This patch adds support for the CS35L41 DSP.
The DSP allows for extra features, such as running
speaker protection algorithms and hibernations.
To utilize these features, the driver must load
firmware into the DSP, as well as various tuning
files which allow for customization for specific
models.
[ Slightly simplified Kconfig changes by tiwai ]
Signed-off-by: Vitaly Rodionov <vitaly.rodionov@cirrus.com>
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220630002335.366545-5-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs35l41 part contains a DSP which is able to run firmware.
The cs_dsp library can be used to control the DSP.
These controls can be exposed to userspace using ALSA controls.
This library adds apis to be able to interface between
cs_dsp and hda drivers and expose the relevant controls as
ALSA controls.
[ Note: the dependency of CONFIG_SND_HDA_CS_DSP_CONTROLS Kconfig is
corrected. Also, this Kconfig isn't enabled now but will be
actually enabled in a later patch -- tiwai ]
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220630002335.366545-2-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use atomic_try_cmpxchg instead of atomic_cmpxchg (*ptr, old, new) == old in
ep_state_update. x86 CMPXCHG instruction returns success in ZF flag,
so this change saves a compare after cmpxchg (and related move instruction
in front of cmpxchg).
No functional change intended.
Signed-off-by: Uros Bizjak <ubizjak@gmail.com>
Link: https://lore.kernel.org/r/20220713151946.4743-1-ubizjak@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Each kernel doc comment expects the definition of the return value and
the summary for each struct / enum in a proper format. This patch
adds or fixes the missing entries for compress-offload API.
Link: https://lore.kernel.org/r/20220713104759.4365-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If early probe of HDAudio bus driver fails e.g.: due to missing
firmware file, snd_hda_codec_shutdown() ends in manipulating
uninitialized codec->pcm_list_head causing page fault.
Iinitialization of HDAudio codec in ASoC is split in two:
- snd_hda_codec_device_init()
- snd_hda_codec_device_new()
snd_hda_codec_device_init() is called during probe_codecs() by HDAudio
bus driver while snd_hda_codec_device_new() is called by
codec-component's ->probe(). The second call will not happen until all
components required by related sound card are present within the ASoC
framework. With firmware failing to load during the PCI's deferred
initialization i.e.: probe_work(), no platform components are ever
registered. HDAudio codec enumeration is done at that point though, so
the codec components became registered to ASoC framework, calling
snd_hda_codec_device_init() in the process.
Now, during platform reboot snd_hda_codec_shutdown() is called for every
codec found on the HDAudio bus causing oops if any of them has not
completed both of their initialization steps. Relocating field
initialization fixes the issue.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220706120230.427296-7-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AVS HDAudio bus driver does not tie with codec drivers tighly. Codec
device and its respective driver cleanup procedures are split and may
not occur one after the other. Device cleanup is performed only on
snd_hdac_ext_bus_device_remove() i.e. it's the bus driver's
responsibility. If codec component probing fails, put_device() found in
snd_hda_codec_device_new() may lead to page fault. Relocate it to
snd_hda_codec_new() to address the problem on ASoC side while keeping
status quo for snd_hda_intel.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220706120230.427296-5-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AVS HDAudio bus driver does not tie with codec drivers tighly and
snd_hda_codec_device_new() can be called after codec's module reload. In
such case, rpm is forbidden and invoking pm_runtime_forbid()
unconditionally causes device's usage_count to become unbalanced. This
is later caught by WARN_ON() found in sound/soc/hda.c. Detect such
circumstance and bump the usage_count instead.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220706120230.427296-4-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If snd_hda_hdmi_codec module is denylisted and any event causes i915
enumeration to fail, is_likely_hdmi_codec() ends in null-ptr-deref.
As snd_soc_hda is an ASoC-based driver, its initialization is delayed
until all the necessary components appear in the system - allowing
actual sound card to enumerate. snd_hda_codec_configure() gets called by
the avs-driver core during probe_codecs() but the
snd_hda_codec_device_new(), necessary to complete codecs initialization,
happens only when codec-component of hda sound card is being probed.
Denylisting snd_hda_codec_hdmi module causes snd_hda_codec_configure()
to reach: codec_bind_generic() -> is_likely_hdmi_codec() which makes use
of ->wcaps and at this point the it isn't initialized yet - again,
requires completion of snd_hda_codec_device_new().
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220706120230.427296-3-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sparse reports
sound/pci/hda/patch_cs8409-tables.c:79:25: warning: symbol 'cs8409_cs42l42_pincfgs_no_dmic' was not declared. Should it be static?
cs8409_cs42l42_pincfgs_no_dmic is only used by cs8409_fixups table as an
initializer for the hda_fixup element v.pins. Both are defined in the
patch_cs8408-table.c file but only cs8409_fixups is used externally in
patch_cs8409.c. So cs8409_cs42l42_pincfgs_no_dmic should have a static
storage class specifier.
The other v.pins initializers in cs8409_fixups table, though declared
extern in patch_cs8409.h are also only used in patch_cs8409-tables.c.
So change all the v.pins initializers to static.
Fixes: 9e7647b507 ("ALSA: hda/cs8409: Move arrays of configuration to a new file")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20220704142836.636204-1-trix@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TRACE_EVENT() macro is broken up into various parts to be efficient.
The TP_fast_assign() is just to record the event into the ring buffer, and
is to be done as fast as possible as this occurs during the actual running
of the code. The slower this is, the slower the code that is being traced
becomes.
The TP_printk() is processed when reading the tracing buffer. This is
considered the slow path. Any processing that can be moved from the
TP_fast_assign() to the TP_printk() should do so.
For some reason, the entire string processing of the trace events
hda_send_cmd, hda_get_response, and hda_unsol_event was moved from the
TP_printk() into the TP_fast_assign(). On top of that, the
__dynamic_array() was used with a fixed size of HDAC_MSG_MAX, which is
useless as a dynamic_array as it will always allocate HDAC_MSG_MAX bytes
on the ring buffer and even save that amount into the event (as it expects
the size to be dynamic, which using a fixed size defeats that purpose).
Instead, just save the necessary elements in the TP_fast_assign() and do
the string manipulation in the slow path.
The output should be the same.
Signed-off-by: Steven Rostedt (Google) <rostedt@goodmis.org>
Link: https://lore.kernel.org/r/20220703110605.07a86fb2@rorschach.local.home
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer end is being re-assigned the same value as it was initialized
with in the previous statement. The re-assignment is redundant and
can be removed.
Cleans up clang scan-build warning:
sound/isa/wavefront/wavefront_synth.c:582:17: warning: Value stored
to 'end' during its initialization is never read
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://lore.kernel.org/r/20220629102744.139673-1-colin.i.king@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes the need to power cycle the Dell WD15 dock if it has been
attached to a Windows machine.
The Windows driver puts the ALC4020 USB audio controller into
'manual mode', and then does all the power management and other
configuration itself, by sending HD audio commands directly to the
ALC3263 audio codec via vendor-type USB messages. If manual mode is off,
this is all handled by the firmware, and works well enough.
If manual mode is turned on, the latency of the SET INTERFACE command
goes from several hundred ms to less than 1 ms
(see https://bugzilla.suse.com/show_bug.cgi?id=1089467), but I'm not
sure if the additional code that would be required is worth it.
Funnily enough, the Windows driver tries to turn off manual mode when
the dock is disconnected, which doesn't work for obvious reasons.
Additionally, fix a bug in dell_dock_init_vol, which didn't work because
the Control Selector was missing.
Now, it properly resets the volume to 0dB.
Fixes: 964af639ad ("ALSA: usb-audio: Initialize Dell Dock playback volumes")
Signed-off-by: Jan Schär <jan@jschaer.ch>
Link: https://lore.kernel.org/r/20220627171855.42338-2-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Dell WD15 dock has a headset and a line out port. Add support for
detecting if a jack is inserted into one of these ports.
For the headset jack, additionally determine if a mic is present.
The WD15 contains an ALC4020 USB audio controller and ALC3263 audio codec
from Realtek. It is a UAC 1 device, and UAC 1 does not support jack
detection. Instead, jack detection works by sending HD Audio commands over
vendor-type USB messages.
I found out how it works by looking at USB captures on Windows.
The audio codec is very similar to the one supported by
sound/soc/codecs/rt298.c / rt298.h, some constant names and the mic
detection are adapted from there. The realtek_add_jack function is adapted
from build_connector_control in sound/usb/mixer.c.
I tested this on a WD15 dock with the latest firmware.
Signed-off-by: Jan Schär <jan@jschaer.ch>
Link: https://lore.kernel.org/r/20220627171855.42338-1-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v5.19
A collection of fixes for v5.19, quite large but nothing major - a good
chunk of it is more stuff that was identified by mixer-test regarding
event generation.
The patch applies the same quirks used for SC-01 at firmware v1.1.0 to
the ones running v1.0.0, with respect to hard-coded sample rates.
I got two more units and successfully tested the patch series with both
firmwares.
The support is now complete (not accounting ASIO).
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-2-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fiero SC-01 is a USB sound card with two mono inputs and a single
stereo output. The inputs are composed into a single stereo stream.
The device uses a vendor-provided driver on Windows and does not work
at all without it. The driver mostly provides ASIO functionality, but
also alters the way the sound card is queried for sample rates and
clocks.
ALSA queries those failing with an EPIPE (same as Windows 10 does).
Presumably, the vendor-provided driver does not query it at all, simply
matching by VID:PID. Thus, I consider this a buggy firmware and adhere
to a set of fixed endpoint quirks instead.
The soundcard has an internal clock. Implicit feedback mode is required
for the playback.
I have updated my device to v1.1.0 from a Windows 10 VM using a vendor-
provided binary prior to the development, hoping for it to just begin
working. The device provides no obvious way to downgrade the firmware,
and regardless, there's no binary available for v1.0.0 anyway.
Thus, I will be getting another unit to extend the patch with support
for that. Expected to be a simple copy-paste of the existing one,
though.
There were no previous reports of that device in context of Linux
anywhere. Other issues have been reported though, but that's out of the
scope.
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-1-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
madera_out1_demux_put returns the value of
snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for
the kcontrol. This is obviously different to the expected return a 1 if
the control was updated value. This results in spurious notifications to
user-space. Update the handling to only return a 1 when the value is
changed.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623105120.1981154-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The mixer controls for ASP TX3/4 are set to values that are not included
in their enumeration control. This will cause spurious event
notifications when the controls are first changed, as the register value
changes whilst the actual visible enumeration value does not. Use the
register patch to set them to a known value, zero, which equates to zero
fill, thereby avoiding the spurious notifications.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623105120.1981154-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAPM keeps a copy of the current value of mux/demux controls,
however this value is only initialised in the case of autodisable
controls. This leads to false notification events when first
modifying a DAPM kcontrol that has a non-zero default.
Autodisable controls are left as they are, since they already
initialise the value, and there would be more work required to
support autodisable muxes where the first option isn't disabled
and/or that isn't the default.
Technically this issue could affect mixer/switch elements as well,
although not on any of the devices I am currently running. There
is also a little more work to do to address the issue there due to
that side supporting stereo controls, so that has not been tackled
in this patch.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623105120.1981154-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The initial settings will be written before the codec probe function.
But, the rt711->component doesn't be assigned yet.
If IO error happened during initial settings operations, it will cause the kernel panic.
This patch changed component->dev to slave->dev to fix this issue.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20220621090719.30558-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We discoverd that the state of BCLK on, LRCLK off and SD_MODE on
may cause the speaker melting issue. Removing LRCLK while BCLK
is present can cause unexpected output behavior including a large
DC output voltage as described in the Max98357a datasheet.
In order to:
1. prevent BCLK from turning on by other component.
2. keep BCLK and LRCLK being present at the same time
This patch switches BCLK to GPIO func before LRCLK output, and
configures BCLK func back during LRCLK is output.
Without this fix, BCLK is turned on 11 ms earlier than LRCK by the
da7219.
With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by
the rockchip codec.
Signed-off-by: Judy Hsiao <judyhsiao@chromium.org>
Link: https://lore.kernel.org/r/20220615045643.3137287-1-judyhsiao@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
