Cppcheck complains about a possible NULL pointer dereference but it
actually looks like the NULL assignment is not needed (same loop is
used in other parts of the file without it).
ALSA: core: pcm_memory: dereference pointer after NULL checks
Fix cppcheck warnings:
sound/core/pcm_memory.c:380:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:384:6: note: Assuming that condition
'!substream' is not redundant
if (PCM_RUNTIME_CHECK(substream))
^
sound/core/pcm_memory.c:380:26: note: Null pointer dereference
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:433:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:436:6: note: Assuming that condition
'!substream' is not redundant
if (PCM_RUNTIME_CHECK(substream))
^
sound/core/pcm_memory.c:433:26: note: Null pointer dereference
struct snd_card *card = substream->pcm->card;
^
Fix cppcheck, the fallthrough only makes sense within the conditional
block
sound/core/memalloc.c:161:3: style:inconclusive: Statements following
return, break, continue, goto or throw will never be
executed. [unreachableCode]
fallthrough;
^
Connor McAdams [Tue, 25 Aug 2020 20:10:38 +0000 (16:10 -0400)]
ALSA: hda/ca0132 - Add AE-7 custom controls.
Add headphone gain and DAC filter controls, which use the same commands
as the AE-5. Also, change input source enumerated control item count to
exclude front microphone.
Connor McAdams [Tue, 25 Aug 2020 20:10:26 +0000 (16:10 -0400)]
ALSA: hda/ca0132 - Clean up ca0132_alt_out_select.
Remove the output structures that were in use before and instead set the
DSP commands line by line. Now that the commands use is known, it makes
the functionality more clear this way.
Remove the surround output selection and merge it with the speaker
output selection. Now that the extra commands that were being run on
surround output setting are known, there's no need to have it be
separate.
Connor McAdams [Tue, 25 Aug 2020 20:10:24 +0000 (16:10 -0400)]
ALSA: hda/ca0132 - Add bass redirection controls.
Add bass redirection controls for surround outputs. This uses the DSP to
redirect audio below the bass redirection crossover frequency to the LFE
channel from the front/rear L/R speakers. This only goes into effect if
the speakers aren't set as full range, and only if the surround
configuration has an LFE channel.
Add functions for setting full-range speakers and controls to
enable/disable the setting. Setting a speaker to full-range means that
the channels won't have their bass redirected to the LFE channel.
Add a surround channel configuration enumeration control. Setting up
different channel configurations allows the DSP to upmix stereo audio
into multi-channel audio, and allows for redirection of bass to a
subwoofer.
Takashi Iwai [Sun, 23 Aug 2020 11:32:51 +0000 (13:32 +0200)]
ALSA: usb-audio: Correct wrongly matching entries with audio class
The commit 51ab5d77dcb8 ("ALSA: usb-audio: Properly match with audio
interface class") converted the quirk entries that have both vid/pid
pair and bInterface fields to match with all those with a new macro
USB_AUDIO_CLASS(). However, it turned out that those are false
conversions; all those (but the unknown KeithMcMillen device) are
actually with vendor-specific interface class, hence the conversions
broke the matching.
This patch corrects those entries to the right one,
USB_DEVICE_VENDOR_SPEC() (and USB_DEVICE() for KeithMcMillen to be
sure), and drop the unused USB_AUDIO_CLASS macro again.
Takashi Sakamoto [Sun, 23 Aug 2020 07:55:45 +0000 (16:55 +0900)]
ALSA: firewire-digi00x: exclude Avid Adrenaline from detection
Avid Adrenaline is reported that ALSA firewire-digi00x driver is bound to.
However, as long as he investigated, the design of this model is hardly
similar to the one of Digi 00x family. It's better to exclude the model
from modalias of ALSA firewire-digi00x driver.
This commit changes device entries so that the model is excluded.
Takashi Sakamoto [Sun, 23 Aug 2020 07:55:37 +0000 (16:55 +0900)]
ALSA; firewire-tascam: exclude Tascam FE-8 from detection
Tascam FE-8 is known to support communication by asynchronous transaction
only. The support can be implemented in userspace application and
snd-firewire-ctl-services project has the support. However, ALSA
firewire-tascam driver is bound to the model.
This commit changes device entries so that the model is excluded. In a
commit 53b3ffee7885 ("ALSA: firewire-tascam: change device probing
processing"), I addressed to the concern that version field in
configuration differs depending on installed firmware. However, as long
as I checked, the version number is fixed. It's safe to return version
number back to modalias.
Takashi Iwai [Mon, 17 Aug 2020 08:21:40 +0000 (10:21 +0200)]
ALSA: usb-audio: Properly match with audio interface class
There are a few entries in the quirk table that set the device ID with
USB_DEVICE() macro while having an extra bInterfaceClass field. But
bInterfaceClass field is never checked unless the proper match_flags
is set, so those may match incorrectly with all interfaces.
Introduce another macro to match with the vid/pid pair and the audio
class interface, and apply it to such entries, so that they can match
properly.
Takashi Iwai [Mon, 17 Aug 2020 08:21:38 +0000 (10:21 +0200)]
ALSA: usb-audio: Move device rename and profile quirks to an internal table
So far we've added the devices that need vendor/product string renames
or the profile setup into the standard quirk table in quirks-table.h.
This table is imported into the primary USB audio device entry, hence
it's all exported for the probing so that udev and co can take a look
at it. OTOH, for renaming or profile setup, we don't need to expose
those explicit entries because the probe itself follows the standard
way. That said, we're exposing unnecessarily too many entries.
This patch moves such internal quirk entries into the own table, and
reduces the exported device table size. Along with the moving items,
re-arrange the entries in the proper order.
Sameer Pujar [Wed, 19 Aug 2020 15:32:10 +0000 (21:02 +0530)]
ALSA: hda: avoid reset of sdo_limit
By default 'sdo_limit' is initialized with a default value of '8'
as per spec. This is overridden in cases where a different value is
required. However this is getting reset when snd_hdac_bus_init_chip()
is called again, which happens during runtime PM cycle.
Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()'
function which would be called only once.
Mike Pozulp [Tue, 18 Aug 2020 16:54:44 +0000 (09:54 -0700)]
ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops
which have the no headphone sound bug, like my Samsung Notebook. The
Galaxy Book owner confirmed that this patch fixes the bug.
Takashi Iwai [Wed, 19 Aug 2020 06:03:04 +0000 (08:03 +0200)]
Merge tag 'asoc-fix-v5.9-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.9
A bunch of fixes that came in during the merge window, mostly for issues
that were uncovered by the changes to report errors on invalid register
access plus one important fix in that code itself.
Tom Yan [Mon, 17 Aug 2020 17:20:11 +0000 (01:20 +0800)]
ALSA: usb-audio: ignore broken processing/extension unit
Some devices have broken extension unit where getting current value
doesn't work. Attempt that once when creating mixer control for it. If
it fails, just ignore it, so that it won't cripple the device entirely
(and/or make the error floods).
Dinghao Liu [Thu, 13 Aug 2020 08:41:10 +0000 (16:41 +0800)]
ASoC: intel: Fix memleak in sst_media_open
When power_up_sst() fails, stream needs to be freed
just like when try_module_get() fails. However, current
code is returning directly and ends up leaking memory.
Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: wm8994: Avoid attempts to read unreadable registers
The driver supports WM1811, WM8994, WM8958 devices but according to
documentation and the regmap definitions the WM8958_DSP2_* registers
are only available on WM8958. In current code these registers are
being accessed as if they were available on all the three chips.
When starting playback on WM1811 CODEC multiple errors like:
"wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5"
can be seen, which is caused by attempts to read an unavailable
WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent
commit "e2329ee ASoC: soc-component: add soc_component_err()".
This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling
is only done for WM8958.
ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
The ADC2 and DAC2 are not available on WM1811 device. This patch moves
the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing
them and avoid unreadable register access on WM1811.
This allows to get rid of warnings during boot like:
wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5
Mike Pozulp [Mon, 17 Aug 2020 04:32:17 +0000 (21:32 -0700)]
ALSA: hda/realtek: Add model alc298-samsung-headphone
The very quiet and distorted headphone output bug that afflicted my
Samsung Notebook 9 is appearing in many other Samsung laptops. Expose
the quirk which fixed my laptop as a model so other users can try it.
Alexander Tsoy [Sat, 15 Aug 2020 00:21:03 +0000 (03:21 +0300)]
ALSA: usb-audio: Add capture support for Saffire 6 (USB 1.1)
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.
Fixes: 11e424e88bd4 ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB") Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Cc: <stable.vger.kernel.org> Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mike Pozulp [Fri, 14 Aug 2020 04:53:44 +0000 (21:53 -0700)]
ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
The Flex Book uses the same ALC298 codec as other Samsung laptops which
have the no headphone sound bug, like my Samsung Notebook. The Flex Book
owner used Early Patching to confirm that this quirk fixes the bug.
Takashi Iwai [Wed, 12 Aug 2020 07:02:56 +0000 (09:02 +0200)]
ALSA: hda/realtek - Fix unused variable warning
The previous fix forgot to remove the unused variable that triggers a
compile warning now:
sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable]
Fix it.
Fixes: 404690649e6a ("ALSA: hda - reverse the setting value in the micmute_led_set") Reported-by: Stephen Rothwell <sfr@canb.auug.org.au> Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: q6routing: add dummy register read/write function
Most of the DAPM widgets for DSP ASoC components reuse reg field
of the widgets for its internal calculations, however these are not
real registers. So read/writes to these numbers are not really
valid. However ASoC core will read these registers to get default
state during startup.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
To fix this add dummy read/write function to return default value.
Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20200811120205.21805-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM
Looks like the q6afe-dai dapm widget registers are set as "0",
which is a not correct.
As this registers will be read by ASoC core during startup
which will throw up errors, Fix this by making the registers
as SND_SOC_NOPM as these should be never used.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
Takashi Iwai [Mon, 10 Aug 2020 13:46:31 +0000 (15:46 +0200)]
ASoC: Make soc_component_read() returning an error code again
Along with the recent unification of snd_soc_component_read*()
functions, the behavior of snd_soc_component_read() was changed
slightly; namely it returns the register read value directly, and even
if an error happens, it returns zero (but it prints an error
message). That said, the caller side can't know whether it's an error
or not any longer.
Ideally this shouldn't matter much, but in practice this seems causing
a regression, as John reported. And, grepping the tree revealed that
there are still plenty of callers that do check the error code, so
we'll need to deal with them in anyway.
As a quick band-aid over the regression, this patch changes the return
value of snd_soc_component_read() again to the negative error code.
It can't work, obviously, for 32bit register values, but it should be
enough for the known regressions, so far.
Fixes: cf6e26c71bfd ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()") Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200810134631.19742-1-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
Hui Wang [Tue, 11 Aug 2020 12:24:30 +0000 (20:24 +0800)]
ALSA: hda - reverse the setting value in the micmute_led_set
Before the micmute_led_set() is introduced, the function of
alc_gpio_micmute_update() will set the gpio value with the
!micmute_led.led_value, and the machines have the correct micmute led
status. After the micmute_led_set() is introduced, it sets the gpio
value with !!micmute_led.led_value, so the led status is not correct
anymore, we need to set micmute_led_polarity = 1 to workaround it.
Now we fix the micmute_led_set() and remove micmute_led_polarity = 1.
Fixes: 87dc36482cab ("ALSA: hda/realtek - Add LED class support for micmute LED") Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 3 Aug 2020 14:39:58 +0000 (16:39 +0200)]
ALSA: echoaduio: Drop superfluous volatile modifier
The dsp_registers field of struct echoaduio has the volatile modifier,
but it's basically superfluous; the field is accessed only for the
base pointer of readl() and writel(), hence marking with __iomem alone
should suffice. OTOH, having the volatile prefix causes a compile
warning like:
sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers]
ASoC: amd: Replacing component->name with codec_dai->name.
Replacing string compare with "codec_dai->name" instead of comparing with
"codec_dai->component->name" in hw_params because,
Here the component name for codec RT1015 is "i2c-10EC5682:00"
and will never be "rt1015-aif1" as it is codec-dai->name.
So, strcmp() always compares and fails to set the
sysclk,pll,bratio for expected codec-dai="rt1015-aif1".
Kai-Heng Feng [Mon, 10 Aug 2020 13:31:06 +0000 (21:31 +0800)]
ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.
Disable the volume control to workaround the issue.
Hector Martin [Mon, 10 Aug 2020 08:24:00 +0000 (17:24 +0900)]
ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Hui Wang [Mon, 10 Aug 2020 02:16:59 +0000 (10:16 +0800)]
ALSA: hda - fix the micmute led status for Lenovo ThinkCentre AIO
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Kai-Heng Feng [Fri, 7 Aug 2020 08:05:12 +0000 (16:05 +0800)]
ALSA: hda/realtek: Fix pin default on Intel NUC 8 Rugged
The jack on Intel NUC 8 Rugged rear panel doesn't work.
The spec [1] states that the jack supports both headphone and
microphone, so override a Pin Complex which has both Amp-In and Amp-Out
to make the jack work.
Node 0x1b fits the requirement, and user confirmed the jack now works
with new pin config.
Mirko Dietrich [Thu, 6 Aug 2020 12:48:50 +0000 (14:48 +0200)]
ALSA: usb-audio: Creative USB X-Fi Pro SB1095 volume knob support
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Mohan Kumar [Wed, 5 Aug 2020 09:52:21 +0000 (15:22 +0530)]
ALSA: hda/tegra: Add 100us dma stop delay
Tegra HDA has audio data buffer for upto tens of frames, this buffer
can help to avoid underflow. HW will keep issuing new data fetch
request when buffers are not full and current BDL is not done. When SW
disable DMA RUN bit for a stream, HW can't cancel the already issued data
fetch request and hence it can't stop DMA. HW has to wait for all issued
data fetch request get data returned before it stops DMA.
This HW behavior is not in sync with HDA spec which says DMA RUN bit
should be cleared within 1 audio frame. For Tegra, DMA RUN bit was
active for more than one audio frame, due to this the timeout in
snd_hdac_stream_sync function is not helping. When Stream reset set
and clear happens during DMA RUN bit active state it results in Memory
Decode error.
Unfortunately, there is no way to detect when these data accesses have
completed, but testing has shown that a 100us delay between Stream reset
set and clear operation for Tegra avoids the memory decode error.
Therefore, adding a 100us dma stop delay.
Mohan Kumar [Wed, 5 Aug 2020 09:52:20 +0000 (15:22 +0530)]
ALSA: hda: Add dma stop delay variable
A variable dma_stop_delay is added as a new member in hdac_bus
structure to avoid memory decode error incase DMA RUN bit is not
disabled in the given timeout from snd_hdac_stream_sync function and
followed by stream reset which results in memory decode error between
reset set and clear operation.
Mohan Kumar [Wed, 5 Aug 2020 09:52:19 +0000 (15:22 +0530)]
ASoC: hda/tegra: Set buffer alignment to 128 bytes
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer
alignment to be multiple of 128 bytes. This fix is applied as gstreamer
alsasink gets stuck with the default buffer-time and latency-time
parameters with 4 byte buffer alignment.
Takashi Iwai [Tue, 4 Aug 2020 18:58:15 +0000 (20:58 +0200)]
ALSA: seq: oss: Serialize ioctls
Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases. This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.
Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency. There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.
Kai-Heng Feng [Tue, 4 Aug 2020 15:58:34 +0000 (23:58 +0800)]
ALSA: hda/hdmi: Add quirk to force connectivity
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Takashi Iwai [Mon, 3 Aug 2020 14:46:30 +0000 (16:46 +0200)]
ASoC: fsl: Fix unused variable warning
The variable rtd was left unused in psc_dma_free(), even unnoticed
during conversion to a new style:
sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable]
Takashi Iwai [Mon, 3 Aug 2020 14:18:50 +0000 (16:18 +0200)]
ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_i2s.c:167:12: warning: 'tegra210_i2s_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_i2s.c:179:12: warning: 'tegra210_i2s_runtime_resume' defined but not used [-Wunused-function]
Takashi Iwai [Mon, 3 Aug 2020 14:18:49 +0000 (16:18 +0200)]
ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_dmic.c:43:12: warning: 'tegra210_dmic_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_dmic.c:55:12: warning: 'tegra210_dmic_runtime_resume' defined but not used [-Wunused-function]
Takashi Iwai [Mon, 3 Aug 2020 14:18:48 +0000 (16:18 +0200)]
ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:579:12: warning: 'tegra_ahub_runtime_resume' defined but not used [-Wunused-function]
Takashi Iwai [Mon, 3 Aug 2020 14:18:47 +0000 (16:18 +0200)]
ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_admaif.c:232:12: warning: 'tegra_admaif_runtime_resume' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
Takashi Iwai [Mon, 3 Aug 2020 14:18:46 +0000 (16:18 +0200)]
ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra186_dspk.c:74:12: warning: 'tegra186_dspk_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra186_dspk.c:86:12: warning: 'tegra186_dspk_runtime_resume' defined but not used [-Wunused-function]
Kai-Heng Feng [Mon, 3 Aug 2020 14:26:08 +0000 (22:26 +0800)]
ALSA: usb-audio: Add support for Lenovo ThinkStation P620
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.
USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0
Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.
USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)
So turn off the FU to avoid the error.
Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Hui Wang [Thu, 30 Jul 2020 12:31:38 +0000 (20:31 +0800)]
ASoC: amd: renoir: restore two more registers during resume
Recently we found an issue about the suspend and resume. If dmic is
recording the sound, and we run suspend and resume, after the resume,
the dmic can't work well anymore. we need to close the app and reopen
the app, then the dmic could record the sound again.
For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2
-r 48000 test.wav", then suspend and resume, after the system resume
back, we speak to the dmic. then stop the arecord, use aplay to play
the test.wav, we could hear the sound recorded after resume is weird,
it is not what we speak to the dmic.
I found two registers are set in the dai_hw_params(), if the two
registers are set during the resume, this issue could be fixed.
Move the code of the dai_hw_params() into the pdm_dai_trigger(), then
these two registers will be set during resume since pdm_dai_trigger()
will be called during resume. And delete the empty function
dai_hw_params().
Fabio Estevam [Mon, 3 Aug 2020 11:52:33 +0000 (08:52 -0300)]
ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list
Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause
audio distortion.
This change was sent as a comment below the --- line when submitting
commit 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so
it was not supposed to get merged.
Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to
fix the regression.
Fixes: 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE") Reported-by: Shengjiu Wang <shengjiu.wang@gmail.com> Signed-off-by: Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20200803115233.19034-1-festevam@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Shengjiu Wang [Mon, 3 Aug 2020 02:13:31 +0000 (10:13 +0800)]
ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
With this case:
aplay -Dhw:x 16khz.wav 24khz.wav
There is sound distortion for 24khz.wav. The reason is that setting
PLL of WM8962 with set_bias_level function, the bias level is not
changed when 24khz.wav is played, then the PLL won't be reset, the
clock is not correct, so distortion happens.
The resolution of this issue is to remove fsl_asoc_card_set_bias_level.
Move PLL configuration to hw_params and hw_free.
After removing fsl_asoc_card_set_bias_level, also test WM8960 case,
it can work.
Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/1596420811-16690-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Mon, 3 Aug 2020 12:41:43 +0000 (14:41 +0200)]
Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.9
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
Hui Wang [Mon, 3 Aug 2020 06:46:38 +0000 (14:46 +0800)]
Revert "ALSA: hda: call runtime_allow() for all hda controllers"
This reverts commit 9a6418487b56 ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Connor McAdams [Mon, 3 Aug 2020 00:29:25 +0000 (20:29 -0400)]
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Huacai Chen [Sun, 2 Aug 2020 09:26:40 +0000 (17:26 +0800)]
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Mark Brown [Fri, 31 Jul 2020 18:36:00 +0000 (19:36 +0100)]
Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski@intel.com>:
Provide a mechanism for true two-step component registration. This
mimics device registration flow where initialization is the first step
while addition goes as second in line. Drivers may choose to modify
component's fields before registering component to ASoC subsystem via
snd_soc_add_component.
Patchset achieves status quo - behavior of snd_soc_register_component
remains unchanged.
Cezary Rojewski (3):
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Two step component registration
Modify snd_soc_add_component so it calls snd_soc_component_initialize
no longer and thus providing true two-step registration. Drivers may
choose to change component's fields before actually adding it to ASoC
subsystem.
Move 'name' field initialization responsibility back to
snd_soc_component_initialize to prepare snd_soc_add_component function
for being called separatelly as a second registration step.
Enabling a whole subsystem from a single driver 'select' is frowned
upon and won't be accepted in new drivers, that need to use 'depends on'
instead. Existing selection of DMADEVICES will then cause circular
dependencies. Replace them with a dependency.
Commit b73287f0b074 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
changed the meaning of dpcm_playback/dpcm_capture and now requires the
CPU DAI BE to aligned with those flags.
This broke all Amlogic cards with uni-directional backends (All gx and
most axg cards).
While I'm still confused as to how this change is an improvement, those
cards can't remain broken forever. Hopefully, next time an API change is
done like that, all the users will be updated as part of the change, and
not left to fend for themselves.
projects / mirror_ubuntu-jammy-kernel.git / log