005003a 9001d000
42 00050042 9001e400
42 0005004a 9001f800
02 0005004a 90ffffff
(AudioFire2 with firmware version 5.7.)
$ dmesg
snd-fireworks fw1.0: Detect discontinuity of CIP: 00 02
These models, AudioFire8 (since Jul 2009 ) and Gibson Robot Interface
Pack series uses the same ARM binary as their firmware. Thus, this
quirk may be observed among them.
This commit adds a new member for AMDTP structure. This member represents
the value of dbc field in a first AMDTP packet. Drivers can set it with
a preferred value according to model's quirk.
Tested-by: Johannes Oertei <johannes.oertel@uni-due.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9c6893e0be.
The fix is superseded by the next commit as a better implementation
for supporting AudioFire2/AudioFire4/AudioFirePre8 quirks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_hdac_chip_readl return can never be less than zeros,
so no point in checking for the return value
This fixes following static checker warnings in
snd_hdac_ext_bus_parse_capabilities
sound/hda/ext/hdac_ext_controller.c:47
snd_hdac_ext_bus_parse_capabilities()
warn: unsigned 'offset' is never less than zero.
sound/hda/ext/hdac_ext_controller.c:54
snd_hdac_ext_bus_parse_capabilities()
warn: unsigned 'cur_cap' is never less than zero.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes issue in assigning host stream in case of
decoupled mode. The check to verify if the stream is already
in use was wrong so fix that
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I2S2_DAC pin can be used for I2S or GPIO. We should set it as GPIO
if we use GPIO5 as DMIC1 data pin.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This fixes the following warning, that is seen with gcc 5.1:
warning: logical not is only applied to the left hand side of comparison [-Wlogical-not-parentheses].
Signed-off-by: Tomer Barletz <barletz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MacBook Pro 5,2 with ALC889 codec had already a fixup entry, but this
seems not working correctly, a fix for pin NID 0x15 is needed in
addition. It's equivalent with the fixup for MacBook Air 1,1, so use
this instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102131
Reported-and-tested-by: Jeffery Miller <jefferym@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PM ops could be triggered before HDA is done initializing
and cause PM to set HDA controller to D3Hot. This can result
in "CORB reset timeout#2, CORBRP = 65535" and "no codecs
initialized". Additionally, PM ops can be triggered before
azx_probe_continue finishes (async probe). This can result
in a NULL deref kernel crash.
To fix this, avoid PM ops if !chip->running.
Signed-off-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add initialization for sst_byt.dev at init stage, which fix the
'NULL device *' warning issues.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add initialization for sst_hsw.dev at init stage, which fix the
'NULL device *' warning issues.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2.
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Vostro 5480 (1028:069a) needs the very same quirk used for Vostro
5470 model to make bass speakers properly working.
Reported-and-tested-by: Paulo Roberto de Oliveira Castro <p.oliveira.castro@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new Dell laptop with codec 256 can't detect headset mic when
headset was inserted on the machine. From alsa-info, we check
init_pin_configs and need to define the new register value for pin
0x1d & 0x1e because the original macro ALC256_STANDARD_PINS can't
match pin definition. Also, the macro ALC256_STANDARD_PINS is
simplified by removing them. This makes headset mic works on laptop.
Codec: Realtek ALC256
Vendor Id: 0x10ec0256
Subsystem Id: 0x102806f2
BugLink: https://bugs.launchpad.net/bugs/1478497
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Toshiba Satellite S50D has another model with a different PCI SSID
(1179:fa93) while the previous fixup was for 1179:fa91. Adjust the
fixup entry with SND_PCI_QUIRK_MASK() to match with both devices.
Reported-by: Tim Sample <timsample@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on.
Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.
AudioFire2 has a quirk to transfer a first packet with non-zero in
its dbc field. This causes ALSA Fireworks driver to detect discontinuity.
As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk.
This commit adds a support for the quirk to handle AudioFire2 packets.
For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of
AudioFire2's firmwares.
02 00050002 90ffffff <-
42 0005000a 90013000
42 00050012 90014400
42 0005001a 90015800
02 0005001a 90ffffff
42 00050022 90019000
42 0005002a 9001a400
42 00050032 9001b800
02 00050032 90ffffff
42 0005003a 9001d000
42 00050042 9001e400
42 0005004a 9001f800
02 0005004a 90ffffff
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headset was plugged in the Dell desktop, the mic of headset
can't be detected and workable.
According to the alsa-info, we found the differece between alsa and
init_pin_configs on the machine, so we need to add the pin configs to
make headset work.
Codec: Realtek ALC3234
Vendor Id: 0x10ec0255
Subsystem Id: 0x102806bb
BugLink: https://bugs.launchpad.net/bugs/1477900
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch complains that we have nested checks for "spdif_present". It
turns out the current behavior isn't correct, we should remove the first
check and keep the second.
Fixes: 1077a02481 ('ALSA: hda - Use generic parser for Cirrus codec driver')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot of small fixes here, a few to the core:
- Fix for binding DAPM stream widgets on devices with prefixes assigned
to them
- Minor fixes for the newly added topology interfaces
- Locking and memory leak fixes for DAPM
- Driver specific fixes
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Merge tag 'asoc-fix-v4.2-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.2
A lot of small fixes here, a few to the core:
- Fix for binding DAPM stream widgets on devices with prefixes assigned
to them
- Minor fixes for the newly added topology interfaces
- Locking and memory leak fixes for DAPM
- Driver specific fixes
The settings in current code does not match the datasheet, fix it.
DAC Control - Address 03h
DAC Digital Interface Format (Bits 5:4)
DAC_DIF1 DAC_DIF0 Description
0 0 Left Justified, up to 24-bit data (default)
0 1 I²S, up to 24-bit data
1 0 Right-Justified, 16-bit Data
1 1 Right-Justified, 24-bit Data
Transmitter Control 2 - Address 12h
Transmitter Digital Interface Format (Bits 7:6)
Tx_DIF1 Tx_DIF0 Description Format Figure
0 0 Left Justified, up to 24-bit data (default)
0 1 I²S, up to 24-bit data
1 0 Right-Justified, 16-bit Data
1 1 Right-Justified, 24-bit Data
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The de-emphasis sampling rate selection is controlled by BIT[3:4] of
PCM1681_DEEMPH_CONTROL register. Do proper left shift to set it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
