When configured for default synchronisation (Rx syncs to Tx) and the
SAI operates in consumer mode (clocks provided externally to Tx), a
synchronisation error occurs on Tx on the first attempt after device
initialisation when the playback stream is started while a capture
stream is already active. This results in channel shift/swap on the
playback stream.
Subsequent streams (ie after that first failing one) always work
correctly, no matter the order, with or without the other stream active.
This issue was observed (and fix tested) on an i.MX6UL board connected
to an ADAU1761 codec, where the codec provides both frame and bit clock
(connected to TX pins).
To fix this, always initialize the 'other' xCR4 and xCR5 registers when
we're starting a stream which is synced to the opposite one, irregardless
of the producer/consumer status.
Fixes: 51659ca069 ("ASoC: fsl-sai: set xCR4/xCR5/xMR for SAI master mode")
Signed-off-by: Maarten Zanders <maarten@zanders.be>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20251024135716.584265-1-maarten@zanders.be
Signed-off-by: Mark Brown <broonie@kernel.org>
In non-interleaved mode, several QMC channels are used in sync.
More details can be found in commit 188d9cae54 ("ASoC: fsl:
fsl_qmc_audio: Add support for non-interleaved mode.")
At the time being, an interrupt is requested on each channel to
perform capture/playback completion, allthough the completion is
really performed only once all channels have completed their work.
This leads to a lot more interrupts than really needed. Looking at
/proc/interrupts shows ~3800 interrupts per second when using
4 capture and 4 playback devices with 5ms periods while
only 1600 (200 x 4 + 200 x 4) periods are processed during one second.
The QMC channels work in sync, the one started first is the one
finishing first and the one started last is the one finishing last,
so when the last one finishes it is guaranteed that the other ones are
finished as well. Therefore only request completion processing on the
last QMC channel.
On my board with the above exemple, on a kernel started with
'threadirqs' option, the QMC irq thread uses 16% CPU time with this
patch while it uses 26% CPU time without this patch.
Acked-by: Herve Codina <herve.codina@bootlin.com>
Signed-off-by: Christophe Leroy <christophe.leroy@csgroup.eu>
Link: https://patch.msgid.link/bbd5167d190bbb45c3a4cd6ef2dece8817e0cc1e.1758209158.git.christophe.leroy@csgroup.eu
Signed-off-by: Mark Brown <broonie@kernel.org>
If a defer probe happens for the cpu device, the return value for
cpu_pdev "ret = -EINVAL" breaks defer probe. As the cpus->dai_name
can be replaced by cpus->of_node, to simplify the code, remove related
code for cpu_pdev.
Fixes: 6a5f850aa8 ("ASoC: fsl: Add imx-hdmi machine driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Message-ID: <20250820031641.2158261-1-shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>:
On some Qualcomm platforms such as QCS6490-RB3Gen2, the multiple
WSA8830/WSA8835 speakers share a common reset (shutdown) GPIO.
To handle such cases, use the reset controller framework along with the
"reset-gpio" driver.
Tested on:
- QCS6490-RB3Gen2
Use the regmap_write() for software reset in fsl_sai_config_disable would
cause the FSL_SAI_CSR_BCE bit to be cleared. Refer to
commit 197c53c8ec ("ASoC: fsl_sai: Don't disable bitclock for i.MX8MP")
FSL_SAI_CSR_BCE should not be cleared. So need to use regmap_update_bits()
instead of regmap_write() for these bit operations.
Fixes: dc78f7e591 ("ASoC: fsl_sai: Force a software reset when starting in consumer mode")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250807020318.2143219-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
There is two different cases for getting channel status data:
1. With PHY exists, there is firmware running on M core, the firmware
should fill the channel status to RAM space, driver need to read them
from RAM.
2. Without PHY, the channel status need to be obtained from registers.
For the XCVR module on i.MX95, even though it only supports SPDIF, the
channel status needs to be obtained from RAM space, which is processed
by firmware. Firmware is necessary to trigger the FSL_XCVR_IRQ_NEW_CS
interrupt.
This change also applies for the SPDIF & ARC function on i.MX8MP which
has the firmware.
Fixes: e6a9750a34 ("ASoC: fsl_xcvr: Add suspend and resume support")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250710030405.3370671-3-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peng Fan <peng.fan@nxp.com>:
This patchset is a pick up of patch 1,2 from [1]. And I also collect
Linus's R-b for patch 2. After this patchset, there is only one user of
of_gpio.h left in sound driver(pxa2xx-ac97).
of_gpio.h is deprecated, update the driver to use GPIO descriptors.
Patch 1 is to drop legacy platform data which in-tree no users are using it
Patch 2 is to convert to GPIO descriptors
Checking the DTS that use the device, all are using GPIOD_ACTIVE_LOW
polarity for reset-gpios, so all should work as expected with this patch.
[1] https://lore.kernel.org/all/20250408-asoc-gpio-v1-0-c0db9d3fd6e9@nxp.com/
On an imx8mm platform with an external clock provider, when running the
receiver (arecord) and triggering an xrun with xrun_injection, we see a
channel swap/offset. This happens sometimes when running only the
receiver, but occurs reliably if a transmitter (aplay) is also
concurrently running.
It seems that the SAI loses track of frame sync during the trigger stop
-> trigger start cycle that occurs during an xrun. Doing just a FIFO
reset in this case does not suffice, and only a software reset seems to
get it back on track.
This looks like the same h/w bug that is already handled for the
producer case, so we now do the reset unconditionally on config disable.
Signed-off-by: Arun Raghavan <arun@asymptotic.io>
Reported-by: Pieterjan Camerlynck <p.camerlynck@televic.com>
Fixes: 3e3f8bd569 ("ASoC: fsl_sai: fix no frame clk in master mode")
Cc: stable@vger.kernel.org
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://patch.msgid.link/20250626130858.163825-1-arun@arunraghavan.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Current Kconfig menu at [ALSA for SoC audio support] has no rules.
So, some venders are using menu style, some venders are listed each drivers
on top page, etc. It is difficult to find target vender and/or drivers
because it is very random.
Let's standardize ASoC menu, like below
--- ALSA for SoC audio support
Analog Devices --->
AMD --->
Apple --->
Atmel --->
Au1x ----
Broadcom --->
Cirrus Logic --->
DesignWare --->
Freescale --->
Google --->
Hisilicon --->
...
One concern is *vender folder* alphabetical order vs *vender name*
alphabetical order were different. For example "sunxi" menu is
"Allwinner".
Link: https://lore.kernel.org/r/8734c8bf3l.wl-kuninori.morimoto.gx@renesas.com
If both playback and capture streams have large buffer size in low power
audio case, the total size will exceed the maximum buffer size for this
sound card.
Capture stream doesn't need so large buffer size in fact. So calculate
a reasonable smaller buffer size and allocate it for capture stream.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://patch.msgid.link/20250421105701.2273588-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
imx_card_parse_of() allocs 2 components for CPU/Platform (A)
static int imx_card_parse_of(...)
{
...
for_each_child_of_node(...) {
dlc = devm_kzalloc(...);
...
link->cpus = &dlc[0];
(A) link->platforms = &dlc[1];
}
...
}
The link might be used as DPCM backend, in such case, link->plaforms
will be not used. The driver overwrite it as Dummy DAI (B).
} else if (!strncmp(link->name, "HiFi-ASRC-BE", 12)) {
/* DPCM backend */
link->no_pcm = 1;
link->platforms->of_node = NULL;
(B) link->platforms->name = "snd-soc-dummy";
}
If it was not used for generic DMAEngine, we can just remove it.
By this patch, created dlc (A) will be just wasted, but it won't leak.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87cydfr1z6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On SNDRV_PCM_TRIGGER_START event, audio data pointers are not reset.
This leads to wrong data buffer usage when multiple TRIGGER_START are
received and ends to incorrect buffer usage between the user-space and
the driver. Indeed, the driver can read data that are not already set by
the user-space or the user-space and the driver are writing and reading
the same area.
Fix that resetting data pointers on each SNDRV_PCM_TRIGGER_START events.
Fixes: 075c7125b1 ("ASoC: fsl: Add support for QMC audio")
Cc: stable@vger.kernel.org
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20250410091643.535627-1-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With DPCM case, for example ASRC->SAI->AMIX, the SAI can be codec dai
device in backend, but __soc_pcm_hw_params() will get the tdm_mask
for channel constraint, tdm_mask is set by snd_soc_dai_set_tdm_slot()
from slot number, but SAI supports flexible channel number with fixed slot
number, so add an empty xlate_tdm_slot_mask() callback to avoid the
channel constraint in __soc_pcm_hw_params().
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250328085744.1893434-3-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If clk_id is zero, it means FSL_SAI_CLK_BUS in fsl_sai_set_dai_sysclk(),
as the clk[FSL_SAI_CLK_BUS]'s rate can't be changed, there is no rate
changed for mclk.
But with audio-graph-card, the clk_id is always zero, in order to allow
to set mclk rate with zero clk_id, update the condition to be if clk_id is
zero, then set the FSL_SAI_CLK_MAST1's rate. This would not change the
original function for master mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250328085744.1893434-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
timer_delete[_sync]() replaces del_timer[_sync](). Convert the whole tree
over and remove the historical wrapper inlines.
Conversion was done with coccinelle plus manual fixups where necessary.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Ingo Molnar <mingo@kernel.org>
Pull sound fixes from Takashi Iwai:
"A collection of device-specific fixes that have been gathered since
the previous pull:
- A few more HD-audio quirks and fixups
- A series of Qualcomm AudioReach fixes
- Various small fixes for ASoC rt5665, WSA, SOF and Cirrus"
* tag 'sound-fix-6.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Fix built-in mic on another ASUS VivoBook model
ALSA: hda/realtek - Support mute led function for HP platform
ASoC: imx-card: Add NULL check in imx_card_probe()
ASoC: codecs: rt5665: Fix some error handling paths in rt5665_probe()
ASoC: q6apm-dai: make use of q6apm_get_hw_pointer
ASoC: qdsp6: q6apm-dai: fix capture pipeline overruns.
ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment.
ASoC: q6apm: add q6apm_get_hw_pointer helper
ASoC: q6apm-dai: schedule all available frames to avoid dsp under-runs
ASoC: SOF: hda/ptl: Move mic privacy change notification sending to a work
ALSA/hda: intel-sdw-acpi: Remove (explicitly) unused header
ALSA: hda/realtek: Enable Mute LED on HP OMEN 16 Laptop xd000xx
ALSA: hda/tas2781: Upgrade calibratd-data writing code to support Alpha and Beta dsp firmware
ASoC: qdsp6: q6asm-dai: fix q6asm_dai_compr_set_params error path
ALSA: hda/realtek: Fix built-in mic breakage on ASUS VivoBook X515JA
ASoC: sma1307: Fix error handling in sma1307_setting_loaded()
ASoC: codecs: wsa884x: Correct VI sense channel mask
ASoC: codecs: wsa883x: Correct VI sense channel mask
firmware: cs_dsp: Ensure cs_dsp_load[_coeff]() returns 0 on success
Pull sound updates from Takashi Iwai:
"We've received lots of commits at this time, as a result of various
cleanup and refactoring works as well as a few new drivers and the
generic SoundWire support. Most of changes are device-specific, little
about the core changes. Some highlights below:
Core:
- A couple of (rather minor) race fixes in ALSA sequencer code
- A regression fix in ALSA timer code that may lead to a deadlock
ASoC:
- A large series of code conversion to use modern terminology for the
clocking configuration
- Conversions of PM ops with the modern macros in all ASoC drivers
- Clarification of the control operations
- Prepartory work for more generic SoundWire SCDA controls
- Support for AMD ACP 7.x, AWINC WM88166, Everest ES8388, Intel AVS
PEAKVOL and GAIN DSP modules Mediatek MT8188 DMIC, NXP i.MX95,
nVidia Tegra interconnects, Rockchip RK3588 S/PDIF, Texas
Instruments SN012776 and TAS5770L, and Wolfson WM8904 DMICs
Others:
- Conversions of PM ops with the modern macros in the rest drivers
- USB-audio quirks and fixes for Presonus Studio, DJM-A9, CME
- HD-audio quirks and fixes ASUS, HP, Lenovo, and others"
* tag 'sound-6.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (651 commits)
ALSA: hda: tas2781-i2c: Remove unnecessary NULL check before release_firmware()
ALSA: hda: cs35l56: Remove unnecessary NULL check before release_firmware()
ALSA: hda/realtek: Bass speaker fixup for ASUS UM5606KA
ALSA: hda/realtek: Fix built-in mic assignment on ASUS VivoBook X515UA
ALSA: hda/realtek: Add support for various HP Laptops using CS35L41 HDA
ALSA: timer: Don't take register_mutex with copy_from/to_user()
ASoC: SDCA: Correct handling of selected mode DisCo property
ASoC: amd: yc: update quirk data for new Lenovo model
ALSA: hda/realtek: fix micmute LEDs on HP Laptops with ALC3247
ALSA: hda/realtek: fix micmute LEDs on HP Laptops with ALC3315
ASoC: SOF: mediatek: Commonize duplicated functions
ASoC: dmic: Fix NULL pointer dereference
ASoC: wm8904: add DMIC support
ASoC: wm8904: get platform data from DT
ASoC: dt-bindings: wm8904: Add DMIC, GPIO, MIC and EQ support
ASoC: wm8904: Don't touch GPIO configs set to 0xFFFF
of: Add of_property_read_u16_index
ALSA: oxygen: Fix dependency on CONFIG_PM_SLEEP
ASoC: ops: Apply platform_max after deciding control type
ASoC: ops: Remove some unnecessary local variables
...
Pull timer cleanups from Thomas Gleixner:
"A treewide hrtimer timer cleanup
hrtimers are initialized with hrtimer_init() and a subsequent store to
the callback pointer. This turned out to be suboptimal for the
upcoming Rust integration and is obviously a silly implementation to
begin with.
This cleanup replaces the hrtimer_init(T); T->function = cb; sequence
with hrtimer_setup(T, cb);
The conversion was done with Coccinelle and a few manual fixups.
Once the conversion has completely landed in mainline, hrtimer_init()
will be removed and the hrtimer::function becomes a private member"
* tag 'timers-cleanups-2025-03-23' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (100 commits)
wifi: rt2x00: Switch to use hrtimer_update_function()
io_uring: Use helper function hrtimer_update_function()
serial: xilinx_uartps: Use helper function hrtimer_update_function()
ASoC: fsl: imx-pcm-fiq: Switch to use hrtimer_setup()
RDMA: Switch to use hrtimer_setup()
virtio: mem: Switch to use hrtimer_setup()
drm/vmwgfx: Switch to use hrtimer_setup()
drm/xe/oa: Switch to use hrtimer_setup()
drm/vkms: Switch to use hrtimer_setup()
drm/msm: Switch to use hrtimer_setup()
drm/i915/request: Switch to use hrtimer_setup()
drm/i915/uncore: Switch to use hrtimer_setup()
drm/i915/pmu: Switch to use hrtimer_setup()
drm/i915/perf: Switch to use hrtimer_setup()
drm/i915/gvt: Switch to use hrtimer_setup()
drm/i915/huc: Switch to use hrtimer_setup()
drm/amdgpu: Switch to use hrtimer_setup()
stm class: heartbeat: Switch to use hrtimer_setup()
i2c: Switch to use hrtimer_setup()
iio: Switch to use hrtimer_setup()
...
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
Change 'dais' property to be optional, that fsl_audmix device can be
linked with SAI device by audio graph card.
