i2sbus_add_dev() is supposed to return the number of probed devices,
i.e. either 1 or 0. However, i2sbus_add_dev() has one error handling
that returns -ENODEV; this will screw up the accumulation number
counted in the caller, i2sbus_probe().
Fix the return value to 0 and add the comment for better understanding
for readers.
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Link: https://lore.kernel.org/r/20221027065233.13292-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code for freeing the emux timer is extremely dangerous:
CPU0 CPU1
---- ----
snd_emux_timer_callback()
snd_emux_free()
spin_lock(&emu->voice_lock)
del_timer(&emu->tlist); <-- returns immediately
spin_unlock(&emu->voice_lock);
[..]
kfree(emu);
spin_lock(&emu->voice_lock);
[BOOM!]
Instead just use del_timer_sync() which will wait for the timer to finish
before continuing. No need to check if the timer is active or not when
doing so.
This doesn't fix the race of a possible re-arming of the timer, but at
least it won't use the data that has just been freed.
[ Fixed unused variable warning by tiwai ]
Cc: stable@vger.kernel.org
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Signed-off-by: Steven Rostedt (Google) <rostedt@goodmis.org>
Reviewed-by: Guenter Roeck <linux@roeck-us.net>
Link: https://lore.kernel.org/r/20221026231236.6834b551@gandalf.local.home
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev_set_name() in soundbus_add_one() allocates memory for name, it need be
freed when of_device_register() fails, call soundbus_dev_put() to give up
the reference that hold in device_initialize(), so that it can be freed in
kobject_cleanup() when the refcount hit to 0. And other resources are also
freed in i2sbus_release_dev(), so it can return 0 directly.
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Signed-off-by: Yang Yingliang <yangyingliang@huawei.com>
Link: https://lore.kernel.org/r/20221027013438.991920-1-yangyingliang@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v6.1
Quite a few fixes here, a lot driver specific, plus some new quirks.
There was a bit of a mess with the runtime PM handling due to some
confusion in the API there which resulted in a number of commits and
reverts but that should all be stable now.
With char becoming unsigned by default, and with `char` alone being
ambiguous and based on architecture, signed chars need to be marked
explicitly as such. This fixes warnings like:
sound/pci/rme9652/hdsp.c:3953 hdsp_channel_buffer_location() warn: 'hdsp->channel_map[channel]' is unsigned
sound/pci/rme9652/hdsp.c:4153 snd_hdsp_channel_info() warn: impossible condition '(hdsp->channel_map[channel] < 0) => (0-255 < 0)'
sound/pci/rme9652/rme9652.c:1833 rme9652_channel_buffer_location() warn: 'rme9652->channel_map[channel]' is unsigned
Signed-off-by: Jason A. Donenfeld <Jason@zx2c4.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221025000313.546261-1-Jason@zx2c4.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With char becoming unsigned by default, and with `char` alone being
ambiguous and based on architecture, signed chars need to be marked
explicitly as such. This fixes warnings like:
sound/pci/au88x0/au88x0_core.c:2029 vortex_adb_checkinout() warn: signedness bug returning '(-22)'
sound/pci/au88x0/au88x0_core.c:2046 vortex_adb_checkinout() warn: signedness bug returning '(-12)'
sound/pci/au88x0/au88x0_core.c:2125 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, (0), en, 0)' is unsigned
sound/pci/au88x0/au88x0_core.c:2170 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, stream->resources, en, 4)' is unsigned
As well, since one function returns errnos, return an `int` rather than
a `signed char`.
Signed-off-by: Jason A. Donenfeld <Jason@zx2c4.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221024162929.536004-1-Jason@zx2c4.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Siarhei Volkau <lis8215@gmail.com>:
The patchset fixes:
- Line In path stays powered off during capturing or
bypass to mixer.
- incorrectly represented dB values in alsamixer, et al.
- incorrect represented Capture input selector in alsamixer
in Playback tab.
- wrong control selected as Capture Master
The "convert-xxx" properties only have an effect for DPCM DAI links.
A DAI link is only created as DPCM if the device tree requires it;
part of this involves checking for the use of "convert-xxx" properties.
When the convert-sample-format property was added, the checks got out
of sync. A DAI link that specified only convert-sample-format but did
not pass any of the other DPCM checks would not go into DPCM mode and
the convert-sample-format property would be silently ignored.
Fix this by adding a function to do the "convert-xxx" property checks,
instead of open-coding it in simple-card and audio-graph-card. And add
"convert-sample-format" to the check function so that DAI links using
it will be initialized correctly.
Fixes: 047a05366f ("ASoC: simple-card-utils: Fixup DAI sample format")
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20221019012302.633830-1-aidanmacdonald.0x0@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If SOUNDWIRE is enabled, then SND_SOC_SC7180 should depend on
SOUNDWIRE to prevent SOUNDWIRE=m and SND_SOC_SC7180=y, which causes
build errors:
s390-linux-ld: sound/soc/qcom/common.o: in function `qcom_snd_sdw_prepare':
common.c:(.text+0x140): undefined reference to `sdw_disable_stream'
s390-linux-ld: common.c:(.text+0x14a): undefined reference to `sdw_deprepare_stream'
s390-linux-ld: common.c:(.text+0x158): undefined reference to `sdw_prepare_stream'
s390-linux-ld: common.c:(.text+0x16a): undefined reference to `sdw_enable_stream'
s390-linux-ld: common.c:(.text+0x17c): undefined reference to `sdw_deprepare_stream'
s390-linux-ld: sound/soc/qcom/common.o: in function `qcom_snd_sdw_hw_free':
common.c:(.text+0x344): undefined reference to `sdw_disable_stream'
s390-linux-ld: common.c:(.text+0x34e): undefined reference to `sdw_deprepare_stream'
Fixes: 3bd975f3ae ("ASoC: qcom: sm8250: move some code to common")
Fixes: 9e3ecb5b16 ("ASoC: qcom: sc7180: Add machine driver for sound card registration")
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Reported-by: kernel test robot <lkp@intel.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Banajit Goswami <bgoswami@quicinc.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Ajit Pandey <ajitp@codeaurora.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: stable@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20221015001228.18990-1-rdunlap@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Line In Bypass control is used as Master Capture at the moment
this is completely incorrect.
Current control routed to Mixer instead of ADC, thus can't affect
Capture path. ADC control shall be used instead.
ADC volume control parameters are different, so the patch fixes that
as well. Manual says (16.6.3.2 Programmable input attenuation amplifier:
PGATM) that gain varies in range 0dB..22.5dB with 1.5dB step.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-4-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAC volume control is the Master Playback Volume at the moment
and it reports wrong levels in alsamixer and other alsa apps.
The patch fixes that, as stated in manual on the jz4725b SoC
(16.6.3.4 Programmable attenuation: GOD) the ctl range varies
from -22.5dB to 0dB with 1.5dB step.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-3-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound fixes from Takashi Iwai:
"Here are a few remaining patches for 6.1-rc1.
The major changes are the hibernation fixes for HD-audio CS35L41 codec
and the USB-audio small fixes against the last change. In addition, a
couple of HD-audio regression fixes and a couple of potential
mutex-deadlock fixes with OSS emulation in ALSA core side are seen"
* tag 'sound-fix-6.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: cs35l41: Support System Suspend
ALSA: hda: cs35l41: Remove suspend/resume hda hooks
ALSA: hda/cs_dsp_ctl: Fix mutex inversion when creating controls
ALSA: hda: hda_cs_dsp_ctl: Ensure pwr_lock is held before reading/writing controls
ALSA: hda: hda_cs_dsp_ctl: Minor clean and redundant code removal
ALSA: oss: Fix potential deadlock at unregistration
ALSA: rawmidi: Drop register_mutex in snd_rawmidi_free()
ALSA: hda/realtek: Add Intel Reference SSID to support headset keys
ALSA: hda/realtek: Add quirk for ASUS GV601R laptop
ALSA: hda/realtek: Correct pin configs for ASUS G533Z
ALSA: usb-audio: Avoid superfluous endpoint setup
ALSA: usb-audio: Correct the return code from snd_usb_endpoint_set_params()
ALSA: usb-audio: Apply mutex around snd_usb_endpoint_set_params()
ALSA: usb-audio: Avoid unnecessary interface change at EP close
ALSA: hda: Update register polling macros
ALSA: hda/realtek: remove ALC289_FIXUP_DUAL_SPK for Dell 5530
Update HDMI volatile registers list as DMA, Channel Selection registers,
vbit control registers are being reflected by hardware DP port
disconnection.
This update is required to fix no display and no sound issue observed
after reconnecting TAMA/SANWA DP cables.
Once DP cable is unplugged, DMA control registers are being reset by
hardware, however at second plugin, new dma control values does not
updated to the dma hardware registers since new register value and
cached values at the time of first plugin are same.
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam@quicinc.com>
Reported-by: Kuogee Hsieh <quic_khsieh@quicinc.com>
Link: https://lore.kernel.org/r/1665637711-13300-1-git-send-email-quic_srivasam@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In wm8962 driver, the WM8962_ADDITIONAL_CONTROL_4 is used as a volatile
register, but this register mixes a bunch of volatile status bits and a
bunch of non-volatile control bits. The dapm widgets TEMP_HP and
TEMP_SPK leverages the control bits in this register. After the wm8962
probe, the regmap will bet set to cache only mode, then a read error
like below would be triggered when trying to read the initial power
state of the dapm widgets TEMP_HP and TEMP_SPK.
wm8962 0-001a: ASoC: error at soc_component_read_no_lock
on wm8962.0-001a: -16
In order to fix this issue, we add event handler to actually power
up/down these widgets. With this change, we also need to explicitly
power off these widgets in the wm8962 probe since they are enabled
by default.
Signed-off-by: Xiaolei Wang <xiaolei.wang@windriver.com>
Tested-by: Adam Ford <aford173@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221010092014.2229246-1-xiaolei.wang@windriver.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for system suspend into the CS35L41 HDA Driver.
Since S4 suspend may power off the system, it is required
that the driver ensure the part is safe to be shutdown before
system suspend, as well as ensuring that the firmware is
unloaded before shutdown. The part must then be restored
on system resume, including re-downloading the firmware.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221011143552.621792-6-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code uses calls from the HDA Codec driver to
determine when to suspend/resume by calling hooks via the
hda_component binding.
However, this means the cs35l41 driver relies on the HDA
Codec driver to tell it when to suspend or resume,
creating an additional external dependency, and potentially
creating race conditions in the future. It is better for
the cs35l41 hda driver to decide for itself when the part
should be suspended or resumed.
This makes supporting system suspend easier.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221011143552.621792-5-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Redesign the creation of ALSA controls so that the cs_dsp
pwr_lock is not held when calling snd_ctl_add(). Instead of
creating the ALSA control from the cs_dsp control_add callback,
do it after cs_dsp_power_up() has completed. The existing
functions are changed to return void instead of passing errors
back - this duplicates the original behaviour, as cs_dsp does
not abort firmware load if creation of a control fails.
It is safe to walk the control list without taking any mutex
provided that the caller is not trying to load a new firmware
or remove the driver in parallel. There is no other situation
that the list can change. So the caller can trigger creation
of ALSA controls after cs_dsp_power_up() has returned. A cs_dsp
control will have a non-NULL priv pointer if we have created
an ALSA control.
With the previous code the ALSA controls were created from
the cs_dsp control_add callback. But this is called with
pwr_lock held (as it is part of the DSP power-up sequence).
The kernel lock checking will show a mutex inversion between
this and the control creation path:
control_add
pwr_lock held, takes controls_rwsem (in snd_ctl_add)
get/put
controls_rwsem held, takes pwr_lock to call cs_dsp.
This is not completely theoretical. Although the time window
is very small, it is possible for these to run in parallel
and deadlock the old implementation.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221011143552.621792-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs_dsp core will return an error if passed a NULL cs_dsp struct so
there is no need for the hda_cs_dsp_write|read_ctl functions to manually
check that. The cs_dsp core will also check the data is within bounds of
the control so the additional bounds check is redundant too. Simplify
things a bit by removing said code.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221011143552.621792-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We took sound_oss_mutex around the calls of unregister_sound_special()
at unregistering OSS devices. This may, however, lead to a deadlock,
because we manage the card release via the card's device object, and
the release may happen at unregister_sound_special() call -- which
will take sound_oss_mutex again in turn.
Although the deadlock might be fixed by relaxing the rawmidi mutex in
the previous commit, it's safer to move unregister_sound_special()
calls themselves out of the sound_oss_mutex, too. The call is
race-safe as the function has a spinlock protection by itself.
Link: https://lore.kernel.org/r/CAB7eexJP7w1B0mVgDF0dQ+gWor7UdkiwPczmL7pn91xx8xpzOA@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221011070147.7611-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
