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The latest fix for the non-contiguous memalloc helper changed the
allocation method for a non-IOMMU system to use only the fallback
allocator. This should have worked, but it caused a problem sometimes
when too many non-contiguous pages are allocated that can't be treated
by HD-audio controller.
As a quirk workaround, go back to the original strategy: use
dma_alloc_noncontiguous() at first, and apply the fallback only when
it fails, but only for non-IOMMU case.
We'll need a better fix in the fallback code as well, but this
workaround should paper over most cases.
Fixes: 9736a32513 ("ALSA: memalloc: Don't fall back for SG-buffer with IOMMU")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Link: https://lore.kernel.org/r/CAHk-=wgSH5ubdvt76gNwa004ooZAEJL_1Q-Fyw5M2FDdqL==dg@mail.gmail.com
Link: https://lore.kernel.org/r/20221112084718.3305-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the non-contiguous page allocation for SG buffer allocation
fails, the memalloc helper tries to fall back to the old page
allocation methods. This would, however, result in the bogus page
addresses when IOMMU is enabled. Usually in such a case, the fallback
allocation should fail as well, but occasionally it succeeds and
hitting a bad access.
The fallback was thought for non-IOMMU case, and as the error from
dma_alloc_noncontiguous() with IOMMU essentially implies a fatal
memory allocation error, we should return the error straightforwardly
without fallback. This avoids the corner case like the above.
The patch also renames the local variable "dma_ops" with snd_ prefix
for avoiding the name conflict.
Fixes: a8d302a0b7 ("ALSA: memalloc: Revive x86-specific WC page allocations again")
Reported-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.22.394.2211041541090.3532114@eliteleevi.tm.intel.com
Link: https://lore.kernel.org/r/20221110132216.30605-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although we tried to fix the regression for the recent changes with
the delayed card registration, it doesn't seem covering the all
cases; e.g. on Roland EDIROL M-100FX, where the generic quirk for
Roland devices is applied, it misses the card registration because the
detection of the last interface (apparently for MIDI) fails.
This patch is an attempt to recover from those failures by calling the
card register also at the error path for the secondary interfaces.
The card register condition is also extended to match with the old
check in the previous patch, too (i.e. the simple check of the
interface number) for catching the probe with errors.
Fixes: 39efc9c8a9 ("ALSA: usb-audio: Fix last interface check for registration")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.suse.com/show_bug.cgi?id=1205111
Link: https://lore.kernel.org/r/20221108065824.14418-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 1f9d3d9869 ("ALSA: hda - set intel audio clock to a
proper value") added a number of misleading comments.
There is no ability to detect if an SCF value was set or not, what the
code does is prevent the use of the 6MHz audio clock represented by
the value 0 in LCTL.SCF. Changing the SCF settings does require the
link to be power-cycled, but in all other cases the link is powered
automatically when exiting reset. In other words, the power-cycle is
an exception to the rule that the HDaudio legacy driver does not need
to program SPA/CPA bits.
In addition, the SCF related changes are only relevant for the first
link.
No functionality change, only comment clarifications.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20221031195505.249929-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
