Charles Keepax [Wed, 14 Jun 2023 14:21:16 +0000 (15:21 +0100)]
ASoC: intel: sof_sdw: Fixup typo in device link checking
The loop checking for multiple different devices on a single sdw link
contains a typo accidentally using i twice instead of j. Correct to the
correct index variable.
Takashi Iwai [Fri, 16 Jun 2023 07:28:27 +0000 (09:28 +0200)]
Merge tag 'asoc-fix-v6.4-rc6-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
A couple more fixes for v6.4, one fixing a misleading error log and
another stopping us seeing spurious failures setting the master volume
on some Tegra systems introduced by a change to how we calculate delay
times.
Lukasz Tyl [Wed, 14 Jun 2023 12:25:24 +0000 (14:25 +0200)]
ALSA: usb-audio: Add quirk flag for HEM devices to enable native DSD playback
This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for
HEM devices which supports native DSD. Prior to this change Linux kernel
was not enabling native DSD playback for HEM devices, and as a result,
DSD audio was being converted to PCM "on the fly". HEM devices,
when connected to the system, would only play audio in PCM format,
even if the source material was in DSD format. With the addition of new
VENDOR_FLG in the quircks.c file, the devices are now correctly
recognized, and raw DSD data is transmitted to the device,
allowing for native DSD playback.
Takashi Iwai [Mon, 12 Jun 2023 13:28:18 +0000 (15:28 +0200)]
ALSA: usb-audio: Fix broken resume due to UAC3 power state
As reported in the bugzilla below, the PM resume of a UAC3 device may
fail due to the incomplete power state change, stuck at D1. The
reason is that the driver expects the full D0 power state change only
at hw_params, while the normal PCM resume procedure doesn't call
hw_params.
For fixing the bug, we add the same power state update to D0 at the
prepare callback, which is certainly called by the resume procedure.
Note that, with this change, the power state change in the hw_params
becomes almost redundant, since snd_usb_hw_params() doesn't touch the
parameters (at least it tires so). But dropping it is still a bit
risky (e.g. we have the media-driver binding), so I leave the D0 power
state change in snd_usb_hw_params() as is for now.
Takashi Iwai [Mon, 12 Jun 2023 12:55:33 +0000 (14:55 +0200)]
ALSA: seq: oss: Fix racy open/close of MIDI devices
Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be
called concurrently from different code paths, we have no proper data
protection against races. Introduce open_mutex to each seq_oss_midi
object for avoiding the races.
Jon Hunter [Tue, 13 Jun 2023 09:34:53 +0000 (10:34 +0100)]
ASoC: tegra: Fix Master Volume Control
Commit 3ed2b549b39f ("ALSA: pcm: fix wait_time calculations") corrected
the PCM wait_time calculations and in doing so reduced the calculated
wait_time. This exposed an issue with the Tegra Master Volume Control
(MVC) device where the reduced wait_time caused the MVC to fail. For now
fix this by setting the default wait_time for Tegra to be 500ms.
Chris Chiu [Tue, 6 Jun 2023 14:57:47 +0000 (22:57 +0800)]
ALSA: hda/realtek: Enable 4 amplifiers instead of 2 on a HP platform
In the commit 7bb62340951a ("ALSA: hda/realtek: fix speaker, mute/micmute
LEDs not work on a HP platform"), speakers and LEDs are fixed but only 2
CS35L41 amplifiers on SPI bus connected to Realtek codec are enabled. Need
the ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED to get all amplifiers working.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com> Fixes: 7bb62340951a ("ALSA: hda/realtek: fix speaker, mute/micmute LEDs not work on a HP platform") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230606145747.135966-1-chris.chiu@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 13:09:42 +0000 (15:09 +0200)]
Merge tag 'asoc-fix-v6.4-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
A lot of routine driver specific fixes here, nothing in the core though
there are a couple of fixes for the generic cards. There's also a few
new quirks for x86 platforms.
Takashi Iwai [Tue, 6 Jun 2023 09:38:55 +0000 (11:38 +0200)]
ALSA: hda: Fix kctl->id initialization
HD-audio core code replaces the kctl->id.index of SPDIF-related
controls after assigning via snd_ctl_add(). This doesn't work any
longer with the new Xarray lookup change. The change of the kctl->id
content has to be done via snd_ctl_rename_id() helper, instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:54 +0000 (11:38 +0200)]
ALSA: gus: Fix kctl->id initialization
GUS driver replaces the kctl->id.index after assigning the kctl via
snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:53 +0000 (11:38 +0200)]
ALSA: cmipci: Fix kctl->id initialization
cmipci driver replaces the kctl->id.device after assigning the kctl
via snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Jun 2023 09:38:52 +0000 (11:38 +0200)]
ALSA: ymfpci: Fix kctl->id initialization
ymfpci driver replaces the kctl->id.device after assigning the kctl
via snd_ctl_add(). This doesn't work any longer with the new Xarray
lookup change. It has to be set before snd_ctl_add() call instead.
Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: <stable@vger.kernel.org> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230606093855.14685-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
firmware: cs_dsp: Log correct region name in bin error messages
In cs_dsp_load_coeff() region_name should be set in the XM/YM/ZM
cases otherwise any errors will log the region as "Unknown".
While doing this also change one error message that logged the
region type ID to log the region_name instead. This makes it
consistent with other messages in the same function.
selftests: alsa: pcm-test: Fix compiler warnings about the format
GCC 11.3.0 issues warnings in this module about wrong sizes of format
specifiers:
pcm-test.c: In function ‘test_pcm_time’:
pcm-test.c:384:68: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 5 \
has type ‘unsigned int’ [-Wformat=]
384 | snprintf(msg, sizeof(msg), "rate mismatch %ld != %ld", rate, rrate);
pcm-test.c:455:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
455 | "expected %d, wrote %li", rate, frames);
pcm-test.c:462:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
462 | "expected %d, wrote %li", rate, frames);
pcm-test.c:467:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \
type ‘long int’ [-Wformat=]
467 | "expected %d, wrote %li", rate, frames);
Simple fix according to compiler's suggestion removed the warnings.
Chancel Liu [Tue, 30 May 2023 10:30:12 +0000 (18:30 +0800)]
ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted
There's an issue on SAI synchronous mode that TX/RX side can't get BCLK
from RX/TX it sync with if BYP bit is asserted. It's a workaround to
fix it that enable SION of IOMUX pad control and assert BCI.
For example if TX sync with RX which means both TX and RX are using clk
form RX and BYP=1. TX can get BCLK only if the following two conditions
are valid:
1. SION of RX BCLK IOMUX pad is set to 1
2. BCI of TX is set to 1
The code in asoc_simple_startup was treating any non-zero return from
snd_pcm_hw_constraint_minmax as an error, when this can return 1 in some
normal cases and only negative values indicate an error.
When this happened, it caused asoc_simple_startup to disable the clocks
it just enabled and return 1, which was not treated as an error by the
calling code which only checks for negative return values. Then when the
PCM is eventually shut down, it causes the clock framework to complain
about disabling clocks that were not enabled.
Fix the check for snd_pcm_hw_constraint_minmax return value to only
treat negative values as an error.
Mark Brown [Thu, 1 Jun 2023 15:43:38 +0000 (16:43 +0100)]
ASoC: mediatek: fix use-after-free in driver remove
Merge series from Trevor Wu <trevor.wu@mediatek.com>:
These patches concern modifications made in mt8186[1]. The clock
unregistration mechanism used in mt8188 and mt8195 is similar with
mt8186, resulting in the same problem existing within the driver.
Therefore, the solution has also been applied to these two platforms.
Trevor Wu [Thu, 1 Jun 2023 03:33:18 +0000 (11:33 +0800)]
ASoC: mediatek: mt8195: fix use-after-free in driver remove path
During mt8195_afe_init_clock(), mt8195_audsys_clk_register() was called
followed by several other devm functions. At mt8195_afe_deinit_clock()
located at mt8195_afe_pcm_dev_remove(), mt8195_audsys_clk_unregister()
was called.
However, there was an issue with the order in which these functions were
called. Specifically, the remove callback of platform_driver was called
before devres released the resource, resulting in a use-after-free issue
during remove time.
At probe time, the order of calls was:
1. mt8195_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8195_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, we can utilize devm_add_action_or_reset() in
mt8195_audsys_clk_register() so that the remove order can be changed to
3->2->1.
Fixes: 6746cc858259 ("ASoC: mediatek: mt8195: add platform driver") Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Reviewed-by: Douglas Anderson <dianders@chromium.org> Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> Link: https://lore.kernel.org/r/20230601033318.10408-3-trevor.wu@mediatek.com Signed-off-by: Mark Brown <broonie@kernel.org>
Trevor Wu [Thu, 1 Jun 2023 03:33:17 +0000 (11:33 +0800)]
ASoC: mediatek: mt8188: fix use-after-free in driver remove path
During mt8188_afe_init_clock(), mt8188_audsys_clk_register() was called
followed by several other devm functions. The caller of
mt8188_afe_init_clock() utilized devm_add_action_or_reset() to call
mt8188_afe_deinit_clock(). However, the order was incorrect, causing a
use-after-free issue during remove time.
At probe time, the order of calls was:
1. mt8188_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8188_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, it's necessary to move devm_add_action_or_reset()
to the appropriate position so that the remove order can be 3->2->1.
Fixes: f6b026479b13 ("ASoC: mediatek: mt8188: support audio clock control") Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Reviewed-by: Douglas Anderson <dianders@chromium.org> Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> Link: https://lore.kernel.org/r/20230601033318.10408-2-trevor.wu@mediatek.com Signed-off-by: Mark Brown <broonie@kernel.org>
Vijendar Mukunda [Thu, 25 May 2023 11:29:55 +0000 (16:59 +0530)]
ASoC: amd: ps: fix for acp_lock access in pdm driver
Sending the mutex address(acp_lock) as platform
data during ACP PDM platform driver register sequence,
its creating copy of the platform data.
Referencing this platform data in ACP PDM driver results
incorrect reference to the common lock usage.
Instead of directly passing the lock address as platform
data, retrieve it from parent driver data structure
and use the same lock reference in ACP PDM driver.
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.
ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
ASoC: codecs: wsa881x: do not set can_multi_write flag
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
ASoC: codecs: wsa883x: do not set can_multi_write flag
regmap-sdw does not support multi register writes, so there is
no point in setting this flag. This also leads to incorrect
programming of WSA codecs with regmap_multi_reg_write() call.
This invalid configuration should have been rejected by regmap-sdw.
Takashi Iwai [Wed, 24 May 2023 10:14:24 +0000 (12:14 +0200)]
Merge tag 'asoc-fix-v6.4-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
A collection of fixes for v6.4, mostly driver specific but there's also
one fix for DPCM to avoid incorrectly repeated calls to prepare() which
can trigger issues on some systems.
It happens because of play_dma_data/capture_dma_data pointers are NULL.
Current implementation assigns these pointers at snd_soc_dai_driver
startup() callback and reset them back to NULL at shutdown(). But
soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open()
before snd_soc_dai_driver startup().
Most generic DMA capable I2S drivers use snd_soc_dai_driver probe()
callback to init DMA pointers only once at probe. So move DMA init
to dw_i2s_dai_probe and drop shutdown() and startup() callbacks.
Takashi Iwai [Thu, 18 May 2023 11:35:20 +0000 (13:35 +0200)]
ALSA: hda: Fix unhandled register update during auto-suspend period
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().
All IPCs using instance_id use 8 bit value. Original commit used 16 bit
value because FW reports possible max value in 16 bit field, but in
practice FW limits the value to 8 bits.
Cezary Rojewski [Fri, 19 May 2023 20:17:09 +0000 (22:17 +0200)]
ASoC: Intel: avs: Account for UID of ACPI device
Configurations with multiple codecs attached to the platform are
supported but only if each from the set is different. Add new field
representing the 'Unique ID' so that codecs that share Vendor and Part
IDs can be differentiated and thus enabling support for such
configurations.
When changing value of kcontrol, FW module to which data should be send
needs to be found. Currently it is done in improper way, fix it. Change
function name to indicate that it looks only for volume module.
This allows to change volume during runtime, instead of only changing
init value.
In the BE hw_params configuration, the existing code checks if any of the
existing FEs are prepared, running, paused or suspended - and skips the
configuration in those cases. This allows multiple calls of hw_params
which the ALSA state machine supports.
This check is not handled for the prepare stage, which can lead to the
same BE being prepared multiple times. This patch adds a check similar to
that of the hw_params, with the main difference being that the suspended
state is allowed: the ALSA state machine allows a transition from
suspended to prepared with hw_params skipped.
This problem was detected on Intel IPC4/SoundWire devices, where the BE
dailink .prepare stage is used to configure the SoundWire stream with a
bank switch. Multiple .prepare calls lead to conflicts with the .trigger
operation with IPC4 configurations. This problem was not detected earlier
on Intel devices, HDaudio BE dailinks detect that the link is already
prepared and skip the configuration, and for IPC3 devices there is no BE
trigger.
Arnd Bergmann [Tue, 16 May 2023 19:50:42 +0000 (21:50 +0200)]
ALSA: oss: avoid missing-prototype warnings
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.
Arnd Bergmann [Tue, 16 May 2023 19:50:41 +0000 (21:50 +0200)]
ALSA: cs46xx: mark snd_cs46xx_download_image as static
snd_cs46xx_download_image() was originally called from dsp_spos.c, but
is now local to cs46xx_lib.c. Mark it as 'static' to avoid a warning
about it lacking a declaration, and '__maybe_unused' to avoid a warning
about it being unused when CONFIG_SND_CS46XX_NEW_DSP is disabled:
sound/pci/cs46xx/cs46xx_lib.c:534:5: error: no previous prototype for 'snd_cs46xx_download_image'
ASoC: rt5682: Disable jack detection interrupt during suspend
The rt5682 driver switches its regmap to cache-only when the
device suspends and back to regular mode on resume. When the
jack detect interrupt fires rt5682_irq() schedules the jack
detect work. This can result in invalid reads from the regmap
in cache-only mode if the work runs before the device has
resumed:
[ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16
Disable the jack detection interrupt during suspend and
re-enable it on resume. The driver already schedules the
jack detection work on resume, so any state change during
suspend is still handled.
This is essentially the same as commit f7d00a9be147 ("SoC:
rt5682s: Disable jack detection interrupt during suspend")
for the rt5682s.
Takashi Iwai [Tue, 16 May 2023 18:44:12 +0000 (20:44 +0200)]
ALSA: hda: Fix Oops by 9.1 surround channel names
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.
Takashi Iwai [Tue, 16 May 2023 18:11:50 +0000 (20:11 +0200)]
Merge tag 'asoc-fix-v6.4-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.4
More fixes that came in since the merge window, the bulk of which are
for the SOF code, I suspect as a result of the wide usage, active
development and large code size rather than huge quality problems.
There's also a couple of MAINTAINERS updates and some new device quirks.
Mark Brown [Mon, 15 May 2023 15:13:09 +0000 (00:13 +0900)]
ASoC: SOF: Intel: hda-mlink: fixes and extensions
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
With additional testing with multiple links and multiple DAI types, we
found a couple of mistakes with refcounts, base address, missing
initialization.
A new helper was also added due to a change in the SoundWire
programming sequences, with the host driver in charge of setting up
the DMA channel mapping instead of the firmware.
The memory allocated for the tuples array assumes that there's 1
instance of all tokens already. So for those tokens that have multiple
instances in topology, we need to exclude the initial instance that has
already been accounted for.
Fixes: 4fdef47a44d6 ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230515085200.17094-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
Simon Trimmer [Fri, 12 May 2023 14:42:37 +0000 (15:42 +0100)]
ASoC: cs35l56: Prevent unbalanced pm_runtime in dsp_work() on SoundWire
Flush the SoundWire interrupt handler work instead of cancelling it.
When a SoundWire interrupt is triggered the pm_runtime is held
until the work has completed. It's therefore unsafe to cancel
the work, it must be flushed.
Topology could have more instances of the tokens being searched for than
the number of sets that need to be copied. Stop copying token after the
limit of number of token instances has been reached. This worked before
only by chance as we had allocated more size for the tuples array than
the number of actual tokens being parsed.
Fixes: 7006d20e5e9d ("ASoC: SOF: Introduce IPC3 ops") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512114630.24439-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
Kai Vehmanen [Fri, 12 May 2023 10:46:38 +0000 (13:46 +0300)]
ASoC: SOF: pm: save io region state in case of errors in resume
If there are failures in DSP runtime resume, the device state will not
reach active and this makes it impossible e.g. to retrieve a possible
DSP panic dump via "exception" debugfs node. If
CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in
cache is stale. If debugfs cache is not used, the region simply cannot
be read.
To allow debugging these scenarios, update the debugfs cache contents in
resume error handler. User-space can then later retrieve DSP panic and
other state via debugfs (requires SOF debugfs cache to be enabled in
build).
Reported-by: Curtis Malainey <cujomalainey@chromium.org Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Curtis Malainey <cujomalainey@chromium.org Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
ASoC: MAINTAINERS: drop Krzysztof Kozlowski from Samsung audio
Remove Krzysztof Kozlowski from maintainer of Samsung SoC Audio drivers
and change the status to maintenance (no one is reality being paid for
looking at this).
Douglas Anderson [Thu, 11 May 2023 16:25:12 +0000 (09:25 -0700)]
ASoC: mediatek: mt8186: Fix use-after-free in driver remove path
When devm runs function in the "remove" path for a device it runs them
in the reverse order. That means that if you have parts of your driver
that aren't using devm or are using "roll your own" devm w/
devm_add_action_or_reset() you need to keep that in mind.
The mt8186 audio driver didn't quite get this right. Specifically, in
mt8186_init_clock() it called mt8186_audsys_clk_register() and then
went on to call a bunch of other devm function. The caller of
mt8186_init_clock() used devm_add_action_or_reset() to call
mt8186_deinit_clock() but, because of the intervening devm functions,
the order was wrong.
Specifically at probe time, the order was:
1. mt8186_audsys_clk_register()
2. afe_priv->clk = devm_kcalloc(...)
3. afe_priv->clk[i] = devm_clk_get(...)
At remove time, the order (which should have been 3, 2, 1) was:
1. mt8186_audsys_clk_unregister()
3. Free all of afe_priv->clk[i]
2. Free afe_priv->clk
The above seemed to be causing a use-after-free. Luckily, it's easy to
fix this by simply using devm more correctly. Let's move the
devm_add_action_or_reset() to the right place. In addition to fixing
the use-after-free, code inspection shows that this fixes a leak
(missing call to mt8186_audsys_clk_unregister()) that would have
happened if any of the syscon_regmap_lookup_by_phandle() calls in
mt8186_init_clock() had failed.
Peter Ujfalusi [Fri, 12 May 2023 11:03:17 +0000 (14:03 +0300)]
ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_config
The commands in sof_ipc_dai_config.flags are encoded as bits:
1 (bit0) - hw_params
2 (bit1) - hw_free
4 (bit2) - pause
These are commands, they cannot be combined as one would assume, for
example
3 (bit0 | bit1) is invalid.
This can happen right at the second start of a stream as at the end of the
first stream we set the hw_free command (bit1) and on the second start we
would OR on top of it the hw_params (bit0).
Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Link: https://lore.kernel.org/r/20230512110317.5180-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions
When a firmware IPC error happens during a pm_runtime suspend, we
ignore the error and suspend anyways. However, the code
unconditionally increases the runtime_pm counter. This results in a
confusing configuration where the code will suspend, resume but never
suspend again due to the use of pm_runtime_get_noresume().
The intent of the counter increase was to prevent entry in D3, but if
that transition to D3 is already started it cannot be stopped. In
addition, there's no point in that case in trying to prevent anything,
the firmware error is handled and the next resume will re-initialize
the firmware completely.
This patch changes the logic to prevent suspend when the device is
pm_runtime active and has a use_count > 0.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
ASoC: SOF: Intel: hda-mlink: add helper to program SoundWire PCMSyCM registers
These registers enable the HDaudio DMA hardware to split/merge data
from different PDIs, possibly on different links.
This capability exists for all types of HDaudio extended links, but
for now is only required for SoundWire. In the SSP/DMIC case, the IP
is programmed by the DSP firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
ASoC: SOF: Intel: hda-mlink: initialize instance_offset member
We defined the values but never initialized it for SoundWire/SSP, fix
this miss.
A Fixes: tag is not provided as instance_offset was not used so far,
so nothing was really broken. This patch is only required for the
SoundWire support in the following patch.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
The base_ptr value needs to be derived from the remap_addr pointer,
not the ml_addr. This base_ptr was used only in debug logs that were
so far not contributed upstream so the issue was not detected. It
needs to be fixed for SoundWire support on LunarLake.
Fixes: 17c9b6ec35c0 ("ASoC: SOF: Intel: hda-mlink: add structures to parse ALT links") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
In hindsight it was a very bad idea to use the same refcount for
Extended and 'legacy' HDaudio multi-links. The existing solution only
powers-up the first sublink, which causes SoundWire and SSP tests to
fail when more than one DAI is used concurrently. Solving this problem
requires per-sublink refcounting, as suggested in this patch.
The existing refcounting remains for 'legacy' HdAudio links, mainly to
avoid changing the obscure programming sequence in
snd_hdac_ext_bus_link_put().
ALSA: hda/realtek: Apply HP B&O top speaker profile to Pavilion 15
The Pavilion 15 line has B&O top speakers similar to the x360 and
applying the same profile produces good sound. Without this, the
sound would be tinny and underpowered without either applying
model=alc295-hp-x360 or booting another OS first.
Signed-off-by: Ryan Underwood <nemesis@icequake.net> Fixes: 563785edfcef ("ALSA: hda/realtek - Add quirk entry for HP Pavilion 15") Link: https://lore.kernel.org/r/ZF0mpcMz3ezP9KQw@icequake.net Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 12 May 2023 07:58:58 +0000 (09:58 +0200)]
ALSA: usb-audio: Add a sample rate workaround for Line6 Pod Go
Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed
48k sample rate like other Line6 models. This patch adds the
corresponding entry to line6_parse_audio_format_rate_quirk().
Dan Carpenter [Tue, 9 May 2023 09:07:11 +0000 (12:07 +0300)]
ALSA: firewire-digi00x: prevent potential use after free
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.
Paweł Anikiel [Mon, 8 May 2023 11:30:37 +0000 (13:30 +0200)]
ASoC: ssm2602: Add workaround for playback distortions
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 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.
ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
These models use 2 CS35L41 amplifiers using SPI for down-facing
speakers.
alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the
down-facing speakers.
Pin configs are needed to enable headset mic detection.
Note that these models lack the ACPI _DSD properties needed to
initialize the amplifiers. They can be added during boot to get working
sound out of the speakers:
https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff
Martin Povišer [Tue, 9 May 2023 15:34:12 +0000 (17:34 +0200)]
ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
A bunch of TI's codecs have binding schemas which force #sound-dai-cells
to one despite those codecs only having a single DAI. Allow for bindings
with zero DAI cells and deprecate the former non-zero value.
Aidan MacDonald [Tue, 9 May 2023 12:51:34 +0000 (13:51 +0100)]
ASoC: jz4740-i2s: Make I2S divider calculations more robust
When the CPU supplies bit/frame clocks, the system clock (clk_i2s)
is divided to produce the bit clock. This is a simple 1/N divider
with a fairly limited range, so for a given system clock frequency
only a few sample rates can be produced. Usually a wider range of
sample rates is supported by varying the system clock frequency.
The old calculation method was not very robust and could easily
produce the wrong clock rate, especially with non-standard rates.
For example, if the system clock is 1.99x the target bit clock
rate, the divider would be calculated as 1 instead of the more
accurate 2.
Instead, use a more accurate method that considers two adjacent
divider settings and selects the one that produces the least error
versus the requested rate. If the error is 5% or higher then the
rate setting is rejected to prevent garbled audio.
Skip divider calculation when the codec is supplying both the bit
and frame clock; in that case, the divider outputs are unused and
we don't want to constrain the sample rate.
Maxim Kochetkov [Fri, 5 May 2023 06:28:20 +0000 (09:28 +0300)]
ASoC: dwc: limit the number of overrun messages
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.
gentwo / linux / .git / log