ASoC: Updates for v6.14
This was quite a quiet release for what I imagine are holiday related
reasons, the diffstat is dominated by some Cirrus Logic Kunit tests.
There's the usual mix of small improvements and fixes, plus a few new
drivers and features. The diffstat includes some DRM changes due to
work on HDMI audio.
- Allow clocking on each DAI in an audio graph card to be configured
separately.
- Improved power management for Renesas RZ-SSI.
- KUnit testing for the Cirrus DSP framework.
- Memory to meory operation support for Freescale/NXP platforms.
- Support for pause operations in SOF.
- Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek
ALC5682I-VE
Clang warns (or errors with CONFIG_WERROR=y):
sound/pci/hda/tas2781_hda_spi.c:110:6: error: variable 'ret' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/pci/hda/tas2781_hda_spi.c:119:9: note: uninitialized use occurs here
119 | return ret;
| ^~~
sound/pci/hda/tas2781_hda_spi.c:110:2: note: remove the 'if' if its condition is always true
110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/pci/hda/tas2781_hda_spi.c:108:9: note: initialize the variable 'ret' to silence this warning
108 | int ret;
| ^
| = 0
Sink the declaration of ret into the if block and just return 0 at the
end of the function, as there is nothing to do if cur_book has already
been changed.
Fixes: bb5f86ea50 ("ALSA: hda/tas2781: Add tas2781 hda SPI driver")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202501192006.Hm9GmKiV-lkp@intel.com/
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20250120-tas2781_hda_spi-fix-wsometimes-uninitialized-v1-1-d7fd104aa63e@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new device_setup option (SCARLETT2_USE_FCP_DRIVER = 0x08) that
allows users to opt in to using the new FCP driver instead of the
existing scarlett2 driver for their device. This provides a way to
test the new FCP driver on existing supported hardware while keeping
the Scarlett2 driver as the default.
When the SCARLETT2_USE_FCP_DRIVER bit is set in device_setup, the
scarlett2 driver initialisation will hand off to the FCP driver
instead of proceeding with its own initialisation. The FCP driver then
provides access to the device via its hwdep interface.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/94ffd7971d73cb0cbea6933b28f7528ce5b9edde.1737048528.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new kernel driver for the Focusrite Control Protocol (FCP),
which is used by Focusrite Scarlett 2nd Gen, 3rd Gen, 4th Gen, Clarett
USB, Clarett+, and Vocaster series audio interfaces. This driver
provides a user-space control interface via ALSA's hwdep subsystem.
Unlike the existing Scarlett2 driver which implements all ALSA
controls in kernel space, this new FCP driver takes a different
approach by providing a minimal kernel interface that allows a
user-space driver to send FCP commands and receive notifications. The
only control implemented in kernel space is the Level Meter, since it
requires frequent polling of volatile data.
While this driver supports all interfaces that the Scarlett2 driver
works with, it is initially enabled only for 4th Gen 16i16, 18i16,
and 18i20 interfaces that are not supported by the Scarlett2 driver.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/597741a9b1198b965561547511d3d345f91cba20.1737048528.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch was used to add TAS2781 devices on SPI support in sound/pci/hda.
It use ACPI node descript about parameters of TAS2781 on SPI, it like:
Scope (_SB.PC00.SPI0)
{
Device (GSPK)
{
Name (_HID, "TXNW2781") // _HID: Hardware ID
Method (_CRS, 0, NotSerialized)
{
Name (RBUF, ResourceTemplate ()
{
SpiSerialBusV2 (...)
SpiSerialBusV2 (...)
}
}
}
}
And in platform/x86/serial-multi-instantiate.c, those spi devices will be
added into system as a single SPI device, so TAS2781 SPI driver will
probe twice for every single SPI device. And driver will also parser
mono DSP firmware binary and RCA binary for itself.
The code support Realtek as the primary codec.
In patch version-10, add multi devices firmware binary support,
to compatble with windows driver, they can share same firmware binary.
Signed-off-by: Baojun Xu <baojun.xu@ti.com>
Link: https://patch.msgid.link/20241216122008.15425-1-baojun.xu@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codec driver can implement .hw_params and/or .prepare from struct
snd_soc_dai_ops. For codec-to-codec links only the former (.hw_params)
callback has been called.
On platforms like Amlogic Meson8/8b/8m2 the SoC's sound card
(sound/soc/meson/gx-card.c) uses a codec-to-codec link for the HDMI
codec output because further digital routing is required after the
backend. The new DRM HDMI (audio) codec framework (which internally
uses sound/soc/codecs/hdmi-codec.c) relies on the .prepare callback
of the hdmi-codec to be called. Implement calls to
snd_soc_dai_prepare() so the .prepare callback is called. In this
case the mandatory part is the call to prepare the sink (which is
the hdmi-codec on those platforms). Also call snd_soc_dai_prepare()
for the source to stay consistent with the rest of the code (even
though it's not strictly necessary to make the DRM HDMI codec
framework work on Amlogic Meson8/8b/8m2).
For platforms or sound cards without a codec-to-codec link with
additional parameters (which applies to most hardware) this changes
nothing as the .prepare callback is already called via
snd_pcm_do_prepare() (as well as dpcm_fe_dai_prepare() and
dpcm_be_dai_prepare()) on those.
Suggested-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com>
Link: https://patch.msgid.link/20250114215617.336105-3-martin.blumenstingl@googlemail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit cb18cd2603 ("ASoC: soc-core: do rtd->id trick at
snd_soc_add_pcm_runtime()") the ID stored in the PCM runtime data can
no longer be safely used to index the priv->dai_props array. This is
because the ID may be modified during snd_soc_add_pcm_runtime(), thus
resulting in an ID that's no longer a valid array index.
To fix this, use the position of the dai_link stored inside the PCM
runtime data relative to the start of the dai_link array as index into
the priv->dai_props array.
Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20250114184314.3583-2-laurentiumihalcea111@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As per specification, SDxLVI shall be at least 1 i.e.: two chunks to
perform a valid transfer. This is true for the PCM transfer code but
not firmware-transfer one.
Technical background:
- the LVI > 0 rule shall be obeyed in PCM transfer
- HW permits LVI == 0 when transfer is SW-controlled (SPIB)
- FW download is not a PCM transfer and is SW-controlled (SPIB)
The above is the fundament which AudioDSP firmware loading functions
have been built upon and worked since 2016. The presented changes are to
align the loading flows and avoid rising more questions in the future.
Achieve the goal by splitting snd_hdac_stream_setup_periods() into
substream-dependent and -independent part. Let snd_hdac_dsp_prepare()
utilize the latter so that both DSP-loading and PCM flows utilize same
BLDE setup loop which already takes care of cutting the buffer based on
azx_dev->period_bytes.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250114184239.120002-1-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
On i.MX943, the FIFO data address is changed and the bit width
of CICOSR is changed.
Add a new compatible string and update driver for these changes.
In the original change, rawmidi_info.tied_device showed -1 for the
unknown or untied device. But this would require the user-space to
check the protocol version and judge the value conditionally, which
is rather error-prone.
Instead, set the tied_device = 0 to be default as unknown, and
indicate the real device with the offset 1, for achieving more
backward compatibility.
Suggested-by: Jaroslav Kysela <perex@perex.cz>
Link: https://patch.msgid.link/20250114104711.19197-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On i.MX943, the FIFO data address is changed to 0x20~0x3c,
compared to previous version, there is a minus 4 offset,
so add a new regmap configuration for it.
And the bit width of CICOSR is changed to 5 bits, from bit
16th to 20th in REG_MICFIL_CTRL2 register, so update its
definition header file.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250114102720.3664667-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Current clock provider/consumer setting is set by dai_link->dai_fmt, and it
is Codec base on Sound Card driver (= SND_SOC_DAIFMT_CBx_CFx).
Current CPU/Codec drivers are already based on its own base
(= SND_SOC_DAIFMT_Bx_Fx). So, Codec clock setting uses dai_link->dai_fmt
as-is, and CPU side clock setting is created from Codec base setting by
flipping. Because of this, we can't set both CPU/Codec clock consumer for
example.
To solve this issue, this patch-set adds new ext_fmt on each DAI.
It can keep compatible with legacy style.
1. SND_SOC_DAIFMT_FORMAT_MASK
2. SND_SOC_DAIFMT_CLOCK
3. SND_SOC_DAIFMT_INV
4. SND_SOC_DAIFMT_CLOCK_PROVIDER
dai_fmt : dai_link->dai_fmt = common settings
ext_fmt : each DAI settings
Legacy
dai_fmt includes 1, 2, 3, 4
New style
dai_fmt includes 1, 2, 3
ext_fmt includes 4
Audio-Graph-Card2 will use this new style by this patch-set.
By this patch, Card2 default behavior (= no "clock-master / frame-master"
settings on DT) will be changed, but no drivers are using it.
In case of no DAI has "clock-master / frame-master" property on DT,
it will be...
Legacy
CPU : provider (because flipped from Codec)
Codec: consumer
New style
CPU : consumer
Codec: consumer
One note is that Simple-Card, Audio-Graph-Card don't implement
this new style to keep compatiblily.
In Overlay case, port order can be random, so we shouldn't use get_next()
function to get next port.
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
There are two MQS instances on the i.MX943 platform.
The definition of bit positions in the control register are
different. In order to support these MQS modules, define
two compatible strings to distinguish them.
Current ASoC is using dai_link->dai_fmt to set DAI format for both
CPU/Codec. But because it is using same settings, and
SND_SOC_DAIFMT_CLOCK_PROVIDER is flipped for CPU, we can't set both
CPU/Codec as clock consumer, for example.
To solve this issue, this patch uses extra format for each DAI which can
keep compatibility with legacy system,
1. SND_SOC_DAIFMT_FORMAT_MASK
2. SND_SOC_DAIFMT_CLOCK
3. SND_SOC_DAIFMT_INV
4. SND_SOC_DAIFMT_CLOCK_PROVIDER
Legacy
dai_fmt includes 1, 2, 3, 4
New idea
dai_fmt includes 1, 2, 3
ext_fmt includes 4
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Stephen Gordon <gordoste@iinet.net.au>
Link: https://patch.msgid.link/87pll0o5j6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is using dai_link->dai_fmt to set DAI format for both
CPU/Codec. But because it is using same settings, and
SND_SOC_DAIFMT_CLOCK_PROVIDER is flipped for CPU, we can't set both
CPU/Codec as clock consumer, for example.
To solve this issue, this patch enable to use extra format for each
DAI which can keep compatibility with legacy system,
1. SND_SOC_DAIFMT_FORMAT_MASK
2. SND_SOC_DAIFMT_CLOCK
3. SND_SOC_DAIFMT_INV
4. SND_SOC_DAIFMT_CLOCK_PROVIDER
Legacy
dai_fmt includes 1, 2, 3, 4
New idea
dai_fmt includes 1, 2, 3
ext_fmt includes 4
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Stephen Gordon <gordoste@iinet.net.au>
Link: https://patch.msgid.link/87r05go5ja.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no audio output if Speaker volume is set above 25.
According to datasheet Rev 2.5 maximum allowed value for the
Speaker output is 0b11001 (25)
0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2:
Left/Right Channel Class-D Driver Gain For DAC Left/Right
Input
(Step size is 1dB.)
00000 = Mute (DEFAULT)
00001 = 0dB
00002 = 1dB
...
11000 = 23dB
11001 = 24dB
So adjust this value in accordance with the datasheet.
Signed-off-by: Maxim Kochetkov <fido_max@inbox.ru>
Link: https://patch.msgid.link/20241219042227.7075-1-fido_max@inbox.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two MQS instances on the i.MX943 platform.
The definition of bit positions in the control register are
different. In order to support these MQS modules, define
two compatible strings to distinguish them.
As one instance is in the always-on domain, another is in the
wakeup domain, so the compatible strings are
"fsl,imx943-aonmix-mqs", "fsl,imx943-wakeupmix-mqs".
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://patch.msgid.link/20250113090321.3193464-3-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes the volume adjustment issue on the Lenovo ThinkBook 16P Gen5
by applying the necessary quirk configuration for the Realtek ALC287 codec.
The issue was caused by incorrect configuration in the driver,
which prevented proper volume control on certain systems.
Signed-off-by: Yage Geng <icoderdev@gmail.com>
Link: https://patch.msgid.link/20250113085208.15351-1-icoderdev@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we notify the sequencer client and port changes upon UMP FB
changes, but those aren't really corresponding to the UMP updates.
e.g. when a FB info gets updated, it's not notified but done only when
some of sequencer port attribute is changed. This is no ideal
behavior.
This patch adds the two new sequencer event types for notifying the
UMP EP and FB changes via the announce port. The new event takes
snd_seq_ev_ump_notify type data, which is compatible with
snd_seq_addr (where the port number is replaced with the block
number).
The events are sent when the EP and FB info gets updated explicitly
via ioctl, or the backend UMP receives the corresponding UMP
messages.
The sequencer protocol version is bumped to 1.0.5 along with it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250110155943.31578-9-tiwai@suse.de
