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A fairly quiet release from a core and framework point of view, but a
very big one from the point of view of new drivers:
- More refectoring from Morimoto-san, this time mainly around DAI
links and how we control the ordering of trigger() callbacks.
- Convert a lot of drivers to use maple tree based caches.
- Lots of work on the x86 driver stack.
- Compressed audio support for Qualcomm.
- Support for AMD SoundWire, Analog Devices SSM3515, Google Chameleon,
Ingenic X1000, Intel systems with various CODECs, Longsoon platforms,
Maxim MAX98388, Mediatek MT8188, Nuvoton NAU8825C, NXP platforms with
NAU8822, Qualcomm WSA884x, StarFive JH7110, Texas Instruments TAS2781.
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Merge tag 'asoc-v6.5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.5
A fairly quiet release from a core and framework point of view, but a
very big one from the point of view of new drivers:
- More refectoring from Morimoto-san, this time mainly around DAI
links and how we control the ordering of trigger() callbacks.
- Convert a lot of drivers to use maple tree based caches.
- Lots of work on the x86 driver stack.
- Compressed audio support for Qualcomm.
- Support for AMD SoundWire, Analog Devices SSM3515, Google Chameleon,
Ingenic X1000, Intel systems with various CODECs, Longsoon platforms,
Maxim MAX98388, Mediatek MT8188, Nuvoton NAU8825C, NXP platforms with
NAU8822, Qualcomm WSA884x, StarFive JH7110, Texas Instruments TAS2781.
On HP EliteBook 835/845/845W G10, the audio LEDs can be enabled by
ALC285_FIXUP_HP_MUTE_LED. So use it accordingly.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Fixes: 3e10f6ca76c4 ("ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops")
Link: https://lore.kernel.org/r/20230626130301.301712-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds the required quirk to enable the Cirrus amp and correct pins
on the ASUS ROG GV601V series.
While this works if the related _DSD properties are made available, these
aren't included in the ACPI of these laptops (yet).
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Link: https://lore.kernel.org/r/20230621085715.5382-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds the required quirk to enable the Cirrus amp and correct pins
on the ASUS ROG G634Z series.
While this works if the related _DSD properties are made available, these
aren't included in the ACPI of these laptops (yet).
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Link: https://lore.kernel.org/r/20230619060320.1336455-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
smatch error:
sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error:
we previously assumed 'rac97' could be null (see line 2072)
remove redundant assignment, return error if rac97 is NULL.
Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*")
Signed-off-by: Su Hui <suhui@nfschina.com>
Link: https://lore.kernel.org/r/20230615021732.1972194-1-suhui@nfschina.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The file is called spdif-in, but we abused it to show only sample rates
from various sources. Rectify it as far as possible (the FPGA doesn't
give us a lot of information).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-10-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is only a very partial fix - the frequency-dependent envelope & LFO
register values aren't adjusted.
But I'm not sure they were even correct at 48 kHz to start with, as most
of them are precalculated by common code which assumes an EMU8K-specific
44.1 kHz word clock, and it seems somewhat unlikely that the hardware's
register interpretation was adjusted to compensate for the different
word clock.
In any case I'm not going to spend time on fixing that, as this code is
unlikely to be actually used by anyone today.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we know the actual word clock, we can:
- Put the resulting rate into the hardware info
- At 44.1 kHz word clock shift the rate for the pitch calculations,
which presume a 48 kHz word clock
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value isn't used yet; the subsequent commits will do that.
This ignores the existence of rates above 48 kHz, which is fine, as the
hardware will just switch to the fallback clock source when fed with a
rate which is incompatible with the base clock multiplier, which
currently is always x1.
The sample rate display in /proc spdif-in is adjusted to reflect our
understanding of the input rates.
This is tested only with an 0404b card without sync card, so there is a
lot of room for improvement.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The actually available clock sources depend on the available audio input
ports and dedicated clock input ports.
This includes refactoring the code to be data-driven to remain
manageable.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, we set the fallback as a side effect of setting the source. But
the fallback makes no sense at all when an internal clock is selected.
Defaulting to 48k for S/PDIF & ADAT makes sense, but as that is the
global default and we're not changing it automatically any more, it's
just fine to leave it entirely to the explicit setting.
This changes the name of the pre-existing control to something more
appropriate (regardless of the split), so users will need to adjust
their mixer settings.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On loongson controller, the value of WALLCLK register
is always 0, which is meaningless, so we return directly.
Signed-off-by: Yanteng Si <siyanteng@loongson.cn>
Signed-off-by: Yingkun Meng <mengyingkun@loongson.cn>
Acked-by: Huacai Chen <chenhuacai@loongson.cn>
Link: https://lore.kernel.org/r/185df71ef413ab190460eb377703214ee7288aeb.1686128807.git.siyanteng@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On loongson controller, after calling snd_hdac_stream_updateb()
to enable DMA engine, the SDnCTL.STRM will become to zero. We
need to access SDnCTL in dword to keep SDnCTL.STRM is not changed.
Signed-off-by: Yanteng Si <siyanteng@loongson.cn>
Signed-off-by: Yingkun Meng <mengyingkun@loongson.cn>
Acked-by: Huacai Chen <chenhuacai@loongson.cn>
Link: https://lore.kernel.org/r/27aeddf5ebbe7c69631cec0e489c1b264be94990.1686128807.git.siyanteng@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On loongson controller, RIRBSTS.RINTFL cannot be cleared,
azx_interrupt() is called all the time. We disable RIRB
interrupt, and use polling mode by default.
Signed-off-by: Yanteng Si <siyanteng@loongson.cn>
Signed-off-by: Yingkun Meng <mengyingkun@loongson.cn>
Acked-by: Huacai Chen <chenhuacai@loongson.cn>
Link: https://lore.kernel.org/r/d309a75424d438b958d90d797b4f1ba45468e090.1686128807.git.siyanteng@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Ensure Firmware Load control and Firmware Type control
returns 1 when the value changes.
Remove fw_mutex from firmware load control put, since it is
unnecessary, and prevents any possibility of mutex inversion.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230606103436.455348-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
The new xarray lookup code requires to know complete kcontrol->id before
snd_ctl_add() call. Reorder the code to make the initialization properly.
Cc: stable@kernel.org # v5.19+
Reported-by: Martin Zidek <zidek@master.cz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606073122.597491-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
- Include the FX bus map, without which the already present send routing
info would require looking up the documentation.
- Include the physical I/O channels as known to the driver
- Make the multi-channel capture map actually name the mapped input
channels rather than "FXBUS" (Audigy) or even "???" (SbLive)
- The latter two are omitted for E-MU cards, as their physical I/O is
routed through the FPGA
- While at it, make the "Card" field somewhat more useful
This includes de-duplicating the label tables between emuproc and emufx,
updating/improving the FX bus label table, and making the SB Live! 5.1
multi-track capture channel mapping hack data-driven.
Tested-by: Jonathan Dowland <jon@dow.land>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include the routing information, which can be actually read back.
Somewhat as a drive-by, make the register dump format less obscure - the
previous one made no sense at all.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems to make little sense to include the FX send routing, but not
the amounts.
This also simplifies the code somewhat, and lines up the output.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The limits were appropriate only for the 2nd set.
FWIW, the channel count 4 for the 2nd set is suspicious as well - at
least P17V_PLAYBACK_FIFO_PTR actually has 8 channels, and comments on
HCFG2 hint at that as well. But all bitmasks are documented only for 4
channels. Anyway, rectifying that is out of scope for this patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 2nd register set belongs to the P16V chip (or embedded P17V module),
so there is nothing to show when no such part is present. Gen2 E-MU
cards have a P17V, but it's entirely unused, so we hide it there as
well.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After commit b8a1a4cd5a98 ("i2c: Provide a temporary .probe_new()
call-back type"), all drivers being converted to .probe_new() and then
03c835f498b5 ("i2c: Switch .probe() to not take an id parameter") convert
back to (the new) .probe() to be able to eventually drop .probe_new() from
struct i2c_driver.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20230525203640.677826-1-u.kleine-koenig@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series handles a few issues related to the ES8316 audio
codec, discovered while doing some testing on the Rock 5B board.
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>
On SB cards the number of captured channels is derived from the voice
mask mixer control. But for E-MU cards this wasn't actually "wired up",
so changing the mask would simply mess up the recording.
We could fix that, but the channel routing through the FPGA makes the
masking redundant. So instead we hide the control, and let the user
specify the PCM channel count the traditional way.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware can deal with primes up to 7 and power-of-two multiples
thereof; the limitation is reflected by the possible buffer sizes.
Note that setting the voice mask will not allow more than 16 channels
even on Sound Blaster Audigy anymore, as 32 seems a bit excessive (the
code overall appears to think so, just not in this case).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to specify that the hardware supports non-standard rates, as
otherwise the sound core creates a constraint which limits the rate to
the specified standard rates. That also made the rate constraint we were
already adding meaningless.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer size register sets the size of the whole buffer, not just one
period. We actually handled it like that, except that the constraint was
set on the wrong parameter. The period size is implicitly constrained by
the buffer size and the fixed period count of 2.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We use independent voices for the channels, so we need to make an effort
to ensure that they are actually in sync.
The hardware doesn't provide atomicity, so we may need to retry a few
times, due to NMIs, PCI contention, and the wrong phase of the moon.
Solution inspired by kX-project.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236023-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For unclear reasons, the extra voice was set up with half the buffer
size instead of the period size. Commit 27ae958cf6 ("emu10k1 driver -
add multichannel device hw:x,3 [2-8/8]") mentions half-loop interrupts,
so maybe this was an artifact of an earlier iteration of the patch.
While at it, also fix periods_min of the regular playback - one period
makes just no sense.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236023-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not include pm_runtime.h header in files where APIs exported by
pm_runtime.h are not used.
Signed-off-by: Claudiu Beznea <claudiu.beznea@microchip.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> # for omap-mcbsp-st.c
Link: https://lore.kernel.org/r/20230517094903.2895238-2-claudiu.beznea@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
