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[ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ]
da7219_aad_suspend() disables jack detection, which should prevent
generating new interrupts by DA7219 while suspended. However, there is a
theoretical possibility that there is a pending interrupt generated just
before suspending DA7219 and not handled yet, so the IRQ handler may
still run after DA7219 is suspended. To prevent that, wait until the
pending IRQ handling is done.
This patch arose as an attempt to fix the following I2C failure
occurring sometimes during system suspend or resume:
[ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended
[ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440
...
[ 355.876462] Call Trace:
[ 355.876468] <TASK>
[ 355.876475] ? update_load_avg+0x1b3/0x615
[ 355.876484] __i2c_transfer+0x101/0x1d8
[ 355.876494] i2c_transfer+0x74/0x10d
[ 355.876504] regmap_i2c_read+0x6a/0x9c
[ 355.876513] _regmap_raw_read+0x179/0x223
[ 355.876521] regmap_raw_read+0x1e1/0x28e
[ 355.876527] regmap_bulk_read+0x17d/0x1ba
[ 355.876532] ? __wake_up+0xed/0x1bb
[ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184]
[ 355.876556] irq_thread+0x13c/0x231
[ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f
[ 355.876570] ? irq_thread_fn+0x4d/0x4d
[ 355.876576] kthread+0x13a/0x152
[ 355.876581] ? synchronize_irq+0xc3/0xc3
[ 355.876587] ? kthread_blkcg+0x31/0x31
[ 355.876592] ret_from_fork+0x1f/0x30
[ 355.876601] </TASK>
which indicates that the AAD IRQ handler is unexpectedly running when
DA7219 is suspended, and as a result, is trying to read data from DA7219
over I2C and is hitting the I2C driver "Transfer while suspended"
failure.
However, with this patch the above failure is still reproducible. So
this patch does not fix any real observed issue so far, but at least is
useful for confirming that the above issue is not caused by a pending
IRQ but rather looks like a DA7219 hardware issue with an IRQ
unexpectedly generated after jack detection is already disabled.
Signed-off-by: Dmytro Maluka <dmy@semihalf.com>
Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 011b559be832194f992f73d6c0d5485f5925a10b upstream.
Pointer substream is being dereferenced on the assignment of pointer card
before substream is being null checked with the macro PCM_RUNTIME_CHECK.
Although PCM_RUNTIME_CHECK calls BUG_ON, it still is useful to perform the
the pointer check before card is assigned.
Fixes: d4cfb30fce03 ("ALSA: pcm: Set per-card upper limit of PCM buffer allocations")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://lore.kernel.org/r/20220424205945.1372247-1-colin.i.king@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7d50b295c4af16f814ee82369c4a234df5228801 upstream.
This reverts commit 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when
direct DMA is available") also the modification commit 467fd0e82b62
("ALSA: pcm: Fix build error on m68k and others").
Poking the DMA internal helper is a layer violation, so we should
avoid that. Meanwhile the actual bug has been addressed by the
Kconfig fix in commit dbed452a078d ("dma-pool: decouple DMA_REMAP from
DMA_COHERENT_POOL"), so we can live without this hack.
Link: https://lore.kernel.org/r/20200717064130.22957-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 467fd0e82b6265b8e6cc078a70bd7592574d5c83 upstream.
The commit 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when direct
DMA is available") introduced a check of the DMA type and this caused
a build error on m68k (and possibly some others) due to the lack of
dma_is_direct() definition. Since the check is needed only for
CONFIG_SND_DMA_SGBUF enablement (i.e. solely x86), use #ifdef instead
of IS_ENABLED() for avoiding such a build error.
Fixes: 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when direct DMA is available")
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20200707111225.26826-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 0b2cbce6898600aae5e87285f1c2000162d59c76 ]
On i.MX8MQ/i.MX8MN/i.MX8MM platform, the sai IP is upgraded.
There are some new registers and new bit definition. This
patch is to complete the register list.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1600323079-5317-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 269f399dc19f ("ASoC: fsl_sai: Disable bit clock with transmitter")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 94741eba63c23b0f1527b0ae0125e6b553bde10e ]
Current code enables TCSR.TE and RCSR.RE together, and disable
TCSR.TE and RCSR.RE together in trigger(), which only supports
one operation mode:
1. Rx synchronous with Tx: TE is last enabled and first disabled
Other operation mode need to be considered also:
2. Tx synchronous with Rx: RE is last enabled and first disabled.
3. Asynchronous mode: Tx and Rx are independent.
So the enable TCSR.TE and RCSR.RE sequence and the disable
sequence need to be refined accordingly for #2 and #3.
There is slightly against what RM recommennds with this change.
For example in Rx synchronous with Tx mode, case "aplay 1.wav;
arecord 2.wav" enable TE before RE. But it should be safe to
do so, judging by years of testing results.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 269f399dc19f ("ASoC: fsl_sai: Disable bit clock with transmitter")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bd55842ed998a622ba6611fe59b3358c9f76773d ]
The PCM memory allocation helpers have a sanity check against too many
buffer allocations. However, the check is performed without a proper
lock and the allocation isn't serialized; this allows user to allocate
more memories than predefined max size.
Practically seen, this isn't really a big problem, as it's more or
less some "soft limit" as a sanity check, and it's not possible to
allocate unlimitedly. But it's still better to address this for more
consistent behavior.
The patch covers the size check in do_alloc_pages() with the
card->memory_mutex, and increases the allocated size there for
preventing the further overflow. When the actual allocation fails,
the size is decreased accordingly.
Reported-by: BassCheck <bass@buaa.edu.cn>
Reported-by: Tuo Li <islituo@gmail.com>
Link: https://lore.kernel.org/r/CADm8Tek6t0WedK+3Y6rbE5YEt19tML8BUL45N2ji4ZAz1KcN_A@mail.gmail.com
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230703112430.30634-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3ad796cbc36a7bc8bfd4de191d791b9490bc112b ]
The DMA-coherent SG-buffer is tricky to use, as it does need the
mapping. It used to work stably on x86 over years (and that's why we
had enabled SG-buffer on solely x86) with the default mmap handler and
vmap(), but our luck seems no forever success. The chance of breakage
is high when the special DMA handling is introduced in the arch side.
In this patch, we change the buffer allocation to use the SG-buffer
only when the device in question is with the direct DMA. It's a bit
hackish, but it's currently the only condition that may work (more or
less) reliably with the default mmap and vmap() for mapping the pages
that are deduced via virt_to_page().
In theory, we can apply the similar hack in the sound/core memory
allocation helper, too; but it's used by SOF for allocating SG pages
without re-mapping via vmap() or mmap, and it's fine to use it in that
way, so let's keep it and adds the workaround in PCM side.
Link: https://lore.kernel.org/r/20200615160045.2703-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: bd55842ed998 ("ALSA: pcm: Fix potential data race at PCM memory allocation helpers")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d4cfb30fce03093ad944e0b44bd8f40bdad5330e ]
Currently, the available buffer allocation size for a PCM stream
depends on the preallocated size; when a buffer has been preallocated,
the max buffer size is set to that size, so that application won't
re-allocate too much memory. OTOH, when no preallocation is done,
each substream may allocate arbitrary size of buffers as long as
snd_pcm_hardware.buffer_bytes_max allows -- which can be quite high,
HD-audio sets 1GB there.
It means that the system may consume a high amount of pages for PCM
buffers, and they are pinned and never swapped out. This can lead to
OOM easily.
For avoiding such a situation, this patch adds the upper limit per
card. Each snd_pcm_lib_malloc_pages() and _free_pages() calls are
tracked and it will return an error if the total amount of buffers
goes over the defined upper limit. The default value is set to 32MB,
which should be really large enough for usual operations.
If larger buffers are needed for any specific usage, it can be
adjusted (also dynamically) via snd_pcm.max_alloc_per_card option.
Setting zero there means no chceck is performed, and again, unlimited
amount of buffers are allowed.
Link: https://lore.kernel.org/r/20200120124423.11862-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: bd55842ed998 ("ALSA: pcm: Fix potential data race at PCM memory allocation helpers")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1f848f12103920ca165758aedb1c10904e193e1 ]
When the tdm lane mask is computed, the driver currently fills the 1st lane
before moving on to the next. If the stream has less channels than the
lanes can accommodate, slots will be disabled on the last lanes.
Unfortunately, the HW distribute channels in a different way. It distribute
channels in pair on each lanes before moving on the next slots.
This difference leads to problems if a device has an interface with more
than 1 lane and with more than 2 slots per lane.
For example: a playback interface with 2 lanes and 4 slots each (total 8
slots - zero based numbering)
- Playing a 8ch stream:
- All slots activated by the driver
- channel #2 will be played on lane #1 - slot #0 following HW placement
- Playing a 4ch stream:
- Lane #1 disabled by the driver
- channel #2 will be played on lane #0 - slot #2
This behaviour is obviously not desirable.
Change the way slots are activated on the TDM lanes to follow what the HW
does and make sure each channel always get mapped to the same slot/lane.
Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20230809171931.1244502-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 81302b1c7c997e8a56c1c2fc63a296ebeb0cd2d0 ]
It's reported that the recording started right after the driver probe
doesn't work properly, and it turned out that this is related with the
codec auto-suspend. Namely, after the probe phase, the usage count
goes zero, and the auto-suspend is programmed, but the codec is kept
still active until the auto-suspend expiration. When an application
(e.g. alsactl) updates the mixer values at this moment, the values are
cached but not actually written. Then, starting arecord thereafter
also results in the silence because of the missing unmute.
The root cause is the handling of "lazy update" mode; when a mixer
value is updated *after* the suspend, it should update only the cache
and exits. At the resume, the cached value is written to the device,
in turn. The problem is that the current code misinterprets the state
of auto-suspend as if it were already suspended.
Although we can add the check of the actual device state after
pm_runtime_get_if_in_use() for catching the missing state, this won't
suffice; the second call of regmap_update_bits_check() will skip
writing the register because the cache has been already updated by the
first call. So we'd need fixes in two different places.
OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with
pm_runtime_get_if_active() (with ign_usage_count=true). This change
implies that the driver takes the pm refcount if the device is still
in ACTIVE state and continues the processing. A small caveat is that
this will leave the auto-suspend timer. But, since the timer callback
itself checks the device state and aborts gracefully when it's active,
this won't be any substantial problem.
Long story short: we address the missing register-write problem just
by replacing the pm_runtime_*() call in snd_hda_keep_power_up().
Fixes: fc4f000bf8c0 ("ALSA: hda - Fix unexpected resume through regmap code path")
Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.com
Suggested-by: Cezary Rojewski <cezary.rojewski@intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230518113520.15213-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ]
The variable codec->regmap is often protected by the lock
codec->regmap_lock when is accessed. However, it is accessed without
holding the lock when is accessed in snd_hdac_regmap_sync():
if (codec->regmap)
In my opinion, this may be a harmful race, because if codec->regmap is
set to NULL right after the condition is checked, a null-pointer
dereference can occur in the called function regcache_sync():
map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c
To fix this possible null-pointer dereference caused by data race, the
mutex_lock coverage is extended to protect the if statement as well as the
function call to regcache_sync().
[ Note: the lack of the regmap_lock itself is harmless for the current
codec driver implementations, as snd_hdac_regmap_sync() is only for
PM runtime resume that is prohibited during the codec probe.
But the change makes the whole code more consistent, so it's merged
as is -- tiwai ]
Reported-by: BassCheck <bass@buaa.edu.cn>
Signed-off-by: Tuo Li <islituo@gmail.com>
Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ]
These models use NSIWAY amplifiers for internal speaker, but cannot put
sound outside from these amplifiers. So eapd verbs are needed to initialize
the amplifiers. They can be added during boot to get working sound out
of internal speaker.
Signed-off-by: dengxiang <dengxiang@nfschina.com>
Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e51df4f81b02bcdd828a04de7c1eb6a92988b61e upstream.
In commit 2cb1e0259f50 ("ASoC: cs42l51: re-hook of_match_table
pointer"), 9 years ago, some random guy fixed the cs42l51 after it was
split into a core part and an I2C part to properly match based on a
Device Tree compatible string.
However, the fix in this commit is wrong: the MODULE_DEVICE_TABLE(of,
....) is in the core part of the driver, not the I2C part. Therefore,
automatic module loading based on module.alias, based on matching with
the DT compatible string, loads the core part of the driver, but not
the I2C part. And threfore, the i2c_driver is not registered, and the
codec is not known to the system, nor matched with a DT node with the
corresponding compatible string.
In order to fix that, we move the MODULE_DEVICE_TABLE(of, ...) into
the I2C part of the driver. The cs42l51_of_match[] array is also moved
as well, as it is not possible to have this definition in one file,
and the MODULE_DEVICE_TABLE(of, ...) invocation in another file, due
to how MODULE_DEVICE_TABLE works.
Thanks to this commit, the I2C part of the driver now properly
autoloads, and thanks to its dependency on the core part, the core
part gets autoloaded as well, resulting in a functional sound card
without having to manually load kernel modules.
Fixes: 2cb1e0259f50 ("ASoC: cs42l51: re-hook of_match_table pointer")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Petazzoni <thomas.petazzoni@bootlin.com>
Link: https://lore.kernel.org/r/20230713112112.778576-1-thomas.petazzoni@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f061e2be8689057cb4ec0dbffa9f03e1a23cdcb2 upstream.
The WM8904_ADC_TEST_0 register is modified as part of updating the OSR
controls but does not have a cache default, leading to errors when we try
to modify these controls in cache only mode with no prior read:
wm8904 3-001a: ASoC: error at snd_soc_component_update_bits on wm8904.3-001a for register: [0x000000c6] -16
Add a read of the register to probe() to fill the cache and avoid both the
error messages and the misconfiguration of the chip which will result.
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20230723-asoc-fix-wm8904-adc-test-read-v1-1-2cdf2edd83fd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 0e4c2b6b0c4a4b4014d9424c27e5e79d185229c5 ]
Clear TX registers on stop to prevent the SPDIF interface from sending
last written word over and over again.
Fixes: a2388a498ad2 ("ASoC: fsl: Add S/PDIF CPU DAI driver")
Signed-off-by: Matus Gajdos <matuszpd@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20230719164729.19969-1-matuszpd@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 89dbb335cb6a627a4067bc42caa09c8bc3326d40 ]
snd_jack_report() is supposed to be callable from an IRQ context, too,
and it's indeed used in that way from virtsnd driver. The fix for
input_dev race in commit 1b6a6fc5280e ("ALSA: jack: Access input_dev
under mutex"), however, introduced a mutex lock in snd_jack_report(),
and this resulted in a potential sleep-in-atomic.
For addressing that problem, this patch changes the relevant code to
use the object get/put and removes the mutex usage. That is,
snd_jack_report(), it takes input_get_device() and leaves with
input_put_device() for assuring the input_dev being assigned.
Although the whole mutex could be reduced, we keep it because it can
be still a protection for potential races between creation and
deletion.
Fixes: 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex")
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Closes: https://lore.kernel.org/r/cf95f7fe-a748-4990-8378-000491b40329@moroto.mountain
Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230706155357.3470-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 60413129ee2b38a80347489270af7f6e1c1de4d0 ]
When using the codec through the generic audio graph card, there are at
least two calls of es8316_set_dai_sysclk(), with the effect of limiting
the allowed sample rates according to the MCLK/LRCK ratios supported by
the codec:
1. During audio card setup, to set the initial MCLK - see
asoc_simple_init_dai().
2. Before opening a stream, to update MCLK, according to the stream
sample rate and the multiplication factor - see
asoc_simple_hw_params().
In some cases the initial MCLK might be set to a frequency that doesn't
match any of the supported ratios, e.g. 12287999 instead of 12288000,
which is only 1 Hz below the supported clock, as that is what the
hardware reports. This creates an empty list of rate constraints, which
is further passed to snd_pcm_hw_constraint_list() via
es8316_pcm_startup(), and causes the following error on the very first
access of the sound card:
$ speaker-test -D hw:Analog,0 -F S16_LE -c 2 -t wav
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument
Note that all subsequent retries succeed thanks to the updated MCLK set
at point 2 above, which uses a computed frequency value instead of a
reading from the hardware registers. Normally this would have mitigated
the issue, but es8316_pcm_startup() executes before the 2nd call to
es8316_set_dai_sysclk(), hence it cannot make use of the updated
constraints.
Since es8316_pcm_hw_params() performs anyway a final validation of MCLK
against the stream sample rate and the supported MCLK/LRCK ratios, fix
the issue by ensuring that sysclk_constraints list is only set when at
least one supported sample rate is autodetected by the codec.
Fixes: b8b88b70875a ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-3-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6f073429037cd79d7311cd8236311c53f5ea8f01 ]
The following error occurs when trying to restore a previously saved
ALSA mixer state (tested on a Rock 5B board):
$ alsactl --no-ucm -f /tmp/asound.state store hw:Analog
$ alsactl --no-ucm -I -f /tmp/asound.state restore hw:Analog
alsactl: set_control:1475: Cannot write control '2:0:0:ALC Capture Target Volume:0' : Invalid argument
According to ES8316 datasheet, the register at address 0x2B, which is
related to the above mixer control, contains by default the value 0xB0.
Considering the corresponding ALC target bits (ALCLVL) are 7:4, the
control is initialized with 11, which is one step above the maximum
value allowed by the driver:
ALCLVL | dB gain
-------+--------
0000 | -16.5
0001 | -15.0
0010 | -13.5
.... | .....
0111 | -6.0
1000 | -4.5
1001 | -3.0
1010 | -1.5
.... | .....
1111 | -1.5
The tests performed using the VU meter feature (--vumeter=TYPE) of
arecord/aplay confirm the specs are correct and there is no measured
gain if the 1011-1111 range would have been mapped to 0 dB:
dB gain | VU meter %
--------+-----------
-6.0 | 30-31
-4.5 | 35-36
-3.0 | 42-43
-1.5 | 50-51
0.0 | 50-51
Increment the max value allowed for ALC Capture Target Volume control,
so that it matches the hardware default. Additionally, update the
related TLV to prevent an artificial extension of the dB gain range.
Fixes: b8b88b70875a ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ]
Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the
DMI table, so that active-high jack-detect will work properly on
these laptops.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 7ca4c8d4d3f41c2cd9b4cf22bb829bf03dac0956 upstream.
Headset microphone on this platform does not work without
ALC897_FIXUP_HEADSET_MIC_PIN fixup.
Signed-off-by: RenHai <kean0048@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230602003604.975892-1-kean0048@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 527c356b51f3ddee02c9ed5277538f85e30a2cdc upstream.
Add a quirk for HP Slim Desktop S01 to fixup headset MIC no presence.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230526094704.14597-1-aichao@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3f649ab728cda8038259d8f14492fe400fbab911 upstream.
Using uninitialized_var() is dangerous as it papers over real bugs[1]
(or can in the future), and suppresses unrelated compiler warnings
(e.g. "unused variable"). If the compiler thinks it is uninitialized,
either simply initialize the variable or make compiler changes.
In preparation for removing[2] the[3] macro[4], remove all remaining
needless uses with the following script:
git grep '\buninitialized_var\b' | cut -d: -f1 | sort -u | \
xargs perl -pi -e \
's/\buninitialized_var\(([^\)]+)\)/\1/g;
s:\s*/\* (GCC be quiet|to make compiler happy) \*/$::g;'
drivers/video/fbdev/riva/riva_hw.c was manually tweaked to avoid
pathological white-space.
No outstanding warnings were found building allmodconfig with GCC 9.3.0
for x86_64, i386, arm64, arm, powerpc, powerpc64le, s390x, mips, sparc64,
alpha, and m68k.
[1] https://lore.kernel.org/lkml/20200603174714.192027-1-glider@google.com/
[2] https://lore.kernel.org/lkml/CA+55aFw+Vbj0i=1TGqCR5vQkCzWJ0QxK6CernOU6eedsudAixw@mail.gmail.com/
[3] https://lore.kernel.org/lkml/CA+55aFwgbgqhbp1fkxvRKEpzyR5J8n1vKT1VZdz9knmPuXhOeg@mail.gmail.com/
[4] https://lore.kernel.org/lkml/CA+55aFz2500WfbKXAx8s67wrm9=yVJu65TpLgN_ybYNv0VEOKA@mail.gmail.com/
Reviewed-by: Leon Romanovsky <leonro@mellanox.com> # drivers/infiniband and mlx4/mlx5
Acked-by: Jason Gunthorpe <jgg@mellanox.com> # IB
Acked-by: Kalle Valo <kvalo@codeaurora.org> # wireless drivers
Reviewed-by: Chao Yu <yuchao0@huawei.com> # erofs
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ]
Two functions are defined and used in pcm_oss.c but also optionally
used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS
is disabled, this causes a warning as the functions are not static
and have no prototype:
sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes]
sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes]
Avoid this by making the prototypes unconditional.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ]
Apply a workaround for what appears to be a hardware quirk.
The problem seems to happen when enabling "whole chip power" (bit D7
register R6) for the very first time after the chip receives power. If
either "output" (D4) or "DAC" (D3) aren't powered on at that time,
playback becomes very distorted later on.
This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10:
https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset-issue/480229
I suspect this happens only when using an external MCLK signal (which
is the case for both of these boards).
Here are some experiments run on a Google Chameleon v3. These were run
in userspace using a wrapper around the i2cset utility:
ssmset() {
i2cset -y 0 0x1a $(($1*2)) $2
}
For each of the following sequences, we apply power to the ssm2603
chip, set the configuration registers R0-R5 and R7-R8, run the selected
sequence, and check for distortions on playback.
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x87 # out, dac
ssmset 0x06 0x07 # chip
OK
(disable MCLK)
ssmset 0x09 0x01 # core
ssmset 0x06 0x1f # chip
ssmset 0x06 0x07 # out, dac
(enable MCLK)
OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x1f # chip
ssmset 0x06 0x07 # out, dac
NOT OK
ssmset 0x06 0x1f # chip
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # out, dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x0f # chip, out
ssmset 0x06 0x07 # dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x17 # chip, dac
ssmset 0x06 0x07 # out
NOT OK
For each of the following sequences, we apply power to the ssm2603
chip, run the selected sequence, issue a reset with R15, configure
R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check
for distortions.
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
OK
(disable MCLK)
ssmset 0x09 0x01 # core
ssmset 0x06 0x07 # chip, out, dac
(enable MCLK after reset)
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x17 # chip, dac
NOT OK
ssmset 0x09 0x01 # core
ssmset 0x06 0x0f # chip, out
NOT OK
ssmset 0x06 0x07 # chip, out, dac
NOT OK
Signed-off-by: Paweł Anikiel <pan@semihalf.com
Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ]
On slow CPU (FPGA/QEMU emulated) printing overrun messages from
interrupt handler to uart console may leads to more overrun errors.
So use dev_err_ratelimited to limit the number of error messages.
Signed-off-by: Maxim Kochetkov <fido_max@inbox.ru
Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4ca110cab46561cd74a2acd9b447435acb4bec5f upstream.
Lenovo M70/M90 Gen4 are equipped with ALC897, and they need
ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work.
The previous quirk for M70/M90 is for Gen3.
Signed-off-by: Bin Li <bin.li@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230524113755.1346928-1-bin.li@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7843380d07bbeffd3ce6504e73cf61f840ae76ca upstream.
This quirk is necessary for surround and other DSP effects to work
with the onboard ca0132 based audio chipset for the EVGA X299 dark
mainboard.
Signed-off-by: Adam Stylinski <kungfujesus06@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=67071
Link: https://lore.kernel.org/r/ZGopOe19T1QOwizS@eggsbenedict.adamsnet
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a4671b7fba59775845ee60cfbdfc4ba64300211b upstream.
Add quirk for GU603 with 0x1c62 variant of codec.
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230505235824.49607-2-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 90670ef774a8b6700c38ce1222e6aa263be54d5f upstream.
Add a quirk for HP EliteDesk 805 to fixup ALC3867 headset MIC no sound.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230506022653.2074343-1-aichao@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit dc4f2ccaedddb489a83e7b12ebbdc347272aacc9 upstream.
These IDs are for AD102, AD103, AD104, AD106, and AD107 gpus with
audio functions that are largely similar to the existing ones.
Tested audio using gnome-settings, over HDMI, DP-SST and DP-MST
connections on AD106 gpu.
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230517090736.15088-1-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3b44ec8c5c44790a82f07e90db45643c762878c6 upstream.
get_line_out_pfx() may trigger an Oops by overflowing the static array
with more than 8 channels. This was reported for MacBookPro 12,1 with
Cirrus codec.
As a workaround, extend for the 9.1 channels and also fix the
potential Oops by unifying the code paths accessing the same array
with the proper size check.
Reported-by: Olliver Schinagl <oliver@schinagl.nl>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/64d95eb0-dbdb-cff8-a8b1-988dc22b24cd@schinagl.nl
Link: https://lore.kernel.org/r/20230516184412.24078-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit c0e72058d5e21982e61a29de6b098f7c1f0db498 ]
This code was supposed to return an error code if init_stream()
failed, but it instead freed dg00x->rx_stream and returned success.
This potentially leads to a use after free.
Fixes: 9a08067ec318 ("ALSA: firewire-digi00x: support AMDTP domain")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://lore.kernel.org/r/c224cbd5-d9e2-4cd4-9bcf-2138eb1d35c6@kili.mountain
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0adf292069dcca8bab76a603251fcaabf77468ca ]
There is no defer probe when adding platform component to
snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime()
snd_soc_register_card()
-> snd_soc_bind_card()
-> snd_soc_add_pcm_runtime()
-> adding cpu dai
-> adding codec dai
-> adding platform component.
So if the platform component is not ready at that time, then the
sound card still registered successfully, but platform component
is empty, the sound card can't be used.
As there is defer probe checking for cpu dai component, then register
platform component before cpu dai to avoid such issue.
Fixes: 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1630665006-31437-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0d727e1856ef22dd9337199430258cb64cbbc658 ]
Smatch complains that:
snd_usb_caiaq_input_init() warn: missing error code 'ret'
This patch adds a new case to handle the situation where the
device does not support any input methods in the
`snd_usb_caiaq_input_init` function. It returns an `-EINVAL` error code
to indicate that no input methods are supported on the device.
Fixes: 523f1dce3743 ("[ALSA] Add Native Instrument usb audio device support")
Signed-off-by: Ruliang Lin <u202112092@hust.edu.cn>
Reviewed-by: Dongliang Mu <dzm91@hust.edu.cn>
Acked-by: Daniel Mack <daniel@zonque.org>
Link: https://lore.kernel.org/r/20230504065054.3309-1-u202112092@hust.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 39db65a0a17b54915b269d3685f253a4731f344c ]
The driver is able to work fine without relying on a mandatory interrupt
being assigned to the I2C device. This is only needed when making use of
the jack-detect support.
However, the following warning message is always emitted when there is
no such interrupt available:
es8316 0-0011: Failed to get IRQ 0: -22
Do not attempt to request an IRQ if it is not available/valid. This also
ensures the rather misleading message is not displayed anymore.
Also note the IRQ validation relies on commit dab472eb931bc291 ("i2c /
ACPI: Use 0 to indicate that device does not have interrupt assigned").
Fixes: 822257661031 ("ASoC: es8316: Add jack-detect support")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20230328094901.50763-1-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1cf2aa665901054b140eb71748661ceae99b6b5a ]
Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then
disabling it immediately after requesting it.
This fixes a possible race where the IRQ might trigger between requesting
and disabling it; and this also leads to a small code cleanup.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211003132255.31743-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 39db65a0a17b ("ASoC: es8316: Handle optional IRQ assignment")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e38c5e80c3d293a883c6f1d553f2146ec0bda35e ]
The Acer Iconia One 7 B1-750 tablet mostly works fine with the defaults
for an Bay Trail CR tablet. Except for the internal mic, instead of
an analog mic on IN3 a digital mic on DMIC1 is uses.
Add a quirk with these settings for this tablet.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20230322145332.131525-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit f342ac00da1064eb4f94b1f4bcacbdfea955797a upstream.
The BIOS botches this one completely - it says the 2nd S/PDIF output is
used, while in fact it's the 1st one. This is tested on DP45SG, but I'm
assuming it's valid for the other boards in the series as well.
Also add some comments regarding the pins.
FWIW, the codec is apparently still sold by Tempo Semiconductor, Inc.,
where one can download the documentation.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197826-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fb4a624f88f658c7b7ae124452bd42eaa8ac7168 upstream.
Smatch Warns:
sound/firewire/tascam/tascam-stream.c:493 snd_tscm_stream_start_duplex()
warn: missing unwind goto?
The direct return will cause the stream list of "&tscm->domain" unemptied
and the session in "tscm" unfinished if amdtp_domain_start() returns with
an error.
Fix this by changing the direct return to a goto which will empty the
stream list of "&tscm->domain" and finish the session in "tscm".
The snd_tscm_stream_start_duplex() function is called in the prepare
callback of PCM. According to "ALSA Kernel API Documentation", the prepare
callback of PCM will be called many times at each setup. So, if the
"&d->streams" list is not emptied, when the prepare callback is called
next time, snd_tscm_stream_start_duplex() will receive -EBUSY from
amdtp_domain_add_stream() that tries to add an existing stream to the
domain. The error handling code after the "error" label will be executed
in this case, and the "&d->streams" list will be emptied. So not emptying
the "&d->streams" list will not cause an issue. But it is more efficient
and readable to empty it on the first error by changing the direct return
to a goto statement.
The session in "tscm" has been begun before amdtp_domain_start(), so it
needs to be finished when amdtp_domain_start() fails.
Fixes: c281d46a51e3 ("ALSA: firewire-tascam: support AMDTP domain")
Signed-off-by: Xu Biang <xubiang@hust.edu.cn>
Reviewed-by: Dan Carpenter <error27@gmail.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230406132801.105108-1-xubiang@hust.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e98e7a82bca2b6dce3e03719cff800ec913f9af7 upstream.
snd_cs8427_iec958_active() would always delete
SNDRV_CTL_ELEM_ACCESS_INACTIVE, even though the function has an
argument `active`.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201219.2197811-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c17f8fd31700392b1bb9e7b66924333568cb3700 upstream.
Like the other boards from the D*45* series, this one sets up the
outputs not quite correctly.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197826-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b09c551c77c7e01dc6e4f3c8bf06b5ffa7b06db5 upstream.
Due to two copy/pastos, closing the MIC or EFX capture device would
make a running ADC capture hang due to unsetting its interrupt handler.
In principle, this would have also allowed dereferencing dangling
pointers, but we're actually rather thorough at disabling and flushing
the ints.
While it may sound like one, this actually wasn't a hypothetical bug:
PortAudio will open a capture stream at startup (and close it right
away) even if not asked to. If the first device is busy, it will just
proceed with the next one ... thus killing a concurrent capture.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197923-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>