The previous fix for addressing the breakage in vmaster slave
initialization, commit a91d66129fb9 ("ALSA: hda - Fix incorrect TLV
callback check introduced during set_fs() removal"), introduced a new
helper to process over each slave kctl. However, this helper passes
only the original kctl, not the virtual slave kctl. As a result,
HD-audio driver (which is the only user so far) couldn't initialize
the slave correctly because it's trying to update the value directly
with the original kctl, not with the mapped kctl.
This patch fixes the situation again by passing both the mapped slaved
and original slave kctls to the function. Luckily there is a single
caller as of now, so changing the call signature is no big matter.
Takashi Iwai [Tue, 21 Nov 2017 16:28:06 +0000 (17:28 +0100)]
ALSA: usb-audio: Add sanity checks in v2 clock parsers
The helper functions to parse and look for the clock source, selector
and multiplier unit may return the descriptor with a too short length
than required, while there is no sanity check in the caller side.
Add some sanity checks in the parsers, at least, to guarantee the
given descriptor size, for avoiding the potential crashes.
Takashi Iwai [Tue, 21 Nov 2017 16:07:43 +0000 (17:07 +0100)]
ALSA: usb-audio: Fix potential zero-division at parsing FU
parse_audio_feature_unit() contains a code dividing potentially with
zero when a malformed FU descriptor is passed. Although there is
already a sanity check, it checks only the value zero, hence it can
still lead to a zero-division when a value 1 is passed there.
Fix it by correcting the sanity check (and the error message
thereof).
Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 21 Nov 2017 16:00:32 +0000 (17:00 +0100)]
ALSA: usb-audio: Fix potential out-of-bound access at parsing SU
The usb-audio driver may trigger an out-of-bound access at parsing a
malformed selector unit, as it checks the header length only after
evaluating bNrInPins field, which can be already above the given
length. Fix it by adding the length check beforehand.
Takashi Iwai [Tue, 21 Nov 2017 15:55:51 +0000 (16:55 +0100)]
ALSA: usb-audio: Add sanity checks to FE parser
When the usb-audio descriptor contains the malformed feature unit
description with a too short length, the driver may access
out-of-bounds. Add a sanity check of the header size at the beginning
of parse_audio_feature_unit().
Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0") Reported-by: Andrey Konovalov <andreyknvl@google.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 21 Nov 2017 15:36:11 +0000 (16:36 +0100)]
ALSA: timer: Remove kernel warning at compat ioctl error paths
Some timer compat ioctls have NULL checks of timer instance with
snd_BUG_ON() that bring up WARN_ON() when the debug option is set.
Actually the condition can be met in the normal situation and it's
confusing and bad to spew kernel warnings with stack trace there.
Let's remove snd_BUG_ON() invocation and replace with the simple
checks. Also, correct the error code to EBADFD to follow the native
ioctl error handling.
Henrik Eriksson [Tue, 21 Nov 2017 08:29:28 +0000 (09:29 +0100)]
ALSA: pcm: update tstamp only if audio_tstamp changed
commit 3179f6200188 ("ALSA: core: add .get_time_info") had a side effect
of changing the behaviour of the PCM runtime tstamp. Prior to this
change tstamp was not updated by snd_pcm_update_hw_ptr0() unless the
hw_ptr had moved, after this change tstamp was always updated.
For an application using alsa-lib, doing snd_pcm_readi() followed by
snd_pcm_status() to estimate the age of the read samples by subtracting
status->avail * [sample rate] from status->tstamp this change degraded
the accuracy of the estimate on devices where the pcm hw does not
provide a granular hw_ptr, e.g., devices using
soc-generic-dmaengine-pcm.c and a dma-engine with residue_granularity
DMA_RESIDUE_GRANULARITY_DESCRIPTOR. The accuracy of the estimate
depended on the latency between the PCM hw completing a period and the
driver called snd_pcm_period_elapsed() to notify ALSA core, typically
determined by interrupt handling latency. After the change the accuracy
of the estimate depended on the latency between the PCM hw completing a
period and the application calling snd_pcm_status(), determined by the
scheduling of the application process. The maximum error of the
estimate is one period length in both cases, but the error average and
variance is smaller when it depends on interrupt latency.
Instead of always updating tstamp, update it only if audio_tstamp
changed.
Takashi Iwai [Fri, 17 Nov 2017 11:08:40 +0000 (12:08 +0100)]
ALSA: hda: Fix too short HDMI/DP chmap reporting
We got a regression report about the HD-audio HDMI chmap, where some
surround channels are reported as UNKNOWN. The git bisection pointed
the culprit at the commit 9b3dc8aa3fb1 ("ALSA: hda - Register chmap
obj as priv data instead of codec"). The story behind scene is like
this:
- While moving the code out of the legacy HDA to the HDA common place,
the patch modifies the code to obtain the chmap array indirectly in
a byte array, and it expands it to kctl value array.
- At the latter operation, the size of the array is wrongly passed by
sizeof() to the pointer.
- It can be 4 on 32bit arch, thus too short for 6+ channels.
(And that's the reason why it didn't hit other persons; it's 8 on
64bit arch, thus it's usually enough.)
The code was further changed meanwhile, but the problem persisted.
Let's fix it by correctly evaluating the array size.
Fixes: 9b3dc8aa3fb1 ("ALSA: hda - Register chmap obj as priv data instead of codec") Reported-by: VDR User <user.vdr@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Julian Scheel [Thu, 16 Nov 2017 16:35:17 +0000 (17:35 +0100)]
ALSA: usb-audio: uac1: Invalidate ctl on interrupt
When an interrupt occurs, the value of at least one of the belonging
controls should have changed. To make sure they get re-read from device
on the next read, invalidate the cache. This was correctly implemented
for uac2 already, but missing for uac1.
Takashi Iwai [Mon, 13 Nov 2017 14:45:57 +0000 (15:45 +0100)]
Merge tag 'asoc-v4.15' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.15
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
Mark Brown [Fri, 10 Nov 2017 21:30:27 +0000 (21:30 +0000)]
Merge tag 'asoc-fix-v4.14-rc6' into asoc-linus
ASoC: Fixes for v4.14
I've been quite lax in sending these due to conference season but here's
a fairly large collection of ASoC updates. The one thing that's not
device specific is Takashi's fix for races between delayed work and PCM
destruction, otherwise everything is specific to an individual device.
# gpg: Signature made Thu 26 Oct 2017 15:11:23 BST
# gpg: using RSA key ADE668AA675718B59FE29FEA24D68B725D5487D0
# gpg: issuer "broonie@kernel.org"
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" [unknown]
# gpg: aka "Mark Brown <broonie@debian.org>" [unknown]
# gpg: aka "Mark Brown <broonie@kernel.org>" [unknown]
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" [unknown]
# gpg: aka "Mark Brown <broonie@linaro.org>" [unknown]
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>" [unknown]
# gpg: WARNING: This key is not certified with a trusted signature!
# gpg: There is no indication that the signature belongs to the owner.
# Primary key fingerprint: 3F25 68AA C269 98F9 E813 A1C5 C3F4 36CA 30F5 D8EB
# Subkey fingerprint: ADE6 68AA 6757 18B5 9FE2 9FEA 24D6 8B72 5D54 87D0
Matthias Reichl [Wed, 8 Nov 2017 20:03:30 +0000 (21:03 +0100)]
ASoC: bcm2835: Support left/right justified and DSP modes
DSP modes and left/right justified modes can be supported
on bcm2835 by configuring the frame sync polarity and
frame sync length registers and by adjusting the
channel data position registers.
Clock and frame sync polarity handling in hw_params has
been refactored to make the interaction between logical
rising/falling edge frame start and physical configuration
(changed by normal/inverted polarity modes) clearer.
Modes where the first active data bit is transmitted immediately
after frame start (eg DSP mode B with slot 0 active)
only work reliable if bcm2835 is configured as frame master.
In frame slave mode channel swap (or shift, this isn't quite
clear yet) can occur.
Currently the driver only warns if an unstable configuration
is detected but doensn't prevent using them.
Signed-off-by: Matthias Reichl <hias@horus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Matthias Reichl [Wed, 8 Nov 2017 20:03:32 +0000 (21:03 +0100)]
ASoC: bcm2835: Enforce full symmetry
bcm2835's configuration registers can't be changed when a stream
is running, which means asymmetric configurations aren't supported.
Channel and rate symmetry are already enforced by constraints
but samplebits had been missed.
As hw_params doesn't check for symmetry constraints by itself
and just returns success if a stream is running this led to
situations where asymmetric configurations were seeming to
succeed but of course didn't work because the hardware wasn't
configured at all.
Fix this by adding the missing samplerate symmetry constraint.
Signed-off-by: Matthias Reichl <hias@horus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Matthias Reichl [Wed, 8 Nov 2017 20:03:29 +0000 (21:03 +0100)]
ASoC: bcm2835: Add support for TDM modes
bcm2835 supports arbitrary positioning of channel data within
a frame and thus is capable of supporting TDM modes. Since
the driver is limited to 2-channel operations only TDM setups
with exactly 2 active slots are supported.
Logical TDM slot numbering follows the usual convention:
For I2S-like modes, with a 50% duty-cycle frame clock,
slots 0, 2, ... are transmitted in the first half of a frame,
slots 1, 3, ... are transmitted in the second half.
For DSP modes slot numbering is ascending: 0, 1, 2, 3, ...
Channel position calculation has been refactored to use
TDM info and moved out of hw_params.
set_tdm_slot, set_bclk_ratio and hw_params now check more
strictly if the configuration is valid. Illegal configurations
like odd number of slots in I2S mode, data lengths exceeding
slot width or frame sizes larger than the hardware limit of
1024 are rejected. Also hw_params now properly checks for
errors from clk_set_rate.
Allowed PCM formats are already guarded by stream constraints,
thus the formats check in hw_params has been removed and
data_length is now retrieved via params_width().
Also standard functions like snd_soc_params_to_bclk are now
being used instead of manual calculations to make the code
more readable.
Special care has been taken to ensure that set_bclk_ratio works
as before. The bclk ratio is mapped to a 2-channel TDM config
with a slot width of half the ratio. In order to support odd ratios,
which can't be expressed via a TDM config, the ratio (frame length)
is stored and used by hw_params.
Signed-off-by: Matthias Reichl <hias@horus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Fri, 10 Nov 2017 14:54:43 +0000 (15:54 +0100)]
ASoC: rt5514: work around link error
The main rt5514 driver optionally calls into the SPI back-end to load
the firmware. This causes a link error when one driver selects rt5514
as built-in and another driver selects rt5514-spi as a loadable module:
sound/soc/codecs/rt5514.o: In function `rt5514_dsp_voice_wake_up_put':
rt5514.c:(.text+0xac8): undefined reference to `rt5514_spi_burst_write'
As a workaround, this adds another silent symbol, to force rt5514-spi
to be built-in for that configuration. I'm not overly happy with
that solution, but couldn't come up with anything better. Using
'IS_REACHABLE()' would break the case that relies on the loadable
module, and all other ideas would result in more complexity.
Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Fri, 10 Nov 2017 14:54:42 +0000 (15:54 +0100)]
ASoC: rt5514: mark PM functions as __maybe_unused
The new functions are only used when CONFIG_PM is enabled,
leading to a harmless warning:
sound/soc/codecs/rt5514-spi.c:474:12: error: 'rt5514_resume' defined but not used [-Werror=unused-function]
sound/soc/codecs/rt5514-spi.c:464:12: error: 'rt5514_suspend' defined but not used [-Werror=unused-function]
This marks them as __maybe_unused to make the build silent
again.
Fixes: 58f1c07d23cd ("ASoC: rt5514: Voice wakeup support.") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: amd: Modified DMA transfer Mechanism for Playback
Before rendering starts, DMA driver copies full buffer valid data
to ACP SRAM for the first time, after that ACP SRAM to I2S
FIFO DMA will be initiated. After rendering first half of ACP SRAM,
IOC will be raised then Audio data will be copied from first half of
System Memory to first half of ACP SRAM. Similarly after rendering
second half of ACP SRAM, IOC will be raised then Audio Data will be
copied from second half of the System Memory to second half of the
ACP SRAM in ping-pong way till rendering stops.
Old design introducing latency issues resulting stutter sound observed
during playback.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Signed-off-by: Akshu Agrawal <Akshu.Agrawal@amd.com> Signed-off-by: Alex Deucher <alexander.deucher@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+
32bit and 24bit audio capture formats for H3/H2+ are broken because the
RX_SAMPLE_BITS and the RX_FIFO_MODE bits of AC_ADC_FIFOC register of the audio
codec are not set to operate in 24bit mode but in 16bit mode only.
The following patch sets the H3 audio codec registers and the DMA bus width
properly when a 24/32bit capture is requested.
Signed-off-by: Andrea Bondavalli <andrea.bondavalli74@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
DSP modes are documented in the data sheet but not enabled in the driver.
The work-around already implemented for DA7218/9 is also required to
make sure the bit clock handling in DSP modes follows ASoC conventions.
Tested with ARD-AUDIO-DA7212 and Minnowmax Turbot boards
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Maxime Ripard [Thu, 9 Nov 2017 09:39:24 +0000 (10:39 +0100)]
ASoC: sun8i-codec: Set the BCLK divider
While the current code was reporting to be able to work in master mode, it
failed to do so because the BCLK divider wasn't programmed, meaning that
the BCLK would run at the PLL's frequency no matter the sample rate.
It was obviously a bit too fast.
Add support to retrieve the divider to use, and set it. Since our PLL is
not always able to generate a perfect multiple of the sample rate, we'll
have to choose the closest divider that matches our setup.
Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Reviewed-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: <stable@vger.kernel.org>
Oder Chiou [Thu, 9 Nov 2017 11:28:10 +0000 (19:28 +0800)]
ASoC: rt5663: Delay and retry reading rt5663 ID register
In the probe, the codec may not be ready for I2C reading or there are some
glitches on the i2c line. So if the i2c reading value is incorrect, it will
read again after delay. This issue is similar the patch
https://patchwork.kernel.org/patch/9681421/. In current project, these 2
devices were connected to the same i2c line, and they met the same problem.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hui Wang [Thu, 9 Nov 2017 00:48:08 +0000 (08:48 +0800)]
ALSA: hda - fix headset mic problem for Dell machines with alc274
Confirmed with Kailang of Realtek, the pin 0x19 is for Headset Mic, and
the pin 0x1a is for Headphone Mic, he suggested to apply
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix this problem. And we
verified applying this FIXUP can fix this problem.
Cc: <stable@vger.kernel.org> Cc: Kailang Yang <kailang@realtek.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
64-bit divides require special operations to avoid build errors on 32-bit
systems.
[Reword the commit message to make it clearer - Alex]
fixes: 61add8147942 (ASoC: amd: Report accurate hw_ptr during dma) Signed-off-by: Guenter Roeck <groeck@chromium.org>
Reviewed-on: https://chromium-review.googlesource.com/678919 Reviewed-by: Jason Clinton <jclinton@chromium.org>
Reviewed-on: https://chromium-review.googlesource.com/681618 Signed-off-by: Alex Deucher <alexander.deucher@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Andrew F. Davis [Wed, 8 Nov 2017 21:24:59 +0000 (15:24 -0600)]
ASoC: cs42l56: Fix reset GPIO name in example DT binding
The binding states the reset GPIO property shall be named
"cirrus,gpio-nreset" and this is what the driver looks for,
but the example uses "gpio-reset". Fix this here.
Fixes: 3bb40619aca8 ("ASoC: cs42l56: bindings: sound: Add bindings for CS42L56 CODEC") Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: rt5514-spi: check irq status to schedule data copy in resume function
For wake on voice use case, we need to copy data from DSP buffer
to PCM stream when system wakes up by voice. However the edge
triggered IRQ could be missed when system wakes up, in that case
the irq function will not be called. If the substream was constructed
beforce suspend, we will schedule data copy in resume function.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
If the rt5514 Wake on Voice device is opened while suspended, it will
be able to wake up the system when a voice command is detected.
This patch also supports user-space policy to override wakeup behavior
by /sys/bus/spi/drivers/rt5514/spi2.0/power/wakeup.
Signed-off-by: Chinyue Chen <chinyue@chromium.org> Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 146568
Addresses-Coverity-ID: 146569 Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Current codec drivers are using snd_soc_read(). It will be replaced
into snd_soc_component_read(), but these 2 are using different style.
For example, it will be
- val = snd_soc_read(xxx, reg);
+ ret = snd_soc_component_read(xxx, reg, &val);
+ if (ret < 0) {
+ ...
+ }
To more smooth replace, let's add snd_soc_component_read32
which is copied from snd_soc_read()
- val = snd_soc_read(xxx, reg);
+ val = snd_soc_component_read32(xxx, reg);
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: Skylake: Optimize UUID handling to fill pin info
Modify skl_tplg_get_uuid() to copy just UUID rather than only
for module UUID and skl_tplg_fill_pin() to fill the pin info
which can include UUID token also.
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com> Acked-By: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: Skylake: Fix uuid_module memory leak in failure case
In the loop that adds the uuid_module to the uuid_list list, allocated
memory is not properly freed in the error path free uuid_list whenever
any of the memory allocation in the loop fails to avoid memory leak.
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com> Acked-By: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Guneshwor Singh [Tue, 7 Nov 2017 10:46:16 +0000 (16:16 +0530)]
ASoC: Intel: Skylake: Fix updown mixer module format
DSP expects length of the coefficient for updown mixer module to be 8.
So fix the max coefficient length and since we are using default values
for coefficient select which is zero, we need not explicitly initialize
it.
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com> Acked-By: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Maxime Ripard [Wed, 8 Nov 2017 15:47:10 +0000 (16:47 +0100)]
ASoC: sun8i-codec: Fix left and right channels inversion
Since its introduction, the codec had an inversion of the left and right
channels. It turned out to be pretty simple as it appears that the codec
doesn't have the same polarity on the LRCK signal than the I2S block.
Fix this by inverting our bit value for the LRCK inversion.
Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Cc: <stable@vger.kernel.org> Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com> Reviewed-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Akshu Agrawal [Wed, 8 Nov 2017 17:24:02 +0000 (12:24 -0500)]
ASoC: amd: Make the driver name consistent across files
This fixes the issue of driver not getting auto loaded with
MODULE_ALIAS.
find /sys/devices -name modalias -print0 | xargs -0 grep 'audio'
/sys/devices/pci0000:00/0000:00:01.0/acp_audio_dma.0.auto/modalias:platform:acp_audio_dma
TEST=boot and check for device in lsmod
[Removed yet more ChromeOS crap from the changelog -- broonie]
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Tested-by: Jason Clinton <jclinton@chromium.org> Reviewed-by: Jason Clinton <jclinton@chromium.org> Signed-off-by: Alex Deucher <alexander.deucher@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The problem here is that the sound/soc/intel/common/ directory
is then entered only for building modules, but the sst-acpi.o
never gets built since it depends on a built-in Kconfig symbol.
That configuration obviously makes no sense since all options
below SND_SOC_INTEL_MACH also depend on something else that
in turn depends on CONFIG_SND_SOC_INTEL_COMMON.
Adding a SND_SOC_INTEL_SST_TOPLEVEL dependency to SND_SOC_INTEL_MACH
solves the build error. I notice we can also consolidate the
'depends on SND_SOC_INTEL_MACH' lines by using an 'if' block to
simplify it further and make sure the configuration stays sane.
Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-By: Vinod Koul <vinod.koul@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Wed, 8 Nov 2017 13:03:19 +0000 (14:03 +0100)]
ASoC: Intel: improve DMADEVICES dependency
As pointed out by Pierre-Louis Bossart, the dependency I added
was broader than necessary, only Baytrail and Haswell/Broadwell
actually need it, the others don't.
At the same time, we have individual entries for the codecs
that all have the 'select' statement but now don't need it
any more.
Fixes: f7a88db6fffd ("ASoC: Intel: fix Kconfig dependencies") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-By: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Signed-off-by: Akshu Agrawal <Akshu.Agrawal@amd.com> Tested-by: Akshu Agrawal <akshu.agrawal@amd.com> Reviewed-by: Jason Clinton <jclinton@chromium.org> Signed-off-by: Alex Deucher <alexander.deucher@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Oder Chiou [Wed, 8 Nov 2017 11:21:47 +0000 (19:21 +0800)]
ASoC: rt5514-spi: Let the buf_size to align with period_bytes
The patch lets the buf_size to align with period_bytes to prevent the
buffer reading over the real size of the DSP buffer and also avoid to
calculate the wrong size of remaining data.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Thu, 7 Sep 2017 13:22:48 +0000 (14:22 +0100)]
ASoC: pcm512x: Scrub my work address from the driver
It's difficult for me to handle upstream mail that ends up in my work
account and this was done outside of work anyway so replace my work
address with my usual address for upstream stuff.
Takashi Iwai [Tue, 7 Nov 2017 15:05:24 +0000 (16:05 +0100)]
ALSA: seq: Fix OSS sysex delivery in OSS emulation
The SYSEX event delivery in OSS sequencer emulation assumed that the
event is encoded in the variable-length data with the straight
buffering. This was the normal behavior in the past, but during the
development, the chained buffers were introduced for carrying more
data, while the OSS code was left intact. As a result, when a SYSEX
event with the chained buffer data is passed to OSS sequencer port,
it may end up with the wrong memory access, as if it were having a too
large buffer.
This patch addresses the bug, by applying the buffer data expansion by
the generic snd_seq_dump_var_event() helper function.
Takashi Iwai [Mon, 6 Nov 2017 19:16:50 +0000 (20:16 +0100)]
ALSA: seq: Avoid invalid lockdep class warning
The recent fix for adding rwsem nesting annotation was using the given
"hop" argument as the lock subclass key. Although the idea itself
works, it may trigger a kernel warning like:
BUG: looking up invalid subclass: 8
....
since the lockdep has a smaller number of subclasses (8) than we
currently allow for the hops there (10).
The current definition is merely a sanity check for avoiding the too
deep delivery paths, and the 8 hops are already enough. So, as a
quick fix, just follow the max hops as same as the max lockdep
subclasses.
Takashi Iwai [Mon, 6 Nov 2017 09:47:14 +0000 (10:47 +0100)]
ALSA: usx2y: Fix invalid stream URBs
The us122l driver creates URBs per the fixed endpoints, and this may
end up with URBs with inconsistent pipes when a fuzzer or a malicious
program deals with the manipulated endpoints. It ends up with a
kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 0 != type 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:471
usb_submit_urb+0x113e/0x1400
Call Trace:
usb_stream_start+0x48a/0x9f0 sound/usb/usx2y/usb_stream.c:690
us122l_start+0x116/0x290 sound/usb/usx2y/us122l.c:365
us122l_create_card sound/usb/usx2y/us122l.c:502
us122l_usb_probe sound/usb/usx2y/us122l.c:588
....
For avoiding the bad access, this patch adds a few sanity checks of
the validity of created URBs like previous similar fixes using the new
usb_urb_ep_type_check() helper function.
ASoC: rsnd: return -EIO if rsnd_dmaen_request_channel() failed
PTR_ERR(NULL) is success. Normally when a function returns both NULL
and error pointers, it means that NULL is not a error.
But, rsnd_dmaen_request_channel() returns NULL if requested resource
was failed.
Let's return -EIO if rsnd_dmaen_request_channel() was failed on
rsnd_dmaen_nolock_start().
This patch fixes commit edce5c496c6a ("ASoC: rsnd: Request/Release DMA
channel eachtime")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Sun, 5 Nov 2017 09:07:43 +0000 (10:07 +0100)]
ALSA: timer: Limit max instances per timer
Currently we allow unlimited number of timer instances, and it may
bring the system hogging way too much CPU when too many timer
instances are opened and processed concurrently. This may end up with
a soft-lockup report as triggered by syzkaller, especially when
hrtimer backend is deployed.
Since such insane number of instances aren't demanded by the normal
use case of ALSA sequencer and it merely opens a risk only for abuse,
this patch introduces the upper limit for the number of instances per
timer backend. As default, it's set to 1000, but for the fine-grained
timer like hrtimer, it's set to 100.
Linus Torvalds [Sun, 5 Nov 2017 20:14:50 +0000 (12:14 -0800)]
Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull x86 fixes from Ingo Molnar:
"Two fixes:
- A PCID related revert that fixes power management and performance
regressions.
- The module loader robustization and sanity check commit is rather
fresh, but it looked like a good idea to apply because of the
hidden data corruption problem such invalid modules could cause"
* 'x86-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
x86/module: Detect and skip invalid relocations
Revert "x86/mm: Stop calling leave_mm() in idle code"
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