The recent series of changes to the caching in the SSI driver have
caused a number of problems to appear in some test systems. These are
still not fully understood but we're coming up to the merge window so
for now let's revert commit 7de2763d9b (ASoC: fsl_ssi: Remove
.num_reg_defaults_raw from regmap_config) as backing that out seems to
resolve the problem on affected systems.
Reported-by: Maciej S. Szmigiero" <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Marek Vasut <marex@denx.de>
Currently there is a memory leak of module on a ENOMEM return path.
Fix this by kfree'ing module before returning.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds SNDRV_PCM_TRIGGER_SUSPEND/RESUME.
Otherwise, it breaks rsnd driver internal start/stop counter
when suspend/resume. This issue was reported/tested by Hiep
Tested-by: Hiep Cao Minh <cm-hiep@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TLV_DB_RANGE_HEAD macro was obsoleted by commit bf1d1c9b61 ("ALSA: tlv:
add DECLARE_TLV_DB_RANGE()").
This commit removes usage of the macro, with the obsoleting macro renamed
to SNDRV_CTL_TLVD_DECLARE_DB_RANGE().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TLV_DB_RANGE_HEAD macro was obsoleted by commit bf1d1c9b61 ("ALSA: tlv:
add DECLARE_TLV_DB_RANGE()").
This commit removes usage of the macro, with the obsoleting macro renamed
to SNDRV_CTL_TLVD_DECLARE_DB_RANGE().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
TLV_DB_RANGE_HEAD macro was obsoleted by commit bf1d1c9b61 ("ALSA: tlv:
add DECLARE_TLV_DB_RANGE()").
This commit removes usage of the macro, with the obsoleting macro renamed
to SNDRV_CTL_TLVD_DECLARE_DB_RANGE().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sparse reports a below warning.
sound/soc/codecs/da7219.c:804:57: warning: dubious: x & !y
The line includes a condition statement; '(a < b) & !c'. Practically, the
evaluated value of this statement equals to the value of '(a < b) && !c'.
Although, it's not an usual way to use bitwise operations as logical
operations to several conditions.
This commit fixes the bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This hardware supports only 2-channel DAI, even mono ADC digital output
has two channels with the same data.
Having min_channels=1 results in broken playback of mono files in setups
where CPU DAI supports mono.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As long as reading datasheet of STAC9766/9767, this driver includes wrong
usage of DECLARE_TLV_DB_LINEAR().
In "8.1.2. Master Volume Registers", attenuation of lineout volumes is
represented in 5 bits by -1.5 dB/step from 0 to -46.5 dB. Thus,
'master_tlv' should be dB step representation.
In "8.1.14. Record Gain", gain of volumes is represented in 4 bits by
1.5 dB/step from 0 to 22.5 dB. Thus, 'record_tlv' should be dB step
representation.
In "8.1.5. PC BEEP Volume", attenuation of volume is represented in 4 bits
by -3 dB/step from 0 to 45 dB. Thus, 'beep_tlv' should be dB step
representation.
In "8.1.7. Stereo or Mic Volume" and so on, gain of volumes is represented
in 5 bits by -1.5 dB from 12 to -34.5 dB. Thus, 'mix_tlv' should be dB
step representation.
Totally, current implementation includes misuse of TLV-related macro.
This commit replaces usage of DECLARE_TLV_DB_LINEAR() with
SNDRV_CTL_TLVD_DECLARE_DB_SCALE(), to give proper information to
applications in user land.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver has some unused variables. They should be removed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As long as reading datasheet of WM8991, this driver includes wrong usage
of DECLARE_TLV_DB_LINEAR().
In "Table 6 Input PGA Volume Range", volume is represented in 5 bits by
1.5 dB/step between -16.5/30.0 dB. Thus, 'in_pga_tlv' should be dB step
representation.
In "Table 34 LOMIX and ROMIX Volume Range", volume is represented in three
bits by -3 dB/step from 0 to -21 dB. Thus, 'out_mix_tlv' should be dB step
represenation.
In "Table 36 LOPGA, ROPGA, LOUT, ROUT and SPKVOL Volume Range", volume is
represented in 7 bits by 1 dB/step from -73 to 6 dB, including mute. Thus,
'out_pga_tlv' should be dB step representation.
In "Table 26 Digital Volume Range", volume is represented in 8 bits by
3/8 dB/step from -71.625 to 0 dB. Thus, 'out_dac_tlv' should be dB step
representation.
In "Table 16 ADC Digital Volume Range", volume is represented in 8 bits by
3/8 dB/step from -71.625 to 17.625 dB. Thus, 'in_adc_tlv' should be dB step
representation.
In "Table 23 Digital Sidetone Volume", volume is represented in 5 bits by
3 dB/step from -36 to 0 dB. Thus, 'out_sidetone_tlv' should be dB step
representation.
In "Table 12 Left Input Mixer Volume Control", volume is represented in
3 bits by 3 dB/step from -12 to 6 dB
Totally, current implementation includes misuse of TLV-related macro.
This commit replaces usage of DECLARE_TLV_DB_LINEAR() with proper macros,
to give proper information to applications in user land.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sparse reports a below warning.
tpa6130a2.c:193:33: warning: symbol 'tpa6130a2_component_driver' was not declared. Should it be static?
The symbol is just used inner the file. Forthermore, it's constant. Thus,
it's better to add static and const qualifier.
This commit adds it.
Fixes: cb7e62256e (ASoC: tpa6130a2: Register component)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sparse reports below warnings.
rt5616.c:1270:24: warning: symbol 'rt5616_aif_dai_ops' was not declared. Should it be static?
rt5616.c:1277:27: warning: symbol 'rt5616_dai' was not declared. Should it be static?
These two symbols are just used inner the file, thus it's better to add
static qualifier.
This commit adds it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
The later Arizona parts do run write sequences to power up and down the
speaker path as such a delay needs to be inserted into the DAPM sequence
to allow this to run. This patch adds appropriate delays into the
existing coalesced delay scheme.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When debugging it is useful to check the total power up/down delay that
is executed as part of the coalesced output delay. This patch adds some
debug prints for this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HP Spectre x360 with CX20724 codec has two speaker outputs while the
BIOS sets up only the bottom one (NID 0x17) and disables the top one
(NID 0x1d).
This patch adds a fixup simply defining the proper pincfg for NID 0x1d
so that the top speaker works as is.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=169071
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add channel and rate constraints for Refcap and dmiccap devices
respectively.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When operating the BE, we should print out the dai_link name of BE other
than FE. This is useful when analyzing the kernel log.
Signed-off-by: Donglin Peng <pengdonglin@smartisan.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
tlv320dac31xx is a subset of tlv320aic31xx:
- it does not have MIC inputs and ADC, thus capture is not supported,
- it has analog inputs AIN1/AIN2 that can be mixed into output.
Although tlv320dac31xx does work with tlv320aic31xx driver, this setup
does register non-existent widgets and non-existent capture stream.
Thus userspace lists non-existent objects in user interfaces, an can
access these, causing operations with device registers that are
declared as "reserved" in tlv320dac31xx datasheet.
This patch fixes this situation by separating controls/widgets/routes
into common, aic31xx-specific, and dac31xx-specific parts. Only parts
that match actual hardware (as declared in "compatible" device tree
property) are registered.
Changes from v1:
- update device tree binding documentation,
- rebased on top of "ASoC: codec duplicated callback function goes to
component on tlv320aic31xx" commit.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently if AAD is enabled in the device, during system suspend
the feature remains, regardless of whether the codec is a wake-up
source or not. This means some additional power is being used
which is unnecessary, and can causes issues with some platforms'
IRQ handlers where state changes during system suspend aren't
captured.
This patch updates the driver to disable AAD during suspend, if
we're not a wake-up source, and then re-enables this on resume.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the reset code in i2c_probe only resets the AAD part of
the device and not the entire codec. This patch updates the driver
to resolve this and ensures that if the codec is still active from
a previous boot then the audio paths are powered down prior to
reset.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the driver exposes range of sample rates between 8KHz to
192KHz as a part of dai_driver. This does not hold true as the limited
number sample rates are allowed in hw_params DAI callback. This patch
limits the sample rates exposed via dai_driver.
Signed-off-by: Vishal Thanki <vishalthanki@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename "sun4i_codec_widgets" to "sun4i_codec_controls" for
consistency with the struct field name.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Code undo operations in power enable errror path explicitly, instead of
reusing power disable path and playing with return values there.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add device tree property to define auxiliary devices to be added to
simle-audio-card.
Together with proper audio routing definition, this allows to use
simple-card in setups where separate amplifier chip is connected to
codec's output.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The booting process for the DSP is clearly separated into two parts, the
preloader brings up the core and downloads code, then the main widget
starts the code actually executing. However the shutdown sequence is all
handled with the main widget.
To allow the preloading to be run independently of the main audio bring
up it makes sense, and is generally just cleaner, for the preloader
widget to shutdown those things it initialised. This patch moves the
appropriate parts of the shutdown process into the preloader widget.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Between when we load the DSP and when it actually starts running put the
core into a lower power state where the memory is retained but nothing
is clocked.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the 1ms msleep in wm_adsp2_ena with a usleep_range, as per
normal guidance on delay functions. Also tighten up the delay a little
as 1ms was quite generous.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio component framework code has not yet landed in the i915 driver
so drop the use of the API for the time being.
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: Jeeja KP <jeeja.kp@intel.com>
We have two new Dell laptop models, they have the same ALC255 pin
definition, but not in the pin quirk table yet, as a result, the
headset microphone can't work. After adding the definition in the
table, the headset microphone works well.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, usb-line6 module exports an array of MIDI manufacturer ID and
usb-pod module uses it. However, the declaration is not the definition in
common header. The difference is explicit length of array. Although
compiler calculates it and everything goes well, it's better to use the
same representation between definition and declaration.
This commit fills the length of array for usb-line6 module. As a small
good sub-effect, this commit suppress below warnings from static analysis
by sparse v0.5.0.
sound/usb/line6/driver.c:274:43: error: cannot size expression
sound/usb/line6/driver.c:275:16: error: cannot size expression
sound/usb/line6/driver.c:276:16: error: cannot size expression
sound/usb/line6/driver.c:277:16: error: cannot size expression
Fixes: 705ececd1c ("Staging: add line6 usb driver")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is used by the Chromebook Pixel 2015.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
[john@metanate.com:
- forward-port driver from Chromium OS 3.14 tree to master
- remove wake on voice function that isn't supported by upstream rt5677
driver
- remote owner assignment in platform_driver (Evan McClain)
- convert to devm_snd_soc_register_card (Evan McClain)
- add a full copyright header based on module license and Chromium OS
Git history
]
Signed-off-by: John Keeping <john@metanate.com>
Tested-by: Genki Marshall <genki@genki.is>
Tested-by: Tom Rini <trini@konsulko.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Chromebook Pixel 2015 uses this codec with the ACPI ID RT5677CE, but
does not use the standard DT property names so add a new function to
parse the codec properties from these ACPI properties.
Also, the GPIOs are only available by index, so we need to register a
mapping to allow machine drivers to access the GPIOs by name.
Signed-off-by: John Keeping <john@metanate.com>
Tested-by: Tom Rini <trini@konsulko.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The second declaration of status is shadowing the status of a higher
scope. This uninitialized status results in garbage being returned
by the !x86_match_cpu(cpu_ids) || !iosf_mbi_available() return exit
path. Fix this by removing the extraneous second declaration of
status.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With codec read sometimes the pin_sense shows invalid monitor present
and eld_valid. Currently driver polls for few times to get the valid
ELD data.
To avoid the latency, Instead of reading ELD from codec, read it
directly from the display driver using audio component framework.
Removed the direct codec helper functions.
Signed-off-by: Sandeep Tayal <sandeepx.tayal@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trival fix, some dev_err/deb_dbg messages are missing a \n, so add it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trival fix, some pr_err messages are missing a \n, so add it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trival fix, some dev_err and deb_dbg messages are missing a \n, so
add it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trival fix, some dev_* messages are missing a \n, so add it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is simple sound card time, we could assign different codec
to a interface without making a specific driver for it. The SPDIF
and I2S interface for Samsung would be possible used by
simple-sound-card, but not sure about the PCM.
Those S3C time entries are left alone as I don't think any new board
would need them.
Signed-off-by: Randy Li <ayaka@soulik.info>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the DSP loading is split into two widgets, the preloader that
is a snd_soc_dapm_dai_link widget which starts a thread to download
the firmware, and the DSP itself which is a snd_soc_dapm_out_drv and
synchronises the thread back in to the DAPM sequence. This allows the
firmware download to be overlapped with the rest of the path bring up.
The use of a snd_soc_dapm_dai_link widget requires the preloader to be part
of the audio path in DAPM, really a supply widget is a better fit for the
preloader. The preloader is something that needs to be done for the DSP to
function, not a part of the audio path itself.
This change makes the DSP preloader widget a supply widget, which as well
as probably being a better fit will also make it much simpler to power up
the preloader widget to trigger firmware download to the core independently
of the audio path coming up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently SYSCLK is attached to every compressed DAI as this follows the
pattern of attaching clocks to the chips inputs and outputs, however, it is
really the DSP that requires the clock here. As firmware download can be a
significant part of the path startup time for these devices occasionally it
would be desirable to download the firmware in advance of the path being
brought up.
To help facilitate this early firmware loading this patch attaches the
SYSCLK to the DSP preloader widget. This also saves us adding a new route
to SYSCLK every time a new compressed DAI is created.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As part of the work to download firmware before the audio path is brought
up the DSP will be put into a low power state between downloading firmware
to the core and starting it running. This will mean that the firmware ALSA
controls are not accessible in the hardware during this period of time.
To prepare for this change we gate access to the hardware in the ALSA
control handlers on the DSP being running rather than simply booted and
move the synchronisation of the control caches out of the preloader delayed
work and into the main DAPM thread after the DSP will have been brought out
of its low power state.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the wm_adsp driver has a flag that indicates the DSP is
"running", this flag is used to gate access to the hardware. However this
flag is actually set in the firmware download thread after the firmware has
been downloaded, but this is before the core is actually started running,
so really it currently indicates that the core has been booted and is
perhaps running.
This patch clearly separates out the concepts of booted (firmware is
downloaded) and running (code is executing on the DSP) within the wm_adsp
driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After moving tpa6130a2 power management to DAPM, if chip can be physically
powered off (either reset_gpio is defined, or regulator indeed removes
power), then volume change no longer works unless chip is on due to
a running stream.
Fix that by entering regcache cache_only mode while chip is off.
Move regcache calls to tpa6130a2_power() to get them at driver init time
as well.
Signed-off-by: Nikita Yushchenko <nikita.yoush@cogentembedded.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since modules ids are generated dynamically, we do not know the id
associate with modules in another pipelines. This limits our ability to
tell DSP about neighbouring modules.
So add a table for quick referencing of allocated module ids.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Post bind parameters of KPB module contains the instance id's of
neighbouring modules in the sink path
Now that module instance ids are generated dynamically we need to update
these parameters as well, so use the table created and update the ids
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use private id's of module instances that are generated during
init_module for the IPC messages to DSP. These id's are freed
up during delete pipeline.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver needs to send unique module instance id to firmware while
creating the module and uses this id to communicate with DSP for setting
parameters while audio use case is ongoing.
But, we have upper bound of instance ID. The current IDs are coming from
topology but it doesn't know the upper bound and can't assign unique
id's subject to upper bounds as we can create a big graph but not all
parts running at same time.
This patch adds a 128bit unique id management routines which are built
on top of ffz() for faster implementation. Unfortunately ffz() works on
32bits values, so additional code is added on top of ffz() to create a
128bit unique id.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A20 has a few extra registers that the A10 doesn't have.
Therefore, use different regmaps for A10 as compared to A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the registers defined in the driver are only usable on the
A20. Rename these registers.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pop happens when mclk applied but dmic's own boot-time
Specify dmic delay times in dt to make sure
clocks are ready earlier than dmic working
Signed-off-by: Wonjoon Lee <woojoo.lee@samsung.com>
Signed-off-by: Xing Zheng <zhengxing@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DragonFly quirk added in 42e3121d90 ("ALSA: usb-audio: Add a more
accurate volume quirk for AudioQuest DragonFly") applies a custom dB map
on the volume control when its range is reported as 0..50 (0 .. 0.2dB).
However, there exists at least one other variant (hw v1.0c, as opposed
to the tested v1.2) which reports a different non-sensical volume range
(0..53) and the custom map is therefore not applied for that device.
This results in all of the volume change appearing close to 100% on
mixer UIs that utilize the dB TLV information.
Add a fallback case where no dB TLV is reported at all if the control
range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also
restrict the quirk to only apply to the volume control as there is also
a mute control which would match the check otherwise.
Fixes: 42e3121d90 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly")
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: David W <regulars@d-dub.org.uk>
Tested-by: David W <regulars@d-dub.org.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A cleanup removed a couple of members from struct snd_soc_codec_driver
after changing codec drivers to no longer use them, but one codec
was missed in the process, giving a build error:
sound/soc/codecs/cq93vc.c:134:2: error: unknown field 'controls' specified in initializer
.controls = cq93vc_snd_controls,
This moves the members from the cq93vc codec driver to its component driver
just like the other codecs already had.
Fixes: 8073aefa60 ("ASoC: remove codec duplicated callback function")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pointer callbacks of ali5451 driver may return the value at the
boundary occasionally, and it results in the kernel warning like
snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
It seems that folding the position offset is enough for fixing the
warning and no ill-effect has been seen by that.
Reported-by: Enrico Mioso <mrkiko.rs@gmail.com>
Tested-by: Enrico Mioso <mrkiko.rs@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver provides no .reg_defaults_raw in regmap_config, so
the .num_reg_defaults_raw is useless, and, in fact harmful. It
triggers kernel crash in regmap_init which tries to access the
register defaults.
Signed-off-by: Marek Vasut <marex@denx.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Same as commit ce492b3b8f
Subject: drm/fsl-dcu: use flat regmap cache
Using flat regmap cache instead of RB-tree to avoid the following
lockdep warning on driver load:
WARNING: CPU: 0 PID: 1 at kernel/locking/lockdep.c:2871 lockdep_trace_alloc+0x104/0x128
DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))
The RB-tree regmap cache needs to allocate new space on first
writes. However, allocations in an atomic context (e.g. when a
spinlock is held) are not allowed. The function regmap_write
calls map->lock, which acquires a spinlock in the fast_io case.
Since the driver uses MMIO, the regmap bus of type regmap_mmio
is being used which has fast_io set to true.
Signed-off-by: Marek Vasut <marex@denx.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Same as commit ce492b3b8f
Subject: drm/fsl-dcu: use flat regmap cache
Using flat regmap cache instead of RB-tree to avoid the following
lockdep warning on driver load:
WARNING: CPU: 0 PID: 1 at kernel/locking/lockdep.c:2871 lockdep_trace_alloc+0x104/0x128
DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))
The RB-tree regmap cache needs to allocate new space on first
writes. However, allocations in an atomic context (e.g. when a
spinlock is held) are not allowed. The function regmap_write
calls map->lock, which acquires a spinlock in the fast_io case.
Since the driver uses MMIO, the regmap bus of type regmap_mmio
is being used which has fast_io set to true.
Signed-off-by: Marek Vasut <marex@denx.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Same as commit ce492b3b8f
Subject: drm/fsl-dcu: use flat regmap cache
Using flat regmap cache instead of RB-tree to avoid the following
lockdep warning on driver load:
WARNING: CPU: 0 PID: 1 at kernel/locking/lockdep.c:2871 lockdep_trace_alloc+0x104/0x128
DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))
The RB-tree regmap cache needs to allocate new space on first
writes. However, allocations in an atomic context (e.g. when a
spinlock is held) are not allowed. The function regmap_write
calls map->lock, which acquires a spinlock in the fast_io case.
Since the driver uses MMIO, the regmap bus of type regmap_mmio
is being used which has fast_io set to true.
Signed-off-by: Marek Vasut <marex@denx.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Same as commit ce492b3b8f
Subject: drm/fsl-dcu: use flat regmap cache
Using flat regmap cache instead of RB-tree to avoid the following
lockdep warning on driver load:
WARNING: CPU: 0 PID: 1 at kernel/locking/lockdep.c:2871 lockdep_trace_alloc+0x104/0x128
DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))
The RB-tree regmap cache needs to allocate new space on first
writes. However, allocations in an atomic context (e.g. when a
spinlock is held) are not allowed. The function regmap_write
calls map->lock, which acquires a spinlock in the fast_io case.
Since the driver uses MMIO, the regmap bus of type regmap_mmio
is being used which has fast_io set to true.
Signed-off-by: Marek Vasut <marex@denx.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5663.c:3203:3-8: 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
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Acked-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warnings:
sound/soc/codecs/rt5663.c:1367:14: warning: duplicate const
sound/soc/codecs/rt5663.c:1577:5: warning:
symbol 'rt5663_button_detect' was not declared. Should it be static?
sound/soc/codecs/rt5663.c:2857:24: warning:
symbol 'rt5663_aif_dai_ops' was not declared. Should it be static?
sound/soc/codecs/rt5663.c:2866:27: warning:
symbol 'rt5663_dai' was not declared. Should it be static?
sound/soc/codecs/rt5663.c:3193:6: warning:
symbol 'rt5663_i2c_shutdown' was not declared. Should it be static?
sound/soc/codecs/rt5663.c:3200:19: warning:
symbol 'rt5663_i2c_driver' was not declared. Should it be static?
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
NO_IRQ has been == 0 on powerpc for just over ten years (since commit
0ebfff1491 ("[POWERPC] Add new interrupt mapping core and change
platforms to use it")). It's also 0 on most other arches.
Although it's fairly harmless, every now and then it causes confusion
when a driver is built on powerpc and another arch which doesn't define
NO_IRQ. There's at least 6 definitions of NO_IRQ in drivers/, at least
some of which are to work around that problem.
So we'd like to remove it. This is fairly trivial in the arch code, we
just convert:
if (irq == NO_IRQ) to if (!irq)
if (irq != NO_IRQ) to if (irq)
irq = NO_IRQ; to irq = 0;
return NO_IRQ; to return 0;
And a few other odd cases as well.
At least for now we keep the #define NO_IRQ, because there is driver
code that uses NO_IRQ and the fixes to remove those will go via other
trees.
Note we also change some occurrences in PPC sound drivers, drivers/ps3,
and drivers/macintosh.
Signed-off-by: Michael Ellerman <mpe@ellerman.id.au>
sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.
NULL check before some freeing functions is not needed.
Based on checkpatch warning
"kfree(NULL) is safe this check is probably not required"
and kfreeaddr.cocci by Julia Lawall.
Generated by: scripts/coccinelle/free/ifnullfree.cocci
CC: Andrej Krutak <dev@andree.sk>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We must do it this way, because e.g. POD X3 won't play any sound unless
the host listens on the bulk EP, so we cannot export it only via libusb.
The driver currently doesn't use the bulk EP messages in other way,
in future it could e.g. sense/modify volume(s).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only initialize PCM for POD HD devices that support it.
No POD HD seems to support MIDI, thus drop the initialization.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes audio in/out and basic initialization via control EP (emulates
what original driver does). The initialization is done similarly to original
POD, firmware and serial IDs are read and exported via sysfs.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all PODs use MIDI via USB data interface, thus allow avoiding
that code and instead using direct processing.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
POD X3 can initialize similarly to older PODs, but it doesn't have the MIDI
interface. Instead, configuration is done via proprietary bulk EP messages.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. POD X3 seems to require playback data to be sent to it to generate
capture data. Otherwise the device stalls and doesn't send any more capture
data until it's reset.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown